Interconnect

Monday, July 14, 2014

Middle Atlantic MPR Now Brings Series Surge Protection To Vertical Power Strips

Also provides series-type surge protection to all cascading modules in each circuit.

Middle Atlantic Products is expanding its Modular Power Raceway (MPR) system to include surge protection capabilities with the introduction of Series Protection modules for the MPR Series.

The debut marks what the company states is the industry’s first vertical power strip to offer series-type non-degrading surge protection, and its first fully configurable power solution to do the same.

The new Series Protection modules will also provide series-type surge protection to all cascading modules in each circuit. A patent-pending technology, Middle Atlantic states that Series Protection is the fastest-responding series type surge suppression currently available.

The MPR system allows integrators to configure their own ETL-listed vertical power solution using off-the-shelf parts. When installed as the first circuit within the power strip, these modules will extend that series type surge protection to all connected units that follow. 

In addition, the module sits on top of raceway channel, which allows circuit wiring between modules and unimpeded connection to the termination.

The new MPR modules join the Middle Atlantic family of rackmount and RackLink power products leveraging Series Protection technology.  Modules are available in 15- and 20-amp configurations with outlet options being controlled or on when connected.

Middle Atlantic Products
Legrand

{extended}
Posted by Keith Clark on 07/14 at 12:30 PM
AVLive SoundRecordingChurch SoundNewsProductAVInterconnectPowerSignalSound ReinforcementStudioPermalink

Real World Gear: Making Choices

Perhaps more than any other pro audio component, microphones are a personal choice, subject to the specific preferences and goals of both users and sound techs/engineers.

Manufacturers continue to create new mics in a wide range of formats, sizes, and price points, stemming from a healthy mix of innovative development, refinement of previous technologies, specialization of mics to specific applications, and changes in manufacturing.

For decades, dynamic mics were pretty much the exclusive choice for live applications due to their ruggedness. Live engineers didn’t want to take delicate, expensive condenser mics on the road. That’s changed in a big way now that condensers have been made more robust and roadworthy, and they’re quite capable of handling a wide range of live applications. More tools to help produce the desired result is indeed a good thing.

Over the years, we’ve focused on the principals of dynamics and condensers, along with ribbon and figure 8 designs – and everything in between. Rather than revisit that territory here, we encourage you to visit Microphone World here on PSW, which provides more than 150 reference articles focusing on “all things microphone” – not just designs (although there’s plenty of that) but also a wide range of applications, techniques, history, and engineer/tech interviews. Related topics are also addressed, including cabling, phantom power, stands and mounts… well, you get the idea.

Certain mics perform “better” in certain applications, although there are few hard and fast rules. Again, a lot of mic selection comes down to individual preference and meeting a specific need. LSI church sound editor Mike Sessler shared an anecdote in a ProSoundWeb post a couple of years ago that gets to the heart of the matter:

“When we bought our new wireless system, I specified a capsule that I planned on using for our worship leader. Turns out, it doesn’t work for him. And as we’ve tried it on many of our vocalists, it doesn’t work for most of them either. In fact, some of them really don’t like it. So here we have a capsule that costs over $500, and for the most part, we and most of our singers prefer capsules that sell for less than half that. Quite honestly, I’d be really ticked if I had ordered 10 of those capsules instead of 10 of the others based on the notion that more expensive = better. In fact, I’m going back and ordering a few more of the less expensive ones, because in our PA, with our singers, they are a superior choice.”

The specifics of each application also impact mic choices. For example, in some acoustic music situations, a drum kit can be viewed as a single instrument, captured with a pair of mics in a stereo configuration (or a single stereo mic). They pick up the kit as a whole, and the balance among the various pieces depends more upon the drummer.

On the other hand, a drum kit can be viewed as a collection of individual instruments, picked up with close mics applied to each drum. Or, perhaps a combination of area and close miking is most fitting for the performance. In each of these applications, success or failure is primarily based upon mic selection and placement (along with the skill of the mix engineer, of course).

But what are the specific mics selected? Beyond the one common component – drums – the specifics of these applications vary widely. It’s not enough to select mics designed and marketed for drums and then assume it will work out for the best. Mic selection is often a matter of trial and error, requiring hands-on investigation.

And as Ken DeLoria noted in a recent article about mic selection, “Whenever you have available time, don’t hesitate to try a second, third, or even a fourth mic on a given instrument. In this way, you can compare it to your go-to selection in real time, at a real event (or at least at sound check). You may just find some surprising results.”

The purpose of this installment of Real World Gear is in line with that thinking, to provide a solid starting point for the process by highlighting a wide variety of microphone designs that deserve further consideration. Enjoy our photo gallery tour of numerous models.

{extended}
Posted by Keith Clark on 07/14 at 10:09 AM
Live SoundFeatureBlogProductSlideshowInterconnectMicrophoneSound ReinforcementWirelessPermalink

RØDE Microphones Announces Updated iXY Microphone For iPhone 5, 5s And 5c

iXY with Lightning connector has a matched pair of 1/2-inch condenser capsules arranged in a stacked X-Y configuration

RØDE Microphones has announced the iXY with Lightning connector for iPhone 5, 5s and 5c, an update to the company’s stereo microphone for Apple iPhone devices that fosters recording at sample rates up to 24-bit/96kHz.

Equipped with the same audio performance as the original 30-pin microphone, the iXY with Lightning connector has a matched pair of 1/2-inch condenser capsules arranged in a stacked X-Y configuration, with on-board high-fidelity analog to digital conversion to help ensure accurate, immersive and true-to-life stereo recordings.

Interchangeable rubber mounting clamps are supplied to suit both iPhone 5/5s and 5c, which also provide shock mounting and help to minimize vibration transferring to the microphone capsules. A foam windshield for outdoor recording and protective storage pouch are also included. The RØDEGrip mount is optionally available for mounting the iXY and iPhone on a camera or microphone stand, and a “deadcat” windshield for high wind conditions will be available shortly.

“The original iXY was an industry leader in both design and function, so we’re thrilled to continue this success with the updated version for iPhone 5,” states Damien Wilson, RØDE global sales and marketing director. “Being able to record at this quality with a device that sits in your pocket has been a game changer not only for audio professionals, but for enthusiasts alike. We saw this first-hand with so numerous entrants to our My RØDE Reel short film competition choosing to use the iXY for their amazing short films”.

In addition, RØDE’s field recording app for iOS devices, RØDE Rec, also recently received an update to increase compatibility and stability on the iOS 7 platform. RØDE Rec and RØDE Rec LE are available now in the App store for iPhone and iPad.

The RØDE iXY is shipping now.

RØDE Microphones

{extended}
Posted by Keith Clark on 07/14 at 06:26 AM
AVLive SoundRecordingNewsProductAVInterconnectMicrophoneSound ReinforcementStudioPermalink

Friday, July 11, 2014

Front-of-House Engineer Rick Camp Utilizing Aviom A360s For Master Mix Live Program

Helps students explore various methods of monitor mixing

Front-of-house engineer Rick Camp has been busy on tour this year with Jennifer Lopez, but when he’s not on the road, he’s busy teaching others the art of live engineering in his Master Mix Live program in Las Vegas. Included in the mix of gear that Camp is using in his Las Vegas studio and teaching space are new Aviom A360 personal mixers.

Camp’s program is unique—it’s small on purpose, accepting only eight students at a time so they get the opportunity to have hundreds of hours of hands-on console time throughout the five-month course. And each student has the opportunity to work personally with Camp, who has experience mixing many famous artists, including Earth, Wind & Fire, Madonna, Toni Braxton, Tracy Chapman, Usher, and many more.

Camp’s decision to integrate an Aviom personal mixers into the Master Mix Live space was two-fold. One of the courses included in the Master Mix Live program focuses on monitor mixing. In this course, students explore various methods of monitor mixing—everything from using wedges to in-ear monitors and personal mixers. Camp also uses the A360s in his recording studio to provide artists’ individual headphone mixes.

“I think that students find it easier to let musicians dial in their own monitor mixes, bur for me in the studio, the personal mixers are invaluable when trying to please performers who want their headphone mix changed every eight bars for one reason or another to accommodate the passage they are playing or singing,” explains Camp.

Camp’s studio includes eight A360 personal mixers, an AN-16/i v.2 input module, and an A-16D Pro A-Net distributor. According to Camp. “The A360 is hands down the best sounding and most flexible personal mixer I’ve seen or used to date.”

Master Mix Live
Aviom

{extended}
Posted by Keith Clark on 07/11 at 02:07 PM
AVLive SoundChurch SoundNewsAVInterconnectMixerMonitoringNetworkingSound ReinforcementStagePermalink

Park Road Post Production In New Zealand Invests In DirectOut MADI Solution

Fosters a much more flexible MADI system for the mix team

Park Road Post Production, a premier post production facility based in Wellington, New Zealand, has recently invested in a new MADI solution from DirectOut Technologies to improve workflow in the mix area.

Owned by Academy Award-winning film director Peter Jackson, best known for having directed and produced ‘The Hobbit’ and ‘The Lord of the Rings’ trilogy, Park Road was established as a one-stop shop with integrated picture and sound departments all under one roof. The facility forms part of a creative community that includes the Academy Award-winning Weta Digital and Weta Workshop and the soundstages of Stone Street Studios.

Park Road head of sound John Neill explains that the initial reason for the upgrade was to reduce the number of devices within the mix area in order to reduce background fan noise. In doing so, he found that he actually ended up with a much more flexible MADI system for the mix team in the bargain.

Neill worked with Wellington-based AV solution provider Protel and DirectOut Australia/NZ distributor Professional Audio & Television to integrate a number of MA2CHBOX headphone amplifiers for MADI signals, and an ANDIAMO.XT AD/DA/DD converter into the existing environment, at the heart of which is a Euphonix System 5 console.

The MA2CHBOX BNC units are used to provide the Protools editors on the mix stage with a direct feed of their first 64 tracks via headphones. According to Neill, “This meant that the feed existed even when staff were opening other projects on the console. The challenge was to get a split of the MADI feed and send it via a long cable to the mix theatre. This was achieved by isolating the split to the long cable on the console in such a way that it neither affected, nor was affected by the console status.”

Meanwhile, the ANDIAMO.XT ensures high-quality A/D (MADI/AES) and D/D conversion in a compact form factor at an affordable price point.

“The DirectOut products were the only devices to offer all the functionality we required at the standard we demanded,” concludes Neill. “The project is a total success and we’ll definitely extend this system to other areas in the future.”

DirectOut Technologies

{extended}
Posted by Keith Clark on 07/11 at 10:39 AM
Live SoundRecordingNewsDigitalDigital Audio WorkstationsInterconnectNetworkingSignalStudioPermalink

Wednesday, July 09, 2014

19th Century Connecticut Castle Modernizes With QSC Audio

New system includes AcousticDesign and AcousticPerformance loudspeakers, CXD and CX amplifiers, controlled by a Q-Sys Core 250i integrated platform

Bill Miller’s Castle, a 24-acre wedding, corporate meeting and special events facility located east of New Haven on Connecticut’s central shoreline, recently completed a major upgrade to its installed sound system.

Design and integration of the venue’s new multipurpose sound system was provided by KJR Engineering of Middletown, CT and includes 38 QSC Audio AcousticDesign and AcousticPerformance loudspeakers installed in 23 zones powered by CXD and CX amplifiers, all controlled by a Q-Sys Core 250i network audio system.

Built in 1880 and maintained by the Millers—a family of renowned gymnasts, acrobats and dancers—since the early 1960s, the picturesque building was initially converted into a gymnastics and dance studio before becoming what was then the first discotheque in Connecticut, the “Sugar Shack.”

The present-day Bill Miller’s Castle now features a large dance floor and multi-level dining areas under three-story cathedral ceilings with leaded stained glass windows and crystal chandeliers, plus balconies, a cocktail bar, buffet and outdoor deck with gazebo.

The new main system has AcousticPerformance AP-5102 2-way loudspeakers on either side of the main stage, plus two GP-212sw subwoofers, to support the bands and DJs that frequently perform at castle events.

The main system is augmented by a series of delay zones utilizing AcousticDesign AD-S8T surface-mount 2-way loudspeakers: one as a mono delay for the lower balcony, two stereo pairs as delay zones in the main room, two more on a mono delay line further into the main room, and one more stereo delay pair for the upper balcony.  There is also one more mono pair of AD-S8T in the chapel, with one serving as the main speaker the other as a delay zone.

“The room was a challenge because of the numerous ceiling height changes which required the application of a little EQ and temporal alignment,” reports Bobby Kuhl, president and senior audio engineer with KJR Engineering. “Fortunately, the loudspeakers are all voiced through the CXD amps, which allowed us to take full advantage of the amp’s built-in Intrinsic Correction tunings.”

A main rack houses the Q-Sys Core 250i DSP platform, two CXD4.5 4-channel 2,000-watt amplifiers and a pair of 4-channel 1,400-watt CXD4.3, plus a CX108V 8-channel amplifier for the 70-volt system. “I believe we installed one of the first AD-S8T/CXD combinations in the United States and that we were one of the first installers of the advanced CXD amps,” says Kuhl. “As an early adopter, I did three rooms for Wesleyan University in Connecticut last year using CXD amps. The CXD amps are just terrific.”

A 70-volt system provides coverage of the rest of the facility, including the bar and reception room, which each outfitted with four AD-Ci52ST 2-way ceiling loudspeakers. an AD-S52T surface-mount 2-way loudspeaker in the office; another AD-S52T in one bridal suite; and three more AD-S52T loudspeakers, as main and delay zones for the second bridal suite. The exterior perimeter system includes three AD-S52T loudspeakers.

“The rack in the stage area has a Q-Sys I/O Frame with 16 inputs and a bunch of computer and playback media, including a couple of hard drives,” says Kuhl. A custom patchbay supports I/O for a diverse selection of media, including Blu-ray Disk, DVD, USB, FireWire 800 I/O, HDMI output.

Two portable racks each loaded with a standalone Q-Sys I/O-22 unit, two channels of wireless microphone receivers, a low-power amp for local monitoring and a custom patch panel is linked back via Q-LAN to the Core 250i over redundant Cat6 cables and helps expand the system to 24 inputs.

Kuhl adds that parts of the install required some unconventional approaches. “For example, as part of the 70-volt system we used a ceiling-mount box for the AD-C820 in the gazebo and turned it upside down so that it fires straight up using the roof as a reflector,” states Kuhl. “This allows for very even coverage even at extremely low volumes. To minimize sight lines and improve aesthetics, we painted the can to match the gazebo and mounted it to the cross beam.”

When Kuhl initially toured the facility, Miller family patriarch Jan Miller, explained that he wanted to be able to remotely control the background music during events and whenever he took prospective clients on a walk-through of the amenities.

Using Q-Sys Designer software, Kuhl created a simple UCI for two iPads, one housed in a wall mount next to the system’s power up/down sequencer, and a second, mobile tablet. With that the Millers are able to control the entire system with two IPads and according to Kuhl, “They are extremely happy with the results, Jan texts me weekly with glowing reports from clients and the visiting performers.”


image


QSC Audio

{extended}
Posted by Keith Clark on 07/09 at 04:20 PM
AVLive SoundNewsAVInstallationInterconnectLoudspeakerNetworkingProcessorSound ReinforcementSubwooferPermalink

Invaders From Outer Space: Electromagnetic Radiation & Interference

Electromagnetic interference, often referred to as EMI, is a degradation of the performance of a piece of equipment or system caused by electromagnetic radiation.

Electromagnetic radiation is the travel of alternating electric and magnetic fields through space at 186,273 miles (about a billion feet) per second, or the speed of light.

A wavelength, as shown in Figure 1 below, is the physical distance traveled by a wave during one cycle. When the equipment or system is closer than about a sixth of a wavelength from the radiation source, it’s said to be in the nearfield of the source.

Most audio systems are in the nearfield at all frequencies below about 1 MHz, including 60 Hz (hum) and its harmonics (buzz). In the nearfleld, either the magnetic or electric field dominates, and each requires a different kind of treatment to eliminate interference.

For example, twisting of signal pairs reduces pickup of magnetic fields while shielding reduces pickup of electric fields. (We’ve discussed these topics in previous articles here).

However, when radiation sources have frequencies above about 1 MHz, the magnetic and electric fields travel together and are collectively called a wave. Most audio systems are in the far-field of the radiation source at these “radio” frequencies, hence the older term RFI for radio-frequency interference.

Figure 1: High-frequency magnetic and electric fields travel fogether as a “wave.”

Generally, the newer term EMI also refers exclusively to radio-frequency interference. RF interference may originate in either intentional or unintentional radiators. Familiar intentional radiators include broadcast stations, amateur transmitters, and cell phones.

Unintentional radiators include devices such as electric welders, brush-type motors, relays, and switches that produce arcs, which are potent sources of brief bursts of wide-band interference. Less obvious sources include arcing or corona discharge on power line insulators (common in coastal or high-humidity areas) or malfunctioning fluorescent or neon lighting. Of course, lightning (the ultimate spark) is a well-known producer of momentary RF interference to virtually anything electronic.

A list of frequencies and corresponding wavelengths.

About Antennas
An antenna converts an electrical signal into electromagnetic radiation and vice-versa. In other words, a good receiving antenna is also a good transmitting antenna. Most intentional transmitters and receivers use highly efficient antennas carefully “tuned” for the purpose.

But, depending on frequency, almost any conductor can become an antenna—even if we call it an audio cable, an RS-232 cable, or a printed-circuit board! [1] RF interference is most often picked up by signal cables, which then deliver the induced voltage to the audio equipment.

A cable, or circuit board trace, becomes a particularly efficient antenna when its physical length is a half, quarter, or eighth of a wavelength at the interference frequency. Therefore, any cable longer than about 75 feet O/S wavelength at 1,600 kHz) can inadvertently become an AM radio antenna and a cable only a few feet long can become an efficient VHF/FM antenna because of the shorter wavelengths.

For actual field measurements of AM radio levels induced in various cables (for more, see Reference #2 following this article). Sometimes a cable both radiates and conducts RFI. Electric welder cables can radiate RF energy from the arc as well as conduct it, via the power supply, to the AC power line.

Premises AC power wiring can then conduct RF interference over significant distances, but the transmission generally becomes very loss-y at frequencies over a few hundred kHz. This is only a hint of how complex EMI issues can become.

Effect On Equipment
RF interference voltages induced in system cables connected to electronic equipment would be of little concern if it weren’t for the fact that most equipment does a very poor job of ignoring it. Some audio equipment appears to have been designed with utter disregard for the fact that it may be used in an electrical environment that’s teeming with RF radiation from an AM radio transmitter a half-mile away or a Blackberry sitting 6 inches from the microphone.

At frequencies well above the audio range, the circuitry in audio equipment becomes much more complex than that, shown by its schematic diagram. Wires and PC board traces become antennas and inductors, capacitors begin to behave like inductors, resistors begin to behave like capacitors, and power transformers begin to behave like coupling capacitors.

Worst of all, semiconductors (diodes, transistors, and ICs) tend to demodulate the RF interference, producing audible artifacts. Symptoms range from actual music or voices or tones (buzz in the case of TV transmitter interference) to subtle distortions, often described as a “veiled” or “grainy” quality in the reproduced audio. [3] The enabling electronic mechanisms include Simple rectification, amplifier asymmetrical non-linearities, and asymmetrical slew-rate limiting.

Immunity to RFI is part of good equipment design. Well-designed equipment will include RF low-pass filters at all inputs and outputs (including power and data ports) as well as proper shielding and grounding techniques to prevent significant RF voltage from ever reaching active circuitry that might be affected by it. Although testing for RFI susceptibility is now mandated in Europe, the CE mark is certainly no guarantee of trouble-free performance.

Figure 2: Clamp-on Ferrite cores. (click to enlarge)

Sadly, much of the equipment available today still has very poor immunity. Under unfavorable conditions, external measures may be needed to achieve adequate immunity.

External Add-On Solutions
Ferrite “clamshell” cores, like those shown in Figure 2, are easily installed over the outside of existing cables and can be very effective for interference over about 20 MHz, although certain ferrite formulations work well at frequencies under 3 MHz.

Figure 3: Filters to eliminate AM radio interference. (click to enlarge)

In most cases, they work best when placed on the cable at or near the equipment. Often they’re more effective if the cable is looped through the core several times.

If this is inadequate, or the frequency is lower (such as AM radio), a low-pass (high-frequency reject) RFI filter may be necessary in the signal line. The schematics in Figure 3 show sample filters for unbalanced and balanced line-level applications.

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

References
[1] Henry Ott, Dipoles For Dummies—Part 1, 2002.

[2] Jim Brown & Bill Whitlock, Common-Mode to Differential-Mode Conversion in Shielded Twisted-Pair Cables (Shielded-Current-Induced Noise), Audio Engineering Society 114th Convention, 2003, Preprint #5747, www.aes.org.

[3] Deane Jensen & Gary Sokolich, Spectral Contamination Measurement, Audio Engineering Society 85th Convention, 1988, Preprint #2725, www.aes.org.

[4] Bill Whitlock, Understanding and Controlling RF Interference, Sound & Video Contractor, February 1999.

{extended}
Posted by Keith Clark on 07/09 at 02:45 PM
AVFeatureStudy HallAVInterconnectPowerSignalWirelessPermalink

Tuesday, July 08, 2014

Going Direct: The Ins & Outs Of DI Boxes

Given the wide variety of audio sources that are connected to the microphone and line inputs of the mix console, the availability of high-quality DI boxes is a true blessing. Electric basses, acoustic guitars with piezo transducers, other stringed and wind instruments with pickups, effects units, CD players, computers, and more contribute to the overall audio palette of the event.

DI boxes (also called direct boxes) are the tools that allow the disparate sources, each with its own distinct functions, output levels, and impedance characteristics. to connect with the sound reinforcement system properly – without adding excessive noise and/or altering frequency response. 

Primary Applications
DI boxes provide three basic functions. First, they convert unbalanced signals from sources such as instrument pickups and electronic instruments into balanced signals that can travel longer distances without induced interference or signal degradation.

Second, they help with impedance matching, especially from high-impedance sources like passive guitar pickups being fed into low-impedance mic inputs on the mixing console.

Third, while performing the above electronic functions, DIs act as an interface to change from one connector type to another, typically from 1/4-inch to XLR. 

As an added benefit, the audio transformer within a passive DI will break a ground loop. Some active DIs also place a transformer within the input-to-output path for isolation. Because the transformer passes the signal from the primary to the secondary coil via induction versus requiring a physical connection, the ground current cannot flow and create hum and buzz. 

Passive & Active
DI boxes are made in both passive and active formats, and each has its primary uses and advantages. Basically, a passive design requires no external power to function, and its internal audio-quality transformer performs the conversion functions.

Radial Engineering offers numerous types of DIs, ranging from the passive StageBug SB-5 for laptop computers to the active PZ-DI optimized for orchestral instruments.

An active DI requires power from a phantom power source and/or from a battery. Electronic circuitry is used for the signal balancing and impedance matching functions. Distinguishing the two types, whether they’re labeled or not, can usually be done by noting the absence or presence of a battery compartment, an on/off switch, and an LED. (Note, however, that there are active units that do not have all of these elements.) 

A basic rule of thumb is to use a passive DI with an active source, and an active DI with a passive source. A standard electric guitar and many basses are passive sources, as are most acoustic guitars with under-saddle pickups and other instruments with piezo pickups. Keyboards, active pickup systems, effects and other electronic devices, as well as audio sources with a battery or an AC plug, are active sources.

Mick Conley, mix engineer for country musician Marty Stuart, says that when connecting a bass with passive pickups, he “usually uses an active DI to help with the signal level and clarity,” while a bass with active pickups will have “a passive DI, since there’s no need for the extra output, and a passive DI can help keep the tone from getting too aggressive.”

Mikail Graham, sound engineer at the Grass Valley Center for the Arts, tells me, “I tend to use active DIs for bass and guitars, with passive units primarily for keys and various processors.” He adds that “keyboards and rhythm boxes, as well as the ever encroaching array of vocal effects processors, also benefit greatly when used with a DI.”

Nick Malgieri, AV manager and audio engineer at Stanford’s Bing Concert Hall, states, “Generally speaking, I prefer active DIs. I find the higher output and high-presence tone to be better for most applications. Passives work as well and are often preferred by rock ‘n’ roll engineers who prefer a softer, rounded tone, although I find them to sound a bit flat and the low output can drive up noise floor.”

The design decision of whether an active DI will allow battery powering or use phantom power from the console relates to the potential limitations that battery power poses to the maximum signal level the DI can handle, as well as the fact that batteries deliver less voltage as they’re used up, and that they can run out of juice in the middle of a show. Units such as the Radial Engineering PZ-DI and the Klark Teknik DN100 deliberately forego the battery option.

There may be circumstances where a convenient source of phantom power is not available at a particular location, and an active DI is necessary; for example to connect mixing consoles in different locations while using the ground lift, or acting as a line balancer. The Countryman Type 10 active DI works with both phantom and battery power, and it also includes a power monitoring circuit with a pair of LEDs and a power-test switch that tracks the relative levels and can transition between sources to maintain the best performance. 

I/O Impedance
The output impedance of a passive electric guitar or bass pickup can be in the hundreds of kilohms, and that of an under-saddle piezo pickup even greater – and the impedance varies with the frequency of the note being played.

The Klark Teknik DN100 foregoes battery operation.

In order to transfer the audio signal correctly, rather than attenuating the instrument’s lower or higher frequencies, the input impedance of the next device in the signal chain must be considerably higher. In this regard there are numerous choices; for example, the impedance of the BSS AR-133 is 1 megohm while the Leon Audio Mk 2A is 33 megohms. (Both are active units.)

When connecting an acoustic guitar with piezo pickups, Conley notes, “I like the Radial PZ-DI because of the impedance button on the side of the unit; it helps to match the impedance better and therefore warms the tone.” The PZ-DI has impedance settings of 220K, 1M, and 10M to accommodate devices from lower to very high output impedance. 

Passive units have an input impedance about an order of magnitude lower, with typical units in the range of 50 to 150 kilohms. The lower output impedances of keyboards, effects, CD players, and the like – ranging from a few hundred ohms to several kilohms – means that the input of a passive DI will be more than sufficient to accept the audio signal without introducing frequency-response problems.

Some makers offer DI boxes with single and dual channels, as seen here with the Countryman Type 10 and 10S (“S” stands for stereo).

Also, the passive DI can be less prone to overload distortion with high signal levels, since transformers saturate at higher levels rather than distort, which can be more pleasing to the ear. 

The XLR output of a DI box is low impedance, similar to that of a microphone, so that it properly interfaces with the mic input of a mixing console. The output level is also closer to that of a microphone, so that the normal range of the channel’s trim control is able to make any fine level adjustments, rather than seeing a signal that is too low or too hot.

Attenuation buttons or “pads” on a DI are often available to make larger adjustments to the signal level before it is sent to the console, with a variety of values depending on the DI, in the range of -10 dB to -40 dB.

Going The Distance
An instrument-level signal going through an unbalanced 1/4-inch guitar cable is only going to travel a few feet before the capacitance of the cable will roll off some of the high frequencies.

For a typical performing-length cable, this can be part of the desired sound when connected to a nearby amp. However, taking that lower level signal all the way to the console while unbalanced would undo the tone of the instrument, and open it up to induced electrical noises as it travels by various other cables carrying AC and other signals.

A DI box converts the unbalanced signal to a balanced one right at the stage, so that it can be more resistant to interference as it travels the sometimes hundreds of feet back to the console inputs.

Along with the balanced XLR output that takes the signal to the analog snake (or digital converter box at the side of the stage) and out to the sound reinforcement system, DIs will usually have an additional unbalanced jack that loops the unadulterated instrument signal to the performer’s on-stage amplifier, so that the guitar or bass amp “sees” the instrument’s pickups. Thus the DI can also function as a signal splitter.

In addition to filling their functions of signal balancing, correcting impedance mismatches, and breaking ground loops, DI’s are audio devices that are inserted directly in the path between the transducer capturing the audio source and the mixing and amplification components of the sound reinforcement system. The quality and transparency of the signal they output is critical to how the listener will hear the sound of the instrument.

A look inside the Radial JDI, outfitted with a Jensen transformer (center component).

The key component in a passive DI is the transformer, and its design and quality is an important differentiator between a passable and a high-performance professional unit. This choice can have audible results. Companies such as Lundahl and Jensen specialize in manufacturing transformers with excellent audio characteristics – and with prices to match.

The quality of the circuitry within an active DI is also a critical factor in its audio response characteristics, as well as being a differentiator of the higher-quality boxes. Key measures are frequency response and flatness across the audio spectrum. 

Road Ready
Because DI boxes are distributed around the stage in unprotected locations, they’re usually ruggedly built devices, often weighing a pound or two, though a few more diminutive (yet still rugged) units can also be had.

Connectors and switches are typically recessed within an extruded chassis, with perhaps 1/8-inch-thick metal surrounding the more heavy-duty units. Some models also include thick rubber side bumpers that function as non-slide feet while offering some protection to switches and attached cable connectors.

Internal durability is also a factor. The quality of the switches, connectors, electronic components, and circuit boards directly affect how well the DI performs its functions, how long it lasts, and its immunity to induced noise. In most cases you get what you pay for, and since the cost of even a relatively expensive DI is inconsequential compared to the price of a good instrument or mixing console, investing in quality is wise. 

Gary Parks is a pro audio writer who has worked in the industry for more than 25 years, including serving as marketing manager and wireless product manager for Clear-Com, handling RF planning software sales with EDX Wireless, and managing loudspeaker and wireless product management at Electro-Voice.

{extended}
Posted by Keith Clark on 07/08 at 03:53 PM
Live SoundFeatureBlogStudy HallInterconnectProcessorSignalSound ReinforcementStagePermalink

Monday, July 07, 2014

ARX Shipping Blue DI Bluetooth Direct Box

ARX's new Blue DI features 44.1 & 48Khz sample rates at 32bits, is compatible with Bluetooth Version V2.1 EDR, 3.0 and 4.0.

ARX Systems is now shipping the unique “Blue DI” Bluetooth Direct Box, a new addition to their popular AudiBox range of Precision Tools for Audio Professionals .

SmartPhones, tablets and other Bluetooth enabled devices have rapidly become the program source of convenience for corporate AV presentations, seminars, musicians, D.Js, entertainment venues, system demonstrations and testing, and many other audio playback applications.

ARX’s Engineering team developed the Blue DI to fulfill the demand for a truly Professional wireless Active Direct Box interface allowing users to connect today’s and tomorrow’s Bluetooth enabled devices to the pro audio world of balanced signals and XLR connectors via a robust wireless connection of over 12 meters range.

ARX’s new Blue DI features 44.1 & 48Khz sample rates at 32bits, is compatible with Bluetooth Version V2.1 EDR, 3.0 and 4.0, can be powered by +48VDC phantom power from a mixing console, or an external DC PSU and features industry standard balanced left and right line level XLR outputs.

ARX Systems

{extended}
Posted by Julie Clark on 07/07 at 05:20 PM
Live SoundRecordingChurch SoundNewsProductBusinessInstallationInterconnectManufacturerPowerPermalink

Sunday, July 06, 2014

Flexible Capture: Recording Options Of Digital Consoles

Digital mixing consoles provide wealth of capabilities for recording, designed to provide simple onboard 2-track capability to interfacing directly with computer-based multi-track recording systems.

Whether configured to do so from the factory or by using optional output cards, many consoles can output MADI, a digital protocol with 64 channels of audio, or ADAT optical, a digital protocol that sends eight channels down each optical cable. Both of these are used by many recording systems for multi-channel audio transport. AES/EBU and S/PDIF are two other common digital audio protocols to interface recording and playback equipment.

Digital snake systems, stage boxes and networks present further advantages when it comes to recording. Instead of having just one isolated output located at the snake head, many digital transport systems have multiple splits that can be placed anywhere along the network, accommodating remote recording, webcasts, broadcast feeds, and any other sends that may be required.

All of that said, we thought it instructive to take a look at some specific capabilities of current digital console series. Note that this isn’t intended to be comprehensive, but rather is a roundup of highlights that can serve as the basis for your own further investigation.

Yamaha QL Series. Provides convenient recording capabilities for everything from basic 2-track to multi-track recording and playback. A standard USB flash drive plugged into the front-panel USB port serves as media for direct 2-track recording in mp3 format, where, for example, the recording can be handed to performers as soon as the show is finished.

Sound files in mp3, AAC, or WMA format saved on the flash drive from a computer or other source can be played back as well for handy background music or sound effects without the need for extra playback equipment.

On the other end of the spectrum, with Dante Virtual Soundcard software it’s possible to transfer audio directly to a Windows or Mac computer connected to the Dante network.

With an appropriate DAW such as Steinberg Nuendo Live (sold separately) running on the computer, up to 64 tracks can be recorded simultaneously. (Note that Nuendo Live is included with Yamaha CL Series consoles.) It’s a great way to capture professional caliber live performances and also is useful in creating the tracks needed for virtual sound checks. (More here)

DiGiCo SD Series. Capabilities vary depending on model but suffice to say there’s plenty of facilities. For example, the proprietary Stealth Digital Processing engine applied to the SD7 provides 896 simultaneous optical, 224 MADI, 24 AES/EBU and 24 analog connections.

Further, the SD Rack supplies up to192 kHz high resolution analog I/O converters and a choice (via option cards) of multiple digital formats, including MADI, AES, and ADAT. Users can also select other sample rate options for specific outputs – MADI at 48 kHz for recording feeds, for example.

UB MADI presents another option, feeding a MADI stream in and out of a PC or Mac via USB. Bus-powered, it uses a USB-B type socket and standard cabling, taking up minimal space while providing quality location recording or a virtual sound check system.

In addition, DiGiGrid MGB (coaxial link) and MGO (optical link) interfaces foster plugging in a coaxial MADI-enabled device to Waves SoundGrid for recording, processing and playback of up to 128 audio channels. It can even record to two computers simultaneously. (More here)

Soundcraft Vi Series. The new Vi3000 provides integration into Dante audio networks and access to DAWs for live multi-track recording and virtual sound checks via MADI.

Also included are MIDI, USB and Ethernet ports, along with a DVI output and four channels of AES I/O. And optical MADI interface is fitted as standard, allowing direct connection to a Pro Tools HD recording system via a third-party converter box or any MADI compatible device.

The ADAT card provides two optical 8-channel ADAT inputs and outputs, with selectable 44.1/48/88.2/96 kHz operation. Optical inputs and outputs are provided on Toslink connectors and can be used to record to, for example, a hard disk recorder or other device with ADAT inputs and outputs, as well as receive playback.

In addition, the MADI card offers a simple recording solution for the Vi Series. Additional MADI cards can be fitted by exchanging with other I/O cards. And, both standard and compact stage boxes offer expansion slots for Studer D21m I/O cards, allowing connection to most popular digital formats and also accommodating a MADI recording interface. (More here)

Midas PRO Series. The DL371 processing engine is loaded with four modules for a PRO3, five for a PRO6 and six for a PRO9 in the standard configuration.

The engine has dual-redundant HyperMac ports (both Cat-5e and optical), and there are also eight AES50 ports that facilitate connections to three different stage boxes and/or other AES50 I/O hardware.

It is also possible to use a PRO3, PRO6 or PRO9 with a DL431 stage box, Klark Teknik DN9696 audio recorder (up to 96 tracks), and DN9650 network bridge, which offers the ability to convert the Midas AES50 format to just about any third-party platform. (More here)

Another interesting approach with a Midas PRO2 was presented here by Todd Hartmann, audio engineering coordinator for The Austin (Texas) Stone Community Church.

Devised by Jim Roese of RPM Dynamics, the RPM-TB48 I/O is a stand-alone solution with no external interfaces required, providing a 48-channel, 96-kHz, 24-bit recording/playback solution. It utilizes a pair of Lynx Studio Technology AES50 to PCI cards, all mounted in a Sonnet Thunderbolt chassis.

Because the processor load of the conversion is being handled by the interface, the load on the CPU of the recording computer is very low.

A pair of Neutrik Ethercon cables connects the recording interface to the PRO2 via two of the AES50 ports on the console surface.

Allen & Heath GLD Series. Provides the ability to record and playback a stereo signal on a USB memory stick, and at the other end of the spectrum, standard iLive audio I/O option cards for Dante, MADI, EtherSound and Allen & Heath’s ACE protocols can be fitted to foster multi-channel recording/playback.

For example, M-Dante, M-MADI and M-MMO cards are all available for GLD to enable integration with other systems, including multi-track recording. These cards can be fit to the I/O module expansion slot in a GLD-80 mixer.

The Mini Multi-Out card provides a variety of formats of multi-channel digital output at 48 kHz sampling rate, including ADAT (three optical ports for up to 24 tracks) and iDR (two 8-channel links to the iDR Series installed product range).

Any GLD signal can be patched to any of the 56 outputs for flexible recording. GLD can transport up to 16 signals directly to the iDR-8 and iDR-4 digital mix processors, and also use the 8-channel iDR-out (analog XLR) and iDR-Dout (AES, SPDIF, Toslink digital audio) output expanders for remote feeds. (More here)

SSL Live. MADI I/O connects the SSL Live-Recorder option, a 1RU device that can record 64 tracks at 96 kHz continuously from the console’s input stage and play back the channels in sound check mode. It exports/imports native (.ptf format) projects directly to/from Pro Tools and to/from Apple XML and Steinberg XML.

Connectivity is via standard optical MADI so it can connect over long distances directly to any MADI-equipped digital console, as well as a venue’s audio distribution infrastructure (i.e., Riedel and Optocore), or routers.

The Live-Recorder system consists of a fully configured 1U PC outfitted with a 128-channel SSL MADI audio interface and with Soundscape V6.2 recorder/player software, and it has four front-loading RAID bays pre-fitted with two SSD drives.

It’s connected to the console (directly or via a router) using 2 x 64 channel pptical MADI connections, supports MTC and MMC via MIDI over Ethernet (or any other MTC/MMC-capable USB synchronizer or MIDI interface) for external system transport control, and it can sync via MADI or word clock (via BNC). The software also offers crash recovery routines which will retrieve incomplete audio recordings on reboot after a host system catastrophic failure such as sudden power loss. (More here)

Avid S3L. The open networked architecture and modular nature of this platform presents a high degree of flexibility.

Simply connect a laptop (with Pro Tools or other DAW installed) to the mixer’s Ethernet AVB network (using a single Cat-5e cable) for up to 64 tracks of audio recording/playback.

VENUE Link makes it straightforward to control live mixing and recording/playback setups as one, and users can also perform virtual sound checks globally or on a per-channel basis with the input switch feature, which enables performers to sound check live alongside pre-recorded tracks.

Further, with complete EUCON support, the S3L can be used as a stand-alone control surface to mix sessions recorded in Pro Tools, Logic, Cubase, and other popular DAWs.

And, it can function as a 4 x 4 I/O device for recording in the studio as well as remote locations like hotel rooms or tour buses. (More here)

PreSonus StudioLive AI-series. In a straightforward approach, a pair of bi-directional FireWire s800 (IEEE 1394b) ports connect StudioLive AI consoles to a Mac or PC for recording.

The largest model, the StudioLive 32.4.2AI, has an integrated, bi-directional recording interface that can send up to 48 audio streams to a computer and return up to 34 playback streams (48 x 34/40 x 26/32 x 18 streams available) at 24-bit/44.1/48/88.2 (and 96 kHz support is coming in fall 2014, according the company).

The FireWire s800 and Ethernet ports come on a preinstalled card that is user-replaceable with optional Dante, AVB, or Thunderbolt cards.

Designed specifically for StudioLive mixers, Capture 2.1 software adds proprietary Active Integration networking, offering fast setup and recording directly from the mixer, with auto configuration.

It also provides convenient, automated virtual sound check. (More here)

Roland V-Mixer Series. Enables three types of recording and playback solutions: onboard stereo recording via USB port, integrated multi-channel recording and playback via the company’s R-1000 48-track recorder/player, and integrated multi-channel recording using the proprietary REAC platform.

The R-1000 can be used with any REAC digital snake as well as with any digital console with MADI output capabilities by using the Roland S-MADI REAC MADI bridge.

The REAC driver kit is intended for use with all V-Mixer consoles is also compatible with Roland S-4000S, S-2416, S-1608, S-0808, and S-MADI digital snake systems.

In addition, up to 40 channels from a V-Mixing console or digital snake can be routed directly into most ASIO-based DAWs via Cat-5e/6 connected directly to the gigabit network port on a PC. (More here)

QSC TouchMix. Yes, they’re a bit on the smaller scale for the purpose of this discussion, but we wanted to point out that these new miniscule mixers are capable of direct recording to an external USB hard drive – no external computer is required. All inputs plus a stereo mix are created in 32-bit WAV format. Tracks can also be played back on the mixer or imported into most DAW software for over-dubs and post production. (More here)

Mackie DL Series. Also very compact in form factor, these iPad mixers provide the ability to record stereo tracks directly to the iPad. It probably doesn’t get any easier than that. (More here)

{extended}
Posted by Keith Clark on 07/06 at 06:30 AM
Live SoundFeatureBlogConsolesDigital Audio WorkstationsInterconnectMixerNetworkingProcessorSoftwarePermalink

Thursday, July 03, 2014

Care & Feeding: Keeping Gear In Top-Flight Shape

To get the most mileage out of gear, regular equipment inspections and Preventative Maintenance (a.k.a., PM) are a must.

All equipment in your inventory should have PM scheduled at least once a year, and more frequently if it goes out the shop door a lot and/or is exposed to harsh environments.

PM comes down to inspecting, testing, cleaning, lubricating and repairing to keep systems in top operating condition.

In addition to annual PM, all gear should be given a quick inspection during setup and tear down at every gig. This includes a visual inspection, placing a hand on equipment to feel operating temperature, tugging on cable ends to see if strain relief is in good shape, etc.

If irregularities are noted, further inspection should be performed and problems addressed. Not paying attention to small problems allows them to build up to big problems that are much more expensive to correct, and they can also result in a failed gig. Here I’ll share some of the PM approaches I regularly utilize with my own gear.

Electrical

PM for electrical gear like processors, amplifiers, and snake boxes always starts with a complete visual examination. Each unit’s case is opened up for visual inspection of the interior. I’m looking for loose or broken wires, unseated connectors, blown fuses, discolored circuit boards, and so on.

Keeping the inside of components this pristine can only help performance and longevity – just be sure to check the manual before removing cases. Image courtesy of QSC Audio. (click to enlarge)

While the case is open, it’s a great opportunity to run a vacuum and clean out all dust and road gunk that has accumulated inside. Sometimes an air compressor, or at least some “canned air,” is used to blow out the dirt. I also remove filters and clean or replace them per the manufacturer’s instructions.

Next up is checking and cleaning signal connections. If the equipment has faders and knobs, it’s time for cleaning and lubrication (again, per the manufacturer’s recommendations).

All electrical pins and connection surfaces are evaluated for corrosion and misalignment, and input and output connectors are given a thorough cleaning with an electronic cleaner such as Deoxit from Caig Labs. If connectors need to be repaired or replaced, this is the time to do it.

With the case still open, it’s a good ideal to double check all power cable connections, and if the unit has a fixed power cord, to make sure the strain relief is in good shape and the cord has no cuts or tears in the outer jacket. I also run my hand down the cable to feel for internal cable damage. If the unit takes batteries, they get a check, and the battery terminals are cleaned.

Before plugging in and powering anything, I make sure all cleaning fluids or solvents have dried. After a quick check to make sure the equipment is operating correctly, each component is sealed back within its case.

Rack-mount gear is a little harder to access without removing from the rack, but I strongly believe that doing maintenance is so important it’s worth the trouble. Note, however, that opening up some gear may void the factory warranty, so please read and follow all manufacturer instructions on maintenance.

Microphones

Modern microphones are pretty robust and usually don’t require a lot of attention, but they should be inspected after each use because they’re regularly dropped, exposed to liquids, etc.

Because the majority of my jobs are corporate gigs, which are usually relatively tame, I only do serious mic maintenance once a year.

But for those doing outdoor festivals and/or working with much more “raucous” forms of entertainment, there could be need to do maintenance as often as every month.

Many models allow you to remove a damaged grille/head to simply screw on a new one.

Factory replacement heads are usually available, and a few companies also make generic heads that fit popular microphone models.

And sometimes they can be fixed. For round ball-shaped grills, the handle of a large screwdriver can be used to gently pressure out dents. If a dent is a little stubborn, I place the ball on top of a folded towel and tap the screwdriver with a small wooden mallet.

Sometimes mic grills can be “helped” back into shape, or they might need replacement. Also keep an eye on the connectors, which can be subject to abuse. (click to enlarge)

Before grills are re-attached, they should be cleaned with a mix of dish soap and warm water, with a soft bristled toothbrush to help scrub out the dirt. Some folks use Listerine for cleaning, and there’s a foam-based cleaner called Microphome available as well.

Inner foam windscreens can be replaced or washed in a mix of dish soap and warm water. These should be wrung out and air dried completely before being reinstalled. For mics that don’t have removable grills, I use a dry soft bristled brush on the exterior of the grille to remove dirt and then hold the mic upside down to help loose dirt and debris fall away.

Don’t leave batteries inside mics between shows because they can leak and corrode the contacts and generally ruin the electronics. To keep these terminals (as well as mic connectors) clean, I use Deoxit, then wipe them dry with a clean cloth.

Don’t forget the clips! Mic clips should be checked for signs of cracks and missing pieces. Also evaluate the threads and the tightness of the swivel. I normally place a drop of light lubricating oil or WD40 on the threads so they’ll screw easier on to mic stands.

Loudspeakers

Safety is more important than looks or sound, so the first thing I check on loudspeaker cabinets is the rigging, making sure nothing is cracked, bent or distorted. All moving parts should be cleaned and lubricated per the manufacturer’s recommendations.

Also don’t forget to keep an eye on external hardware like handles, corners and grills, fixing anything that requires attention.

Make sure hardware like corner protectors and handles stay firmly attached. (click to enlarge)

Connectors (and their panels) should always get attention as well, to make sure they’re intact and secure. For powered loudspeakers, give the power cord and amplifier a check before testing out the box.

During down times, I power up boxes and run a sweep tone through them to insure that drivers and crossover (if applicable) are O.K. For subwoofers, I usually run a kick drum sound from a drum machine as a general test, in addition to evaluating frequency tones.

Cables

Without cable and interconnect, a PA system is just a bunch of unconnected gear.

Yet cables seem to get the least attention – until they don’t work. After every usage, cables should be checked.

At the start of the wrap process, give the connector at that end the once over, to see if any pins or contacts are corroded or bent, and to confirm that the connector body is in good condition.

Make sure strain relief is tight and that the cable jacket has not pulled out of the connector body.

Then during the wrap, slide a hand along the cable, feeling for flat spots, twists or other irregularities inside the jacket. Check the outer jacket for cuts or tears.

At the end of the process, check out the other connector, then secure the cable and lay it in the proper storage case. (Don’t forget to also do this with AC extension cords.)

Cables that are obviously damaged or that need another look should be set aside. A common practice to mark a suspect cable is to put a half-hitch knot on each end, warning others not to use until it gets checked out. Another tactic is to place pieces of gaff tape over the connector ends.

It really only takes an extra second or two per cable to check them as they’re wrapped, but the extra seconds spent can save minutes (or hours) of chasing down problems at the next gig.

Check cables for obvious damage (such as that shown here), as well as problems under the surface. (click to enlarge)

All cables should also get a more extensive yearly check and some PM, including signal check with a cable tester and a thorough cleaning. When using a cable tester, check for intermittent signals by wiggling the cable where it joins the connector, and also flex the cable at any suspect spots to see if there is a break.

Many times a cable may have a break in one or more of the conductors, but the problem won’t rear its head until it’s flexed or wiggled.

With analog snakes, check the strain reliefs, and also open up the stage boxes to check the internal connections. Clean and lubricate snake reels per the maker’s instructions, and ditto for both fiber optic cabling and reels.

For general cleaning of outer cable jackets, I use a cleaner/degreaser called Simple Green. For removing sticky taperesidue (and this applies to other gear as well), I turn toGoo Gone, a Citrus-based cleaner.

When that won’t cut it, I switch to a stronger solvent called Goof Off, which contains acetone, so caution is strongly advised. It will eat through many materials, so just use enough to get rid of the gunk in the affected area, and then thoroughly wash the area clean of any remaining solvent.

Stands

Ubiquitous and ever supporting, stands are often forgotten about until something breaks. Mechanical stands need maintenance just as much as sound reinforcement equipment.

On mic stands, check the clutch regularly to make sure it operates smoothly. Replacement parts are available from manufacturers to rebuild a loose clutch mechanism. I also remove gaff tape residue (Goo Gone or Goof Off ), dry the tubes with a rag, then work a few drops of oil into the end threads so they screw into the bases and clips easily.

Staples of the PM kit include Deoxit, WD40, Goo Gone and perhaps some Microphome to keep mics fresh. (click to enlarge)

If I spot any damaged threads, I “chase” them (running a thread cutting die over a section to try to repair the threads) or simply cut off the end with a pipe cutting tool and make new threads on the fresh section of pipe.

Evaluate the stand’s base and replace any rubber isolation feet as needed. For tripod stands, check the legs and lubricate the hinge joint with a silicone- or Teflon-based lubricant. For loudspeaker stands, the process is similar, and also address rivets that hold the leg hinges as well as any safety stops on the stands.

Last month I focused on racks and cases (here), and these items also require scheduled maintenance. Check all hardware in general, and make sure rack rails are firmly bolted to the rack shell. Clean and lubricate the handles and hasps with a silicone or teflon lubricant, and clean and grease the castors as per the caster manufacturer’s recommendations.

While it may seem like a large outlay of effort, keeping up with regular PM can reduce problems, enhance system performance, and save significant time and money in the long run.

Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb, and is the owner of Tech Works, a production company based in Las Vegas.

{extended}
Posted by Keith Clark on 07/03 at 08:37 AM
Live SoundFeatureBlogStudy HallProductionAudioAmplifierBusinessInterconnectLoudspeakerMicrophoneProcessorPermalink

Wednesday, July 02, 2014

Details, Details: Setting Up Snake Channel 24

The attention to detail that takes place in preparing a rock show can be mind boggling. For example, I listed out the factors we account for in setting up the lead vocal mic for the Red Hot Chili Peppers.

Check it out:

1) The mic. Anthony has been using Audix OM7 dynamic mics for over 20 years now. The OM7 exhibits high feedback stability and picks up very little room sound compared to other mics. This allows me to capture an “up close and personal” vocal sound, and if I need more air or space, it’s easy to add with a vocal reverb. Also, these mics are very durable, and the spring steel grills don’t dent when dropped.

Conversely, the OM7 pick-up pattern falls off in volume very quickly if you’re not lips to grill on the mic. Also, it tends to be more susceptible to moisture than other microphones. Since Anthony sings close to the mic and we switch to a fresh mic at mid-show and again before encore, those drawbacks are not an issue in this application.

2) The mic stand is integral to the performance so it must be exact. We use the Atlas MS12 – a straight stand, no boom. The “12” stands for 12 pounds, and that is the weight he’s used to swinging around. These stands have a slightly larger diameter tube than the Euro manufactured metric stands and are less likely to bend. The metal clutch is more durable than the plastic clutches, and the larger diameter cast metal base makes it less likely to tip over as well.

3) The cable must be extremely durable. We use Belden 8412 or equivalent with real rubber jacketing (not plastic), braid shield, multiple fiber wraps and preferably a twine filler. It must have minimal stretch, withstand abuse and be resistant to tangling.

4) The clip. We use a Shure SM58 clip because the Audix clip is too rubbery. It’s the older version clip that is not flared up top. It requires the mic to be slid – not popped – into place, and the mic won’t jump out when the stand is swung or bounced. Also, we heat up new clips with a lighter and bend them open so they get the proper grip on the mic – not too tight or loose.

5) Taping the cable. The mic cable is taped to the mic to prevent it from being accidentally unplugged. This is done with a single non-overlapping layer of black gaff tape. It’s critical that the mic (and tape) slide out of the clip not too easily but also not jamming either.

6) Taping the stand. A ring of gaff tape is wrapped around the inner pole to prevent the stand from getting shorter should he bang it on the ground in a downward direction. Each stand is measured to be exactly 55.75 inches in height, base to thread ring. There are always at least two spare built-and-measured stands as backups. We go through dozens over the course of a tour.

7) Boring out the clip. The stock old-school Shure clip has a sharp-ish edge that prevents the taped mic from sliding out smoothly. To solve this, the rear sharp of each mic clip is beveled and rounded out with a Leatherman tool or any sharp knife and checked for “slide.”

8) Clip rotation tension. The clip needs to hold the mic firmly at an angle and not loosen easily. Cheaper imitation clips do not have the tension washer stacks inside and will come loose when repeatedly rotated.

9) Cable length. The mic cable is 50 feet long and plugged into a stagebox located center stage.

10) Spare mic. An identical spare taped vocal mic is coiled at center stage. The main mic, spare, and a wireless mic all are plugged into a three-way SoundTools switcher located at the monitor position. This allows Anthony to grab any of the three mics and have it instantly switched in line to both monitors and house.

There you have it – now you know how we prep snake channel 24!

Dave Rat (www.daverat.com) heads up Rat Sound Systems Inc., based in Southern California, and has also been a mix engineer for more than 25 years.

Read more Dave Rat articles on PSW here.

{extended}
Posted by Keith Clark on 07/02 at 03:08 PM
Live SoundFeatureBlogStudy HallConcertEngineerInterconnectMicrophoneSignalSound ReinforcementStagePermalink

Friday, June 27, 2014

Berklee College of Music Embraces Avid In Groundbreaking New Music Studio Complex

Enables real-time collaboration on projects between new facility in Boston and campus in Valencia, Spain

Berklee College of Music, the world’s largest independent contemporary music college, has deployed Avid professional audio production solutions to help enable real-time, high-definition collaborative workflows between its newly unveiled 16-story tower in Boston and its campus in Valencia, Spain.

An ultra-high speed internet connection links Berklee’s 10-studio audio production complex in Boston, which is among the largest of its kind in the United States, to the campus in Valencia, enabling real-time collaboration on a global level – now a standard practice in many professional project workflows. For example, musicians playing on one campus can be recorded and mixed by students at the other campus. 

“One of our founding philosophies is that students need practical, professional skills for successful, sustainable music careers,” states David Mash, senior vice president for innovation, strategy and technology at Berklee College of Music, and chairman of the executive board of directors for the Avid Customer Association. ”With Avid audio solutions, we can provide access to industry-standard tools that will enable them to excel in the professional world. The Avid Everywhere vision complements our commitment to giving students real-world experience with collaborative workflows on a global scale.”

For both campuses, Berklee has chosen audio solutions that are integrated with the new Avid MediaCentral Platform, giving students access to an even wider range of tools and experts, from music creation to distribution. These include Pro Tools|Software, Pro Tools|HD systems with Avid analog and digital HD interfaces, Sibelius music notation software, and two System 5 digital audio mixing consoles.

Berklee has also selected Pro Tools as the official digital audio workstation (DAW) on all laptops distributed to every incoming student as part of its Berklee Bundle Licensing Program (BBLP), fulfilling its commitment to giving students unlimited access to industry-standard tools.

“As the world’s premier music learning lab, Berklee plays a critical role in the industry by developing the music professionals of tomorrow,” states Jennifer Smith, senior vice president and CMO, Avid. “By embracing change, anticipating trends, advancing its curriculum, and adopting cutting-edge technology, Berklee sets a new standard for music education, and gives students hands-on experience with the same tools that audio professionals use to create award-winning music.”

Avid

{extended}
Posted by Keith Clark on 06/27 at 04:12 PM
Live SoundRecordingNewsDigital Audio WorkstationsEthernetInterconnectMixerNetworkingProcessorSoftwareStudioPermalink

MediaMatrix Announces Claro 12-In x 8-Out Digital Processor

Each of 12 balanced inputs can be configured as a microphone or line input, including phantom power,

MediaMatrix has introduced the new Claro processor, a single rack unit frame that provides 12 balanced inputs, each of which can be configured as a microphone or line input, including phantom power, as well as 8 line outputs. Inputs and outputs are connected via Phoenix Euroblock connectors.

Each input has its own dedicated Acoustic Echo Cancellation (AEC) processor, allowing up to 12 mics to be managed during a teleconference. The Claro also supports VoIP and POTS telephony through its telephone hybrid interface.

A range of control options are also provided. Claro’s rear panel provides 12 GPIO connections, RS-485 and RS-232 connections, and multiple Ethernet connections for interconnection and control. It also contains a complete range of DSP mixing, routing and processing functions, such as EQ, filters, delays, compressor/limiters, meters and test signal generators, all accessible and controllable via an external GUI.

Claro scales across multiple meeting rooms, courtrooms or classrooms, as up to four Claro frames can be interconnected via its audio expansion interface to extend its capacity to 48 inputs and 32 outputs. This allows plenty of capacity to share audio and control across larger conference installations. A front panel USB connection provides access to the processing and control functions or to play back content during conferences.

Claro overview:

• 12 balanced inputs, each configurable as mic or line inputs, via Phoenix connectors
• Acoustic Echo Cancellation and phantom power available on all 12 inputs
• 8 balanced outputs, via Phoenix connectors
• VoIP (SIP) and POTS support via dual RJ11 connectors
• 12 GPIO connections
• Control via RS-232, RS-485 or Ethernet
• Audio Expansion Interface allows up to four Claro frames to be connected via Ethernet
• Full-featured DSP capabilities, including EQ, Filters, Mixers, Delays, Compressor/Limiters, AGC, Meter, Signal Generators and AEC
• Front panel USB connection
• Single rack space

image

 

MediaMatrix

{extended}
Posted by Keith Clark on 06/27 at 10:04 AM
AVLive SoundChurch SoundNewsProductAVDigitalEthernetInterconnectNetworkingProcessorPermalink

Thursday, June 26, 2014

Altinex Introduces TNP155/TNP155S Tilt ‘N Plug Jr. Interconnect Boxes

Easy access to AC power and charging USB

Altinex has announced the new TNP155 and TNP155S Tilt ‘N Plug Jr. interconnect boxes, providing one-touch access to dual AC power receptacles plus two charging USB ports and available in both standard and custom configurations.

As a tabletop connection device, the Altinex TNP155 (brushed black finish) and TNP155S (clear brushed aluminum) are highly functional tools for accessing conventional AC power outlets for the purpose of powering, for example, a notebook PC. Similarly, the dual USB ports are capable of charging smartphones, tablets, and similar USB devices. Further, the boxes can be customized with a wide variety of custom snap-in assemblies.

The input plate of the TNP155/TNP155S is accessed by pushing down on the top cover. The unit then auto-tilts open with assistance from an internal pneumatic spring. Once open, the input plate remains securely in place. The input plate is hidden, or closed, by pressing down on the top cover until the latching mechanism engages.

In its closed position, the top panel lies flush with the table’s top, held in place by the latching mechanism. This secure fit also means less chance for paperwork to catch the TNP155’s edges when being passed across a table.

“Our TNP155 and TNP155S Tilt ‘N Plug Jr. interconnect boxes are the ideal choice for creating a quick and convenient means of powering a wide range of devices typically used in today’s business and education environments,” says Grant Cossey, Altinex VP of sales. “With two AC power outlets, meeting attendees can easily power their laptop computers while those people with smartphones and tablets can charge their devices without having to place their valuable equipment on the floor—where it can easily be stepped on and damaged. With the quick, easy access these units afford combined with their elegant design; the TNP155 and TNP155S create a high-tech visual aesthetic that compliments the décor of any meeting or presentation space.”

The Altinex TNP155 and TNP155S interconnect boxes ship with 6-foot USB cables, 9-10 foot power cables, and a 5-volt, 5-port USB charging unit to power the USB ports from underneath the table. Both the TNP155 and TNP155S carry an MSRP of $325, and will be available in late June.

Altinex

{extended}
Posted by Keith Clark on 06/26 at 06:59 AM
AVLive SoundChurch SoundNewsProductAVEthernetInstallationInterconnectPowerSignalPermalink
Page 50 of 145 pages « First  <  48 49 50 51 52 >  Last »