Tuesday, April 12, 2011
Electro-Voice Announces RE320 Wired Microphone At NAB 2011
The RE320 is based upon the features of the legendary RE20 and RE27N/D broadcast microphones.
Electro-Voice has announced the global launch of the RE320 at NAB 2011, the latest addition to the Electro-Voice RE Performance Group of wired microphones.
The RE320 culls key features of the venerable RE20 and RE27N/D Broadcast Group products into an exceptionally versatile studio and live performance mic.
Engineered for performance in virtually any imaginable application, the RE320 delivers excellent results when mic’ing vocals or instruments in the studio or on stage.
The RE320’s key features include Electro-Voice’s Variable-D proximity control technology; their humbucking coil for noiseless operation; and a high-output neodymium magnet structure capsule that delivers fast and accurate transient response and pronounced high-frequency detail.
Rounding out the feature set is the new “dual personality” switch which essentially creates two mics in one. One setting of the switch engages a response curve that is ideal for voice and most instrument micing, while the other switch position activates a response curve designed specifically for kick drums with dips and peaks exactly where kick drums need some attention.
“We’ve known for many years that the RE20 is sought-after for mic’ing kick drums, despite its broadcast vocal ‘original purpose,’ said Electro-Voice’s Rick Belt comments.
“Adding the kick drum curve switch position as a key RE320 feature reinforces its usefulness as a specialty instrument mic, in addition to its outstanding performance on voice and low-mid to upper-register instruments in the alternate switch position.”
The RE320 will be available at an MSRP of $499.95.
Factory Direct: A Look At The Powersoft Armonía Pro Audio Suite
Advanced power amplifier control software.
Gone are the days when a so-called dumb, brute force power amplifier was the simple connection between front-of-house electronics and the loudspeaker system.
We’re now in the age of the smart amplifier that is both software and audio driven, creating a close marriage between audio control software and the power amplifier.
In short, the power amplifier has now become a technology platform capable of supplanting many other pieces of gear that were stand-alone units like crossovers and room-tuning EQ units.
To meet the challenges of this union, Powersoft has created PC-based Armonía Pro Audio Suite advanced control software to manage its K, Q, D, Duecanali and QTU1400 Series amplifiers.
Armonía provides unique capabilities that allow system engineers to efficiently maximize the interaction between the amplifier and loudspeaker, while providing programmable control of key features like damping, EQ, limiting and crossover.
The full expression of Armonía is available to Powersoft amplifiers equipped with DSP and AESOP (AES3 and Ethernet Simple Open Protocol) cards.
But to develop Armonía, we needed to consider the real-world aspects of designing, implementing and running a sound system for touring or installation.
Back to Bass-ics
Ask any engineer, artist, show producer or audience member about the sound at an event and the bass response is usually at the forefront of comments.
Delivering tight, well defined bass frequencies is of paramount importance to the sonic success of an event. We have incorporated our patented Active DampingControl feature into Armonía for precise tuning of system damping for loudspeakers reproducing frequencies under 400 Hz.
Active DampingControl for maximizing damping parameters.
The value expressed for damping describes the ability of the amplifier to suppress undesirable movement of a cone near the loudspeaker’s resonant frequency.
A loudspeaker’s diaphragm has mass and the cone rigidity, the combination of which forms a resonant system. A loudspeaker also generates electric current from its electromagnetic voice coil moving within the magnetic field of the natural magnet structure.
When audio frequencies are applied to a loudspeaker, excess motion can be generated at or near the resonant frequency. So, the greater the damping number, the more control.
This complex relationship between amplifier and loudspeaker relative to frequency is not the only problem.
The effect of cable resistance on a high power amplifier’s performance also becomes significant at low frequencies, as the cable resistance can affect the output stage’s damping factor. As each installation requires different cable runs, the damping numbers can vary greatly.
Active DampingControl is a proprietary algorithm which compensates for cable resistance. This means cable parameters may be entered in either Metric or American Imperial units for cable run length and wire gauge.
Armonía then sets the cable resistance value and the algorithm calculates and applies the appropriate correction.
System engineers can now precisely tune each different setup based on the physical realities of a particular installation, yielding consistent sonic results from a particular amplifier/loudspeaker combination used over various events.
Impedance, Alive & Well
The LiveImpedance feature for Armonía provides a graphic display of instantaneous load impedance against frequency for each channel. This gives the engineer a constant update on loudspeaker and system performance over time as a function of power transfer.
The instantaneous impedance is calculated from snapshot measurements of output voltage and current.
Real-world amplifier loads like loudspeakers have a complex impedance relationship against frequency and power, which means that the impedance possesses both resistive and reactive components.
Because the components are in quadrature with each other, typically separated in phase by 90 degrees, the impedance has both magnitude and phase.
Both quantities are calculated and displayed by LiveImpedance for constant monitoring by an engineer.
LiveImpedance shows instantaneous load impedance against frequency.
With the understanding that an amplifier’s output stage might be constrained by filter assignments from handling the entire audio frequency spectrum or the audio program may not contain certain frequencies at the instant of snapshot, the display may not be a complete, full spectrum plot at any given moment in time and setup.
We color coded Armonía’s displays to show the magnitude of the impedance in ohms and the phase angle of the load in degrees. The default impedance scale is from 0 to 100 ohms and the default phase scale is ±90 degrees.
The magnitude and phase buttons open drop-down lists of alternatives where the impedance scale may be reset to 0 to 25 or 0 to 50 ohms, and the phase scale to ±180 degrees. While the default impedance scale is linear, a logarithmic alternative may be selected to improve the resolution in the particularly important 1 to 10 ohms range.
Apart from displaying the complex impedance interaction, the LiveImpedance graphs can also display reference data derived from the amplifier’s normal (non-DSP) output impedance measurements and imported “ideal” impedance plots from speaker manufacturers.
All these different graphic representations make the engineer’s task of monitoring high output levels of multiple amplifiers over long periods very simple and further enhance an engineer’s understanding of the relationship between the power amplifier and a particular loudspeaker at a live event.
A third complicating factor we needed to address with Armonía in the relationship between the power amplifier and loudspeaker is optimizing power transfer to a speaker.
To accomplish this, the TruePower limiter feature yields controlled operation for longer driver life by maintaining safe output power levels referenced to frequency and true load impedance. Armonía’s power limiter section allows the user to select one of two methods of power calculation: TruePower and Equivalent Power at 8 ohms.
TruePower is the preferred setting to choose when the number of loudspeakers connected to the amplifier channel is a known quantity.
TruePower limiting fosters controlled operation for longer loudspeaker driver life.
It is based on averaging real-time measurements of output voltage and current to define the limiter’s intervention point for power limiting in order to stay within a speaker’s safe operational range.
With this information, Armonía is able to estimate the instantaneous power available at the amplifier terminals. TruePower also features a peak limiting section to cover all the signal bases.
Choosing the Equivalent Power at 8 ohms setting will result in all power calculations being based on an arbitrary 8-ohm load.
The output voltage is still measured and from that information the output current and power can be derived. Working essentially as an RMS voltage limiter, this mode is useful when the number of loudspeakers connected to the amplifier terminal is unknown.
Modeling In Fashion
Armonía also offers an amplifier modeling feature that can address amplifier DSP and routing online or offline.
In offline mode, Armonía lets a systems engineer plan out an intended system of any size at the office (or even on a tour bus or plane) using virtual amplifiers for programming before connecting to a network.
This is very useful to the engineer to match amplifiers with appropriate DSP functions like crossover and EQ to loudspeakers before the system is on the truck. This also gives a clear picture of system resources needed for a particular event.
When used in online mode, system information can be sent or retrieved from the synchronized network. From here it is possible to copy an amplifier’s presets to the PC running Armonía and vice versa, including the very useful option of copying a set of presets to multiple amplifiers simultaneously.
All parameters designed in offline mode can be sent to the online actual amplifiers and, when used with the optional Powersoft Ethernet KAESOP board, a system engineer can extend the network to a virtually unlimited number of amplifiers through either auto-addressing with self-assigned IP addresses or end-user configurable IP addresses.
Making The Connection
Armonía is an Ethernet-based software package that offers full control of all amplifier parameters while accommodating AES3 audio on the same cable.
The software has built-in redundancy and extensive logging and alerting features designed to provide worry free operation for the sound engineer.
The platform delivers control and monitoring of all amplifier functions including AC mains current draw, headroom, protections and faults. The software also includes advanced grouping options.
Armonía is also compatible with both the latest offerings and many of our legacy products. Powersoft amplifiers can connect to a PC running Armonía in two ways: through an RS-485 serial connection or via Ethernet.
Amplifiers of the K and Duecanali Series can be equipped with either or both methods of connectivity while earlier remotely-controllable models have only an RS-485 connection. A system can be configured using both connection pathways simultaneously, extending the functional life of older amplifiers.
With an optional Ethernet KAESOP board, a network can be extended to a virtually unlimited number of amplifiers.
The range of network topologies which can be used in wiring a real system varies between the two communications methods. Ethernet provides a slightly greater degree of freedom, as standard IT network switches may be used to create multiple hub-and-spoke systems.
A looped Ethernet topology is also permissible, which will provide redundancy in the event of a network failure. An amplifier system using an RS-485 network can either be daisy-chained throughout or use the Powersoft PowerHub as a local switch.
Further, a wireless access point can be made directly to any Ethernet-enabled Powersoft amplifier, allowing wireless control via laptop or tablet PC.
Powersoft employs the IEEE 802.11n standard that currently yields faster, more reliable connectivity, extended usable range, plus offers an increase in available channels over earlier versions. This makes 802.11n far more suitable for use in fixed and tour sound applications where multiple amplifiers need to be addressed.
Ethernet networking also assures faster data refreshing and data exchange between each amplifier and the host computer, delivering real time monitoring of amplifier status, remote connection, mains presence, current draw, input signal, output voltage, internal temperature, short-circuit protection, clip and fault.
Armonía also offers real time control of amplifier on/ off/stand-by levels, solo, mute, sensitivity, max output voltage, max mains current draw, presets locking and input selection (digital/analog/KAESOP).
The final area of focus in the development of Armonía was in providing the capability to program and implement extensive DSP functions affecting both input and output signals.
For those amplifiers equipped with optional DSP cards, a greater range of features and facilities were made available including input and output equalization, alignment delays and FIR filters.
We developed the features list through the input of top industry engineers and system designers to address the real needs of delivering and implementing a sound system, keeping in mind that the amplifier itself was taking over the role previously held by external pieces of equipment.
Equalization was implemented to cover all aspects of system equalization.
As such, Armonía makes available up to three EQ layers of 32 filters each in the input EQ, as well as 16 EQ filters plus two crossover filters in the output EQ. In addition, DSP control was further enhanced by Armonía’s grouping feature which combines input EQ, input delay and input attenuation DSP parameters within a graphic interface that is easy to use.
Preset protection was assured through a multilevel locking feature with invisibility of parameters. The input EQ was designed to be used as either parametric or graphic, depending on the intended mission of the EQ.
Armonía’s DSP remote control also has the capability of importing frequency response curves as ASCII data from the most popular measuring software for an easy re-design of loudspeaker presets on location.
Armonía’s display and graphical interface can be customized according to individual needs.
While the output EQ has a similar general appearance of the input EQ, it has several important differences.
The output EQ curve the amplifier employs was intended to be created from a speaker manufacturers’ data for the particular loudspeaker cabinets or arrays used in a system. It provides a range of hi-pass and low-pass crossover filters as well as 16 individual peaking filters.
It is also possible to import and export crossover/filter response curves. The available filters for crossover include Butterworth, Bessel, Linkwitz-Riley, FIR and Hybrid FIR filters with the turnover frequency fully adjustable for all types.
The filter slope was made selectable in 6dB/oct increments from 6 dB/oct to 48 dB/oct for Butterworth and Bessel types, and in 12 dB/oct increments from 12 dB/oct to 48 dB/oct for Linkwitz-Riley types. FIR and Hybrid FIR filters have slopes which are defined by the turnover frequency, so dedicated slope adjustment parameters are unavailable.
Signal delay capabilities allow the engineer to precisely tune the correct time delay for side fill, under balcony or any speakers that are set up beyond the main front of house system.
The delay functions can insert a per-channel fixed delay in the output equalizer as well as in the input equalizer. The primary use of delays added here is to time-align individual drivers in multi-way loudspeaker systems.
Delay values can be specified as time, in 0.01 millisecond (mS) increments up to 32 mS, or as distance in 1 millimeter increments up to 35-plus feet (10.9 meters).
The distance units may be changed to feet by selecting Imperial instead of International. The distance/ time conversion is based on a sound velocity value of 1,125 feet per second (343 meters per second) in dry air at 68 degrees F (20 degrees C).
The design challenges in creating an amplifier capable of becoming a technology platform controlled by intuitive, feature laden software were legion.
The result was the integration of real-world issues experienced with large sound systems, coupled with hard science, to yield a comprehensive software package.
Claudio Lastrucci is managing director of Powersoft.
Auralex Introduces Cloth Wrapped Foam Panels At NAB 2011
The company is expanding its customer base with the new SonoLite professional grade foam panels.
Auralex Acoustics, Inc. has announced the introduction of a ground-breaking addition to its line of products at NAB 2011; the new SonoLite fabric wrapped Studiofoam Pro panels, with a price that specifically targets those on limited budgets.
The latest entry-level SonoLite fabric wrapped StudiofoamPro panels, now allow the company to expand its customer base by reaching everyday musicians who want to treat a variety of studio spaces.
“Auralex is thrilled to offer this type of product at such an affordable price to both our loyal customers and to those who may be new to Auralex products,” says Eric Smith, founder and president, Auralex Acoustics.
“Auralex’s SonoLite is an ideal acoustical absorption panel for the home-based musician. SonoLite is an aesthetic and price point blend of Auralex’s StudiofoamPro and ELiTE ProPanels, combining the look of the ELiTE ProPanel with StudiofoamPro’s cost.”
SonoLite is a 2′ x 2′ x 1″ fabric wrapped StudiofoamPro panel, available in black or beige, with squared edges that provides an overall Noise Coefficient Rating (NRC) of 0.75.
The new product will be offered at a retail price of $24.99 per panel.
Auralex Acoustics Website
Are Headphones A Viable Tool For Monitoring Your Mix?
Even in a less-than-ideal mixing environment, you can get good mixes if you know how to compensate for problems your system brings to the table. Depending on your situation, headphones may be for you.
A question I’m often asked is “Do You Mix With Headphones”, which I think is something worth discussing.
To start, here are a few comments I’ve received in response to this very question:
I cater to the most popular form of listening. So far, I’ve found that mixing to headphones and then listening on speakers has worked. I’m not Rick Rubin, but neither is anyone else who is not Rick Rubin.
I do the main mix with Sony MDR-CD180 headphones, while checking with iPod buds, little Logitech laptop speakers and finally in my car. Between those, I can pretty much get it in the ballpark.
I must be doing something right – on my last CD, even my most pickiest of listeners actually commented on how good it sounded. (excuse while I break my arm trying to pat myself on the back)
Those both made me laugh out loud. So, Let’s talk about mixing with headphones.
1. Less-than-ideal Mixing Environment
Most people that use headphones do so out of necessity. If they had a properly treated acoustic environment with nice studio monitors, they would likely use those.
But since they don’t have a great mix room, they revert to headphones.
I’ve talked before about acoustic treatment in your studio. It’s absolutely a necessity for both recording and mixing. However, some people just can’t afford to properly treat their entire room.
They may only have enough money to treat a portion of the room to allow them to get a nice, clean recording.
When it comes to mixing, though, frequencies are flying all around the room. There are huge peaks and dips in the frequency response of the room itself. (My room, for example, has some serious issues in the 120-160 Hz range.)
All this craziness can make it very hard to get consistently good mixes. Acoustic treatment will help “flatten out” the frequency response of the room.
Headphones, on the other hand, don’t need acoustic treatment. They sound the same every time.
2. Increased Detail
Most people would agree that you can hear more detail on headphones than on studio monitors.
I always use headphones for editing, for example. I want to make sure I don’t miss any pops or clicks in cross-fades, etc.
When mixing, headphones can give you an added amount of detail with things like EQ, compression, panning, effects (reverbs, delays), level balance, etc.
3. Keeping Things Quiet
Many of us work with other people, or in less than ideal conditions. Also, many of us have day jobs, which makes “studio time” synonymous with “late nights.”
When I first got the studio monitors I have now, I was living in an apartment, and I could only work on music at night. I was so bummed, because I never had a chance to try out the monitors, since they would wake the neighbors.
Obviously there are some definite reasons to use headphones, so lets examine the down-sides.
1. Limited or Exaggerated Frequency Response
One of the reasons we get big 6-inch or 8-inch studio monitors is so we can actually hear what’s happening in the low end.
Headphones typically cannot reproduce the lows the same way that studio monitors can. After all, they’re small little mini-speakers, so we can’t expect them to thump like a 12-inch subwoofer.
Sometimes headphone manufacturers make up for this by boosting the low end in their headphones. This isn’t necessarily wrong, but you need to keep this in mind when mixing on headphones.
Since they probably don’t have a super-flat response, you need to know what the headphones are doing to your mix as you make your mix decisions.
2. Altered Stereo Image
When you’re mixing on studio monitors, when you pan something hard right, you’re still going to hear it in your left ear. The sound will travel from the right monitor, past your face, and into your left ear.
With headphones, this doesn’t happen. If you pan something hard right, it’s only playing in your right ear. This isn’t necessarily bad, but you might choose to pan things differently when mixing on monitors vs. headphones.
Also, the “center” of your mix is very different on headphones. When listening on monitors, anything panned to the center sounds like it is in front of you, between the monitors.
With headphones, anything panned to the center sounds like it’s in the middle of your brain. This may seem like a non-issue, but it can effect how you handle things like lead vocals, bass, kick, snare…anything panned to the center.
The lead vocal might sound great on monitors but way too loud in the headphones. Gotta find a balance somehow.
3. Lack of the “Wow” Factor
There’s something awesome about listening to a mix blaring loudly on a nice set of studio monitors. No matter how good you are at mixing on headphones, you’re really missing out if you never listen to your mixes on monitors.
You need to be able to crank it up and enjoy. For one thing, it’s just fun. Secondly, it’s a great way to check for issues in your mix that you can’t hear at lower volumes with headphones.
A word of caution, don’t listen at super-loud volumes, whether you’re using monitors or headphones. Listen at a “reasonable” level and protect your hearing.
So, that’s my take on mixing with headphones. Frankly, I don’t think there’s a right or wrong answer here. You really need to evaluate your current gear, room, experience, and budget. If you can treat your room and get some nice monitors, great!
If you don’t have a dedicated studio room, or if you’re constantly mixing in different environments (especially live or remote situations), it might be worth your while to get some nice headphones.
What do I do? I’ve had the pleasure of moving several times over the last year, and each room I’ve used for my studio has sounded very different from the one before.
Since they’ve all been temporary, I haven’t been able to really dive in and treat them. So, I use a decent amount of acoustic treatment. However, the room still has some issues.
So I use headphones to give me a “room-less” mixing environment. I mix for a while on headphones, then I mix for a while on monitors. I’ve found that if I make major decisions with headphones and then “check” them on the monitors, it tends to translate much better than the other way around.
The biggest take-away point here is that you need to learn how your system sounds. Even in a less-than-ideal mixing environment, you can get good mixes. You just need to know how to compensate for any problems your system brings to the table.
This is a long learning process, but it’s well worth it. Be sure to let me know your thoughts in the comments below!
Joe Gilder is a Nashville based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.
Extron Expands SMX System MultiMatrix With New USB Matrix Boards At NAB 2011
The SMX System MultiMatrix is ideal for medical imaging systems, conference and training facilities, and other mid-sized applications that require the switching of different signal types.
Extron Electronics has announced the introduction of two new USB matrix switcher boards for the SMX System MultiMatrix modular, field-upgradeable matrix switcher.
The SMX USB matrix boards are designed to route up to eight Host CPUs to up to four peripheral locations equipped with one or more USB 2.0 devices, such as keyboards and mice, Web cams, personal media players, or portable hard drives.
They support data transfer rates up to 480 Mbps and are compatible with USB 2.0/1.1/1.0 specifications. Host and Peripheral Emulation is provided on all ports for reliable, problem-free boot up, even without a tie being made to a connected device.
SMX USB matrix switcher boards are ideal for use in the creation of KVM - keyboard, video, mouse matrix applications when combined with available SMX DVI, HDMI, or VGA matrix switching boards. SMX USB matrix boards are available in two I/O sizes: 4x4 and 8x4.
“The flexible, modular design of the SMX MultiMatrix makes it an ideal platform for KVM switching applications,” says Casey Hall, Vice President of Sales and Marketing for Extron.
“The SMX allows AV integrators to select from the wide range of available DVI, HDMI, VGA, audio, and now USB matrix switcher boards to create a unique KVM for each customer’s needs.”
The SMX System MultiMatrix Series of digital and analog multi-plane matrix switchers combines multiple, independent matrix switchers in a truly modular, field-configurable frame.
SMX frames are available in sizes from 2U up to 5U that are capable of supporting up to 10 separate matrix boards, which can be switched independently or simultaneously, under a single point of control.
The SMX combines the proven reliability and high performance of fixed matrix switchers with the efficiency of a modular matrix switcher design.
The SMX System MultiMatrix is an ideal choice for medical imaging systems, conference and training facilities, and other mid-sized applications that require the switching of different signal types, and it is a cost-effective upgrade path for ongoing I/O or signal format changes.
A Tech Director’s Journey of Blessing Another Church
Mentoring is an integral part of the Technical Director's calling. So, are you doing all you can?
Guest Post from Jeremy Montz
Jeremy Montz took his calling for volunteer church audio to a whole new level – and a completely different state. Let’s take a look at his story and discover how mentoring is an extension of his calling.
Over the years, I have often wondered as to my purpose in the plan of God. I have also puzzled over where my passion for technology, in the areas of church multimedia and sound systems, could possibly fit into such a plan.
Ultimately, asking the question; what am I going to do with the knowledge with which the Lord has blessed me?
This is where I find myself after being involved with sound and multimedia applications for roughly 16 years.
I have been extremely blessed with a budget adequate enough to have five fully-equipped sound systems on campus.
Although not all are completely digital systems or supplied with the priciest equipment, I have researched and, along with my team, installed all of these systems.
The latest project consisted of a new sound system installation for our 800 seat sanctuary, complete from the wiring on up.
The various system installations, along with much research, have gained me considerable personal knowledge and experience.
I do not consider myself an expert, but definitely more knowledgeable than someone that has just been turning knobs for a few years.
Fast-Forward: Present Time
Several months ago, I made contact with a church in a neighboring state that was interested in buying some of our un-needed sound equipment.
Mics, subs, crossover, etc. - those items that get shuffled out of the systems; used but still produce decent sound. I made solid contact with a member of their assembly through this transaction, and thereby began a friendship.
This led to many conversations about their current state of technology and where they would like to go with it.
This friendship resulted in a trip for me and two of my sound team members to assist this church in putting together a usable sound and multimedia system.
Our task included installation of a screen, a 3300 lumen projector, rewiring the sound system, and making any needed adjustments. They had erected a sound/multimedia desk before our arrival and worked hard to make sure that we could accomplish the most possible in one weekend.
Arriving at their location on a Friday evening we assessed the situation. Some of it was better and some worse than expected, but we knew what we had to do and had prepared accordingly.
Saturday morning saw us up early and installing the retractable 8’x10’ screen. We then removed the speakers from their hanging points near the ceiling and painted them white to give the finished product a much more polished look.
Positioning the projector, we made sure it was properly aligned and functional. Lastly, having arranged all of the equipment in the new sound booth, we connected the system.
All of the wiring from the amplifiers to the monitors was replaced and we then reinstalled the painted speakers, spreading them so that the projector could find its target between them without shadows appearing on the screen.
It may not sound as though we did much, but we put in a 15 hour day of work, with no option but to complete the task so that the church could have their scheduled Sunday morning service.
We had never gotten a chance to hear what the system sounded like before we arrived as they had already disconnected the equipment at the FOH position.
So we went into this project a bit blind, not being able to hear what they had experienced in the past and exactly what was causing the described problems. Being satisfied with how well the systems now sounded and functioned, we walked out of the building feeling blessed in knowing that everything was now set in order correctly.
Was it rewarding? You bet it was! At times it was difficult, but when you love doing something, it honestly doesn’t feel like work. What a feeling of accomplishment when everyone from the church is smiling and telling you how much better things look and sound.
Where To Next?
When does our mentoring process come to an end? With one visit? One month? One year? I say it could well create associations and friendships that last for life.
After the weekend at this church, I invited their team to come down to our assembly for a clinic later this Spring.
How can they continue their learning without our continued support? Our setting is different than what they are used to, but the basic principles are always the same.
We all know that this is an ever-expanding and evolving field and we must desire to attain more knowledge than we currently possess. Think about the nuggets of information that you have acquired. Remember the times that you learned the hard way.
Now is perhaps the time to begin looking around at the other churches in your circle of influence to share information and various experiences that you have had.
Twenty years worth of experience is not necessarily required to start mentoring others, as all knowledge and experience is valuable.
By suggesting that you mentor someone, I am not advising that you should get in over your head or volunteer to renovate someone’s sound system if you are unqualified.
Realize your limitations. Most of the guys that are less experienced than you are crave information and guidance but don’t have a good source to go to. Many of them will not ask, but if you offer, they will gladly accept the help.
We all need each other! Maybe you would only feel comfortable reaching out to those in your immediate church organization. I can understand that, but don’t just contemplate it, go ahead and do it. You have not been given the talent in these areas just to take it to your grave, who will profit from that?
So I ask, what do you want the end result of this passion for technology to be? It could lead into far more than one might expect, as in blessing a church with your expertise and, in the process, making new friends for life.
Let’s all put forth an effort to become a mentor to those who would benefit from our knowledge, and to God be the glory.
Jeremy Montz is the Tech Director at Lighthouse Tabernacle
Have you ever mentored or been mentored? What were the benefits? Who could you mentor in the month of May?
Ready to learn and laugh? Chris Huff writes about the world of church audio at Behind The Mixer. He covers everything from audio fundamentals to dealing with musicians. He can even tell you the signs the sound guy is having a mental breakdown.
Clear-Com Unveils HME DX210 Wireless Intercom System At NAB 2011
The latest edition of HME's DX series of digital wireless intercoms offer superior wireless performance for high demand productions.
Clear-Com has announced the introduction of the HME DX210 at this year’s 2011 NAB Show.
The HME DX210 offers the perfect blend of wireless intercom performance, system compatibility with wired intercom systems, and ease of use.
Operating in the 2.4GHz band, the DX210 delivers exceptional sound clarity and enables interference-free communication for highly demanding productions.
The HME DX210 is a newly-designed, two-channel intercom system that has advanced the capabilities of the legendary DX200. The DX210’s wired intercom interface is now compatible with Clear-Com’s or RTS’ 2-wire systems, and provides two separate 2-wire and 4-wire interconnections.
The powerful 1RU base station supports up to 16 full-duplex and 44 half-duplex beltpacks and/or wireless headsets by linking four base stations. The DX210 is paired with the rugged BP210 beltpack and All-in-One WH210 Wireless Headset COMMUNICATORs, which have two intercom buttons (IC1/IC2) with ISO.
The system is also backward compatible with the DX200 COMMUNICATOR models such as BP200, WH200 and WS200 (Wireless Speaker Station).
One of the features in the HME DX210 is a digital auto-nulling circuit, which allows users to eliminate the return audio of the wired 2-wire intercom automatically.
The system offers 2-wire circuit protection to prevent feedback from un-terminated 2-wire channels. The DX210 also has an assignable AUX input for program. The DX210 allows for relay (GPO) actuation with ISO function from either base station or any one of the appropriately configured COMMUNICATORs.
The DX210, as with all DX Series systems, features Spectrum-Friendly™ Technology to avoid frequency conflicts and Digital Frequency-Hopping Spread Spectrum (FHSS) technology for interference-free communication.
“At the 2011 NAB Show Clear-Com will demonstrate the breadth and scope of wireless intercom solutions to industry professionals,” says Craig Frederickson, Product Manager of the HME DX Series.
“The HME DX210 sets a new standard in multi-channel wireless intercom solutions with all the features and interfaces necessary for critical communications, while maintaining the HME legacy of being easy to install and operate.”
Audio-Technica Showcases ES963 Three-Element Multidirectional Boundary Microphone At NAB 2011
Full, efficient and flexible coverage, requiring fewer microphones for the same job
Audio-Technica is pleased to showcase the new ES963 Three-Element Multidirectional Boundary Microphone, part of its contractor-exclusive Engineered Sound line.
The ES963 is designed for surface-mount applications such as high-quality sound reinforcement, conferencing, professional recording, television and other demanding sound pickup applications in the installed sound and sound contracting markets.
The microphone provides full, efficient and flexible coverage of large table areas, giving users a more streamlined mic setup.
Three cardioid condenser elements mounted in the microphone housing are factory set with an angle of 120 degrees between them to provide complete 360-degree coverage (in the hemisphere above mounting surface).
Two positioning levers on the microphone base allow two of the three elements to be reoriented (with no tools or disassembly required) to offer angles of 90/90/180 degrees between elements – ideal for positioning at the end of long conference tables, for example.
Since elements can be powered and used individually or in combination, further flexibility is easily achieved. For example, with two elements powered, the microphone can provide a figure-of-eight polar pattern; with a single element powered, the microphone functions as a standard cardioid boundary microphone.
Four indented recessed pathways on the base of the ES963 enable the user to route the cable as required to accommodate any installation requirements; no tools or disassembly are needed.
The ES963 includes a 7.5-meter (24.6-foot) permanently attached output cable. Its free end is unterminated stripped and tinned pigtails, giving the installer flexibility in interfacing with a variety of equipment.
Small-diameter capsules near the boundary eliminate phase distortion and deliver clear, high-output performance.
The ES963 is equipped with UniGuard RFI-shielding technology, which offers outstanding rejection of radio frequency interference (RFI). The microphone requires 11V to 52V DC phantom power for operation. Elements can be powered and used individually, or in any combination.
The ES963’s heavy die-cast case and non-slip silicon foam bottom pads minimize coupling of surface vibration to the microphone.
The microphone features a low-profile design and a low-reflectance black finish, and comes equipped with a soft protective pouch. Output is low impedance (Lo-Z balanced).
Audio-Technica’s ES963 Three-Element Multidirectional Boundary Microphone is now available with a U.S. MSRP of $495.00.
SSL Brings New Functionality To C10 HD Compact Range With V3 Software Upgrade At NAB 2011
The new feature set demonstrates an ongoing development strategy.
Solid State Logic is proud to announce new V3 Software for the innovative and industry-leading C10 HD Compact Broadcast Console at the 2011 NAB Show.
The new V3 software release introduces a range of new features and options that significantly expand the capability of the C10 HD, demonstrating the company’s ongoing development strategy.
The C10 HD turned the industry upside down by delivering exceptional power at an attractive price point, and the legacy continues with the V3 software.
“The acceptance by the industry of the C10 HD is nothing short of remarkable,” says Piers Plaskitt, CEO of SSL, Inc.
“Sales continue to climb worldwide and, with the addition of the new V3 Software, the C10 is more desirable than ever. The C10 is more powerful and connectable and will continue to lead the industry well into the future.”
The winning formula for C10 is a unique combination of large console power and features delivered in a compact, affordable and extremely intuitive package.
In particular, a range of automated features and simplified controls make the C10 ideal for environments where users of varying skill levels will operate the console.
The new V3 software release includes a range of great new features which significantly expand and enhance the appeal of the product.
Highlights of the new software include ‘C-Play,’ another industry first from SSL, which integrates a professional audio Playout system into the console surface, delivering superior ergonomics for the operator, integrated recall of Playlists with console projects and a competitive price benefit. Compatibility with external studio systems is significantly enhanced.
V3 includes integration with Mosart Medialab Newscast Automation. Mosart is one of the world’s leading production automation systems and adds to existing
support for Sony ELC and Ross Overdrive. Full duplex connectivity with Reidel RockNet Audio Networks (including remote preamp control and compatibility with their Independent Gain System) expands compatibility with installed audio networks.
Audio Follow Video capabilities are also enhanced with independently programmable ramp on/off fade times.
Improving on a class leading set of redundancy capabilities, V3 introduces Loop Redundancy Mode for the SSL MORSE Stagebox and the new Alpha-Link Live-R MADI I/O unit. Building on the success of the Alpha-Link Live low-cost console I/O unit, the new Alpha-Link Live-R unit adds a set of Redundant MADI optical fiber connections to the existing Alpha-Link Live low cost console I/O unit.
The new Loop Redundancy Mode reduces the number of cables required for redundant fibre system installation and doubles the amount of audio signals that can be passed between the C10/C100 and the modular B-RIO I/O Unit.
V3 also brings several surround production additions. The 5.1 Fold-down system adds user adjustable individual center, rear (LS and RS) and LFE gain setting, an overall stereo output level trim and a new M-3 M-6 mono fold-down option. A new 5.1 BLITS Tone Ident generator routes to all 5.1 format PGM, ASG, channel and utility Buses.
Additional significant features include an Automatic Mix Minus Off-air Confidence Cue and Conference Mix option that improves communications in fast paced talk show environments. Gate/Expander and Program Delay options to enhance the C10’s flexibility to precisely re-time audio in applications that introduce video processing delays.
V3 Software for C10 HD is due for release in June 2011.
Solid State Logic
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