Friday, November 04, 2011
Church Sound: How To Resolve A Simple Line Check Problem
A three-step approach to resolving line check problems with vocal microphones
Resolving line check problems can be quick and easy when you use this three-step approach. It’s simple, it follows a logical flow, and it can be done in a very short period of time.
A line check is the process of checking that all instruments and microphones on the stage are sending signals to the sound board. This process happens before a sound check or might be considered part of the sound check. During a line check, you are apt to discover problems like bad cables, bad connections, dead batteries, and (gasp!) dead mixer channels.
I think of a line check as more than just getting the signal but also getting a clear signal as well. In some cases of line check problems, you will neither get volume nor a signal light on the channel. While the lack of the signal light can determine what you check, using the below process list, you’ll find out everything that should be checked when either case arises.
Resolving a simple issue with a vocal microphone
Let’s say the singer is singing into the microphone and you don’t hear anything coming through the main speakers.
Step One: Check for the obvious
These are all things you can do from the sound booth and/or have the person on stage easily check for you.
1) Channel fader. Make sure it’s at the 0 position as a good starting point.
2) Channel gain. Check the channel gain is turned up. If you aren’t seeing the signal light on the channel glowing on and off, then increase the gain to see if that’s the source of the problem.
3) Channel padding. Make sure you haven’t engaged the signal padding where it’s not needed. That could cut your signal so low you don’t hear anything.
4) Sound board volume. Make sure you have the master volume turned up on your sound board. Hey, I wouldn’t mention it if I hadn’t done it myself.
5) Subgroup usage. Make sure the channel isn’t routed to a subgroup. If it is, take it out of the subgroup and listen for the sound.
6) Wireless microphones – receiver power and signal. Make sure the receiver is turned on and the receiver shows a wireless signal. If you don’t see a wireless signal, ask the person on stage to make sure the wireless pack (or wireless handheld) is on. If it’s on but the pack/handheld doesn’t show any associated power light, it might be as simple as battery replacement. See your wireless microphone manual for how it displays a battery-strength light indicator.
7) Wired microphones. Make sure that if it has an on/off switch that it’s turned on. Ask the person on stage.
8) Channel/stage jack pairing. Sometimes, a problem can be as simple as the cable being plugged into the wrong stage jack or you have it marked as the wrong channel. Ask the person on stage to check the jack number.
Step Two: Follow the signal flow
1) Connection into microphone. Re-seat the cable into the microphone. Make sure the channel is off / muted when you do this.
2) Connection at stage jack. Re-seat the cable into the stage jack. Make sure the channel is off / muted when you do this.
3) Connection at mixer. It’s unlikely but it’s *possible* that the connection into the mixer was pulled out for that channel. Make sure it’s properly connected.
Step Three: Time to swap
1) Swap microphones. Swap the microphone for known good one and try again. You might have a microphone that’s gone bad.
2) Swap cables. Swap the microphone cable with a cable that’s known to work.
3) Swap stage jacks. Still not getting a signal to the sound booth? Might be something from the stage jack to the mixer itself. Connect to a different jack/channel and see if the mixer gets that signal. If that does get a signal, also try swapping cables on the back of the mixer from the good channel to the channel that wasn’t receiving the signal. If you still don’t have a signal, you have a bad channel on your board.
Follow this three-step approach to resolving line check problems with vocal microphones and you’ll be moving onto your sound check faster than a vocalist can say “testing 1, 2, 3.“
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.
DiGiCo Launches Expanded I/O Distribution With New MINI & NANO Racks
Instead of all the I/O connections having to be in one place, they can be distributed throughout a venue at the most convenient points
The new DiGiCo MINI and NANO racks offer a wide range of input and output options for any DiGiCo SD audio system.
Multiple DiGiCo mixing consoles can be positioned in an Optocore 2G optical loop, ideally suited to complex live or broadcast productions where multiple consoles need to share and sub-mix I/O. An example of how this can work in the real world is a scenario of front of house, monitors and a live broadcast feed.
Where the MINI and NANO racks come into their own is that, instead of all the I/O connections having to be in one place, they can be distributed throughout a venue at the most convenient points.
“With a digital system it makes no sense to have long lengths of analog cabling between your audio sources or amplifiers/loudspeakers and a central I/O rack,” says DiGiCo marketing director David Webster. “In a theatre you might want 56 mic inputs and 24 outputs as a main I/O rack, then a few more each side of the stage, perhaps a few for an event in the foyer and some more in an adjacent rehearsal room.
“Now you can use an SD-Rack for the main onstage I/O rack, but have a NANO rack each side of the stage, another in the foyer and a MINI rack in the rehearsal room, all communicating and working with up to five redundant consoles.”
Alternatively, at a sports broadcasting event, a combination of I/O racks can be distributed about the field of play, all backed up on a redundant single or multimode optical loop. Up to 14 rack IDs can be defined on each loop providing a full optical distribution system.
The MINI rack has 4 x standard SD hot swappable I/O card slots. These can be populated with any combination of the SD-Rack I/O cards; currently these include Mic/Line, Line output, AES I/O, AES IN, AES OUT, ADAT, AVIOM, DANTE and an in development HD-SDi card. Standard on the rack are MADI I/O connections along with the choice of either HMA, OpticalCon or ST optics.
Half the physical size of the MINI rack, the NANO offers two SD hot swappable I/O card slots, with the same card options. Optical connections are again user defined with HMA, OpticalCon or ST options.
With DiGiCo’s Gain Tracking, all consoles can share the inputs of all racks, while any slot of eight outputs on any rack can be allocated to any console on the optical network, provided it has not been previously allocated by another console.
“Another advantage of the system is cost savings,” continues Webster. “For example, if the FoH engineer only needs eight outputs, he can use a slot of outputs on the rack that the monitor guy is using - so it means you don’t need to buy two racks.”
Together with DiGiCo’s SD and D racks, the MINI and NANO racks provide a completely flexible I/O rack solution for any situation.
Wait - Why Are We Doing This? Progress On The System Interoperability Front
We want to have products all speak the same language, but allow them to retain their unique personalities...
The professional audio industry in 2011 has been abuzz yet again with a recurring theme that I like to refer to as the “can’t we all just get along?” conversation.
As in years past, the topic of unifying standards between different manufacturers’ equipment to make exchange of information easier and more seamless was everywhere.
This has been going on for some time, and has lived under different names and terms: “interconnectivity,” “convergence,” and the most recent model, “interoperability.”
However, this year marked a rather interesting turn of events. An increasing number of folks finally seem to all agree that this is not just a good idea, but further, serious conversations are being had about just how to go about getting it done. Customers are asking for it, manufacturers are honestly and openly talking about it, and huge levels of cooperation are emerging from the sturm and drang that frequently hinders real progress in this area.
Industry standards and initiatives such as AVB (IEEE 802.1 Ethernet Audio/Video Bridging) and X192 (AES standards task group for audio interoperability over high-performance IP networks) are not only starting the conversation about transport and content interoperability, but are backing it up with real work to promote the topic, bringing people to the table and making it happen.
Previously, the topic of interoperability was focused mainly on media content and transport. This is a critical and highly valuable part of the conversation, to be sure, but something was missing: system control. It’s an element that every system designer is acutely aware of and struggles with daily, yet until recently, it was absent from most discussions.
While the ability to exchange media freely between devices provides an obvious benefit, there is still a huge issue of how to tell these devices what to actually do with it once they have it. In other words, how do you control, configure, monitor, reconfigure, operate, adjust, modify, edit and generally manage these devices?
To tackle this issue, a new organization has been formed called the OCA Alliance, made up of individuals from nine companies who share the vision of an open, flexible, powerful system control solution for professional media networks. To this end, the group is collaborating closely to develop a system control network protocol suite known as the Open Control Architecture, or OCA. It’s the goal of the alliance to transfer OCA into the public standards domain as soon as possible, so that anybody may use it.
After the announcement of the OCA Alliance went public, I had numerous conversations with people from highly divergent areas of the AV industry, and a lot of their reactions went pretty much like this:
“You guys are working on an open control standard, huh? Cool!” (Pause) “So that means that anybody would potentially be able to implement and use this in their products? Neat!” (Slightly longer pause) “Wait - why would anybody actually want to do that?”
It’s a valid question. Upon first hearing about a unifying control technology, people are generally filled with joyful visions of how the AV industry might finally have the benefit of control interoperability that MIDI, DMX512 and others have provided to other technologies and markets. However, as one begins to think about the practical ramifications and implementations of a technology like OCA, doubts begin to creep in and one may wonder why exactly this is something that any sane manufacturer would actually want to adopt.
I can assure you that the members of the OCA Alliance are quite sane and have a very clear vision of what a technology like this means for our industry. But to see that vision, we need to look beyond how we have been working within our industry and take a longer view of how we would like to work.
So for the moment, let’s set aside the hows, bits, bytes and technical details of OCA and focus on the whys.
What We’re Talking About
What exactly do we mean when we say “control?”
Before getting into a discussion as to why all of this is important, we need to first get a handle on what we’re talking about, and - equally important - what we’re not talking about, because “control” can mean a lot of different things to a lot of different people.
On the one end of the spectrum, there are low-level details that need to be addressed within a media network. Functions such as configuring the network switches and routers, and discovering all the networked devices, are critical elements. These details are being addressed by the transport and infrastructure standards groups that are creating technologies such as AVB, but it’s not what we’re talking about here.
On the other end of the spectrum are the definitions of how network devices actually function and operate. This includes details such as what kind of DSP features and functions are available in a given device, the parameters that are available within those functions. and how those algorithms are actually coded.
This kind of control approaches dangerous territory, since it may touch on aspects of products that make them unique (for better or for worse).
For example, a frequent topic of discussion in regards to DSPs and filters is how the function of Q is defined within an equalizer algorithm. This is certainly a valid discussion, but is not part of the OCA conversation or concept. Another example might be a compressor function available in a certain product that, alongside the more standard parameters of Threshold, Ratio, Attack and Release, might contain some additional parameters that are unique to that device or implementation.
Interact, Not Standardize
OCA avoids these issues by confining itself to interacting with parameters and functions, but not defining the functions themselves. Details such as unifying or standardizing algorithms, parameters, and device functionality are soundly outside of OCA’s scope. OCA can set the Q parameter of an equalizer - but exactly what the equalizer does with that Q value is up to the equalizer, and is not standardized by OCA.
To clarify, let’s look at the more familiar world of MIDI. When we connect a keyboard controller to a MIDI tone module and press a key, a message is transmitted to the tone module telling it to execute a certain function (make a sound) within certain parameters (at the velocity of the key press, at this specific note, etc). This standard control message behaves exactly the same way regardless of the manufacturer of the tone module, and the desired function occurs.
However, that control message has absolutely nothing to do with the inner workings of the tone module itself. Details such as polyphony, synthesis method, or subjective quality of the sound that is generated, have absolutely nothing to do with the control message itself and are unique to the device that is carrying out the function.
In a nutshell, then, the goal of OCA is to create a standard method to interact with devices and their functions, not to standardize the devices and functions themselves. We want to have products all speak the same language, but allow them to retain their unique personalities.
Now that we’ve identified exactly what OCA is attempting to do and how it intends to fit in with the rest of the industry and technology at large, the next obvious question is “sounds great - but what’s in it for me?” I’ll answer that question here next month.
Ethan Wetzell has worked in audio for over 20 years, in positions ranging from front of house and studio engineer to global product manager for Electro-Voice DSP. He currently works as platform strategist for Bosch Communications Systems and works with the OCA Alliance.
Posted by Keith Clark on 11/04 at 09:49 AM
Wednesday, November 02, 2011
PreSonus FireStudio (26 x 26) Now Lion-Compatible
With this release, all FireStudio Series interfaces, including discontinued models, are compatible with OS X Lion
PreSonus has released FireControl 2626, a stand-alone application that provides Mac OS X 10.7 Lion compatibility for the original FireStudio (26 x 26) audio/MIDI interface.
With this release, all FireStudio Series interfaces, including discontinued models, are compatible with OS X Lion.
FireControl 2626 is exclusively for the FireStudio (26 x 26) and OS X Lion; FireStudio users who are running earlier versions of Mac OS X or who are using Microsoft Windows do not need and should not install this release.
Users of all other FireStudio-series interfaces and users of StudioLive-series mixers should update to the recently released PreSonus Universal Control 1.5.2, which provides Lion compatibility and other enhancements for those products.
FireControl 2626 allows FireWire daisy-chaining of two FireStudio (26 x 26) interfaces, and it is possible to chain FireStudio interfaces with other FireStudio-series interfaces by running both FireControl 2626 and Universal Control 1.5.2 simultaneously.
FireControl 2626 and Universal Control 1.5.2 are free downloads and are available immediately here.
Clear Path: The “Right” DI For Computers In Audio
When it comes to best audio quality practices, they’re sometimes not ideal.
Ready, brace yourself, this one is going to hurt: Computers are not made for audio. There, I finally said it!
Computers are made for crunching numbers, and they also happen to be able to manage audio and video tasks really well because of their tremendous processing power.
However, when it comes to best audio quality practices, they’re sometimes not ideal. Often we use computers to feed tracks to PA systems and as playback machines for things such as backing tracks. Getting the sound from the computer into a sound system is relatively easy: Connect the 3.5-mm (1/8-inch) unbalanced output jack and away you go. If only it were that simple…
Anyone who has done this knows that more often than not, it can introduce a ground loop or induce noise via the unbalanced line. Even PA system noise can find its way into the computer, adding noise to the program material output. Amplify any of this with 20,000 watts and you have a problem.
Several companies produce direct boxes that are specifically designed for computers. These are usually stereo, and more often than not, are passive or transformer based.
In other words, the transformer not only converts the unbalanced signal into a balanced one, but also introduces galvanic isolation to eliminate stray DC currents from traveling in between the computer and the audio system. And when the ground is lifted, all of the audio passes through the transformer disconnecting the ground thus eliminating the ground loop.
Because the computer’s output is buffered (usually by a -10 dB consumer level or headphone jack), a passive DI is perfectly suitable for computers. Transformers can usually handle a lot more signal before distortion when compared to phantom powered active DI boxes. This makes them a better choice when using the headphone jack.
Passive boxes for interface computers include (left to right) the Whirlwind pcDI, Proco AV1B and Radial ProAV2.
The active direct box was originally developed as a means to eliminate loading that would occur on low output electric bass pickups. By introducing a buffer, the bass signal going to the artist’s stage amp would not be affected thus conserving his sound while the PA system would be fed a hotter signal.
Buffers are essentially amplifiers. This means that they need power (voltage and current) to make them work. The preferred power source is 48-volt phantom because it does not require running separate AC for the DI box.
The other hidden advantage of a buffer is that the signal will only go one way. Unlike a transformer that is bi-directional, buffers do not allow signals to go backwards. Where this matters in our world is preventing noise from polluting the computer.
And because most program material is limited during the mastering process, one can get sufficient headroom using phantom power to generate a relatively clean signal. The problem, unless dealt with, is the lack of galvanic isolation; active DIs don’t solve ground loop problems.
There are some DI boxes that combine the benefit of an active direct box with transformer isolation. These are usually a little more expensive than a simple passive or active DI because they offer the best of both worlds. The transformers isolate the computer from the PA, while the buffers inhibit PA noise from polluting the computer.
Peter Janis is the president of Radial Engineering and has worked in professional audio for more than 30 years.
Posted by Keith Clark on 11/02 at 11:50 AM
Monday, October 31, 2011
Gepco Unveils Two-Channel Heavy-Duty Tactical Cat-5e Cable
Solution for applications that require multiple or redundant channels of Cat-5e cables in remote production or staging applications
Gepco International has introduced CTS2504HDX, a 2-channel snake consists of two elements of Gepco’s CT504HDX heavy-duty tactical Cat 5e cable under a rugged TPE jacket.
It is a solution for applications that require multiple or redundant channels of Cat-5e cables in remote production or staging applications.
Typically, the electrical performance and bandwidth of conventional Cat 5 cable is degraded through physical damage when used in portable applications, with the unique double-jacket construction of the CT504HD series designed to eliminate this issue.
While the inner jacket maintains the proper physical spacing between pairs to achieve ISO/IEC or TIA/EIA Cat 5e specifications, the durable TPE outer jacket protects the cable from physical damage or abuse.
In addition to the new CTS2504HDX, the CT504HD Series of heavy-duty Cat 5e cables includes three other types. The original CT504HD has 24 AWG stranded conductors for exceptional flexibility, while the CT504HDX features 24 AWG solid conductors for lower attenuation that allows for the full, recommended TIA distances for Cat 5e network cable. With the same basic construction as the new CTS2504HDX, the CTS4504HDX is a 4-channel snake consisting of four elements of CT504HDX under an overall rugged TPE jacket.
Heavy-duty tactical category 5e Assemblies provide a pre-terminated cabling solution for hostile environments. The CT504HD, CT504HDX and each element of the CTS2504HDX and CTS4504HDX can be terminated with either standard Cat 5 RJ45 connectors or ruggedized Neutrik etherCON connectors.
“The concern among Cat 5e cable users in the professional audio/video industry has been that it isn’t durable enough to handle the traditional wear and tear associated with the workload,” states Joe Zajac, market development manager for Gepco brand products. “Our CT504HD series was designed specifically to meet the needs of portable applications and provides the answer to professionals who are looking for a Cat 5e solution in remote environments.”
Friday, October 28, 2011
The Basics Of Fiber Optic Transmission Systems
Given its increased use throughout the A/V industry, most contractors are now required to understand the basics of terminating and laying fiber optic cable.
With the increasing trend toward the use of fiber instead of co-ax cable in a wide range of applications, most contractors are now required to understand the basics of terminating and laying fiber optic cable.
Contrary to its reputation, fiber is actually quite easy to handle and use.
Fiber, at its most basic level, is a very pure strand of glass through which light can pass over great distances. All fiber optic cable has at its center a fiber core made of such glass, which is used for the actual signal transmission.
The two most common techniques for protecting the fragile fiber are enclosing it in a loose-fitting tube and coating it with a tight-fitting buffer.
In the loose-tube method, the fiber is enclosed in a plastic buffer-tube that is larger in inner diameter than the outer diameter of the fiber itself.
This tube is sometimes filled with a silicone gel to prevent the buildup of moisture. Since the fiber is basically free to “float” within the tube, mechanical forces acting on the outside of the cable do not usually reach the fiber.
In the tight buffer construction, a thick coating of a plastic-type material is applied directly to the outside of the fiber itself.
This results in a smaller diameter of the entire cable and one that is more resistant to crushing and impact. However, because the fiber is not free to “float”, its tensile strength is not as great.
Tight buffer cable is generally lighter and more flexible than loose-tube cable and is usually employed for less severe applications such as within a building or between individual pieces of equipment.
Figure 1: Two methods of fiber-optic buffering.
Like copper wire, fiber optic cable is available in many varieties. There are single and multiple conductor constructions, aerial and direct burial styles, plenum and riser cables and even ultra-rugged military-type tactical cables that will withstand severe mechanical abuse. The cable one chooses is, of course, dependent upon the application.
Both loose tube and tight-buffer constructed cables are available in single-mode and multimode versions. These terms refer to the diameter of actual glass fiber located within the core of the cable. More specifically, they refer to the number of light paths that may pass through the fiber.
Single-mode fiber is so thin (8 to 10 microns, diameter) that only a single path of light can pass through its length. By contrast, multimode fiber, 65 microns in diameter, allows multiple paths of light to travel along its length simultaneously. Although it may seem counter-intuitive, single-mode fiber is able to carry more information over farther distances than multimode fiber.
Terminating Fiber Optic Cable
The procedure for terminating fiber optic cable is a function of the type of connector being used, rather than the type of fiber.
There are two types of connectors most frequently used today: ST and FCPC.
As ST connectors may be used with either multimode or single-mode fiber and do not require any expensive, special equipment — unlike FCPC connectors — this article will focus exclusively on termination using ST-type connectors.
All tools required for this type of termination can be purchased in standard fiber terminating kits available from fiber optic equipment manufacturers.
Fiber optic cable offers the installer a great deal of freedom and flexibility during the actual installation process.
For starters, fiber is light and easy to handle, and much less of it must be laid than the amount of co-ax required to provide an equal level of transmission capacity.
The specifics of how and where fiber can be laid is mostly a function of the type of fiber being used. As discussed in the “Cable Construction” section of this article, fiber is available in a wide range of constructions, each designed to withstand certain types of environmental conditions and application challenges.
Figure 2: The final steps. Apply bead of epoxy to the protruding fiber tip. When dry, score tip with glass scriber and break off end. Sand tip to remove remaining fiber and epoxy particles. Finally, polish the fiber tip using a finer grit micro-polish.
In general, all fiber uses less duct space than co-ax and, in fact, may often be laid without ducts — simply passing between walls and flooring wherever convenient. It can also accommodate structural curves and turns, although any tight bends must have a turning radius of at least 1 inch.
Similar to using electrical cable, the first step in terminating fiber cable is to strip it. This involves stripping back the plastic coating of the fiber cable to reveal the glass core inside. A tool called a fiber-optic stripper, which looks like a small pair of pliers with jaws that grip the coating, is often used in this process.
Once this is done, the stripped material is trimmed back and inserted into a restraining grommet or sleeve, also called a boot.
After the cable is stripped, the ST connector must be prepared for use.
Simply apply a dab of a quick-drying epoxy resin on the end of the optical connector.
Once the resin is applied, immediately insert the fiber into a precision hole in the connector pin.
At this point, with the fiber inserted through the connector hole, the fiber tip should be protruding from the front of the connector pin. Apply a small bead of epoxy to this exposed end, and set the fiber/connector assembly aside to dry properly. Ideally, the epoxy should be allowed to dry overnight, but a 1-hour drying time is sufficient when time is not available.
Once the epoxy is completely dry, use a scribing tool, which looks similar to a paring knife, to score the fiber close to the epoxy bead. It is important that the fiber be cut flush with the end of the connector pin.
Next, the fiber tip must be ground down and polished. A sanding plate is used to smooth away any fiber that may be protruding through the epoxy.
After sanding, you should see a very small black dot on the epoxy. This is the actual end of the fiber.
Last, the fiber must be polished. A polishing wheel coated in a finer grit micro-polish is used to remove any small particles that may still be on the tip of the fiber.
After polishing, a compressed air hose is used to blow off any microscopic particles. Then a lint-free wipe with some rubbing alcohol is used to clean the optic tip.
The termination is now complete, but it is good practice to do some quick testing at this point. Otherwise, problems may arise during or after installation, at which point diagnosis will be more difficult.
The first step is to examine the connector under a fiber-optic scope to make sure it is not exposed, broken, cracked or plucked (i.e., riddled with small holes made by particles as a result of scoring the fiber). Next, the connector should be attached to either a transmission unit or a test fixture that tests the loss in dB of the fiber cable.
Fiber that has been correctly terminated should show no additional loss as a result of the added connector.
If the termination shows no physical problems and the testing indicates an acceptable level of loss for length of cable, then the optical connector is ready for use.
Wipe the tip of the fiber clean and place a protective dust cap on it. Now the process is complete.
While this procedure does get easier with practice, it is not difficult to master and can be done relatively quickly, even by a novice.
In fact, once you are completely familiar with the finishing steps, the most time-consuming aspect of the entire process is waiting for the epoxy to dry.
And, for those who may still have reservations, there are “quick-crimp” connectors that eliminate the epoxy and finishing steps altogether. While these “quick-crimps” are more convenient in the field, the connection has slightly more optical signal loss.
While optical connectors can be used to connect fiber optic cables together, splicing — the process of terminating one fiber directly to another without use of a connector — is often more desirable because it provides lower signal loss. Two of the most common types of splices are the mechanical splice and the fusion splice.
In a mechanical splice, the ends of two pieces of fiber are cleaned and stripped, then carefully butted together and aligned using a mechanical assembly. A gel is used at the point of contact to reduce light reflection and keep the splice loss at a minimum. The ends of the fiber are held together by friction or compression, and the splice assembly features a locking mechanism so that the fibers remain aligned.
A fusion splice involves melting (fusing) together the ends of two pieces of fiber. The result is a continuous fiber without a break. Fusion splices require special, expensive splicing equipment but can be performed very quickly, so the cost becomes reasonable if done in quantity.
Because fusion splices are fragile, protectors and a plastic coating called shrink tubing are usually placed around the spliced area to protect it from breakage.
Fiber is virtually unaffected by outdoor atmospheric conditions and electrical interference; it can be lashed directly to telephone poles or electrical cables without concern for extraneous signal pickup. Because it is so resistant to the environment, fiber is ideal for connecting systems between buildings when cable must be laid outside, underground.
In fact, if the proper type of fiber cable is used, it can be laid directly in the ground with no concern for exposure to moisture or humidity. And if a cable is accidentally severed, there is no risk of a spark causing a fire or endangering personnel.
Amplifiers and Repeaters
Although fiber optic cable is often chosen over co-ax because it can transmit signals over longer distances, there are certainly limitations to how far fiber transmission systems can carry a signal without amplification.
When the desired transmission distance exceeds the maximum distance that a system is designed to support, amplifiers or repeaters are required.
In an AM- or FM-based system, amplifiers are used to boost the strength of an attenuated signal so that it can be transmitted along an additional length of fiber.
Fiber-optic amplifiers are very similar to their traditional electrical counterparts. The transmitted light beam is captured by the amplifier, converted back to a voltage for amplification purposes, and then relaunched as light for transmission over the next span of fiber.
As in copper-based systems, fiber optic amplifiers do pass on any distortions and interference that have been acquired by the signal throughout the transmission, and those distortions are amplified along with the signal.
Therefore, if a signal is amplified enough times, it will become greatly distorted.
This problem is eliminated when using a digital transmission system, as transmission length is extended through the use of repeaters instead of amplifiers.
When a fiber optic system uses digital signaling techniques, a repeater converts the transmitted light beam back into its electrical equivalent, in digital format, and then launches a brand new fiber-optic signal based on the regenerated digital electrical signal. (Note that the signal does not return to its baseband format until it reaches its final destination.)
Because of the digital nature of the transmitted signal, no distortions are picked up by the repeater or passed on in the repeating process. Therefore, theoretically, digital repeaters could be used to transmit a signal over an infinite length of fiber.
This is a significant advantage over traditional AM and FM systems and is not limited to systems designed for the transmission of digital baseband signals. Today, there are fiber-optic systems that use all-digital signaling and processing to transmit traditional analog video, audio and data signals, and do so at a competitive price.
With a little practice, laying and terminating fiber cable should become just as simple as using co-ax, and the advantages are innumerable.
For more information on fiber optic technology, read the educational guides Introduction to Fiber Optics, Fiber Optic Cables and Connectors and Advantages of Digital Fiber Optic Systems, available at the Communications Specialties website.
Wednesday, October 26, 2011
Church Sound: Clearing Up 1/4-Inch Connector Confusion
Always make sure the 1/4-inch cable you are using is the right one for your application
There can be confusion when it comes to the 1/4-inch connector.
Guitar players and sound engineers are each seeking a certain type, but often have or are given the other.
Let’s explore the simple 1/4-inch connector that has come to complicate our world.
We can start with how it is known: audio jack, phone jack, phone plug, jack plug. Specific types and variations include the stereo or mono plug, mini-jack, mini-stereo, headphone jack, longframe, tiny telephone (TT) connector and Bantam plug.
Technically, the term “jack” refers to the female type (socket) whereas the word “plug” describes the male type (pictured), but the terms are often used interchangeably so we won’t split hairs.
—The term 1/4-inch (or 6.3mm) refers to the diameter of the plug or jack. Miniaturized versions include 1/8-inch (3.5mm) and 3/32-inch (2.5mm).
—The pointed end of the plug is called the tip (3), and the shaft is known as the sleeve (1). If the connector has two or more bands around the shaft (4), the space between them is called the ring (2).
—Each conductor will be wired in a specific way depending on the application. More on that in a moment.
—TS (Tip/Sleeve), or 2-conductor connectors, are typically used to transfer unbalanced mono analog audio signals.
—TRS (Tip/Ring/Sleeve), or 3-conductor connectors, are typically used to transfer balanced mono or unbalanced stereo analog audio signals.
—Less common, 4- and 5-conductor connectors are used on some devices to transfer send and receive audio or for audio + video signals.
In their original application, 2-conductor 1/4-inch plugs were used by telephone operators to connect one caller with another in the days of the manual telephone exchange.
Today, common uses of 1/4-inchconnectors include:
—Audio outputs for headphones and earphones (1/4-inch or 3.5mm TRS).
—Audio inputs on loudspeakers (1/4-inch TS).
—Line-level I/O connections on mixers, power amplifiers and signal processors (1/4-inch TRS or TS).
—Send/Return (Insert) points on mixing consoles (1/4-inch TRS or TS).
—Audio inputs and outputs on guitars, keyboards and instrument amplifiers (1/4-inch TS).
—Effects pedals for electric guitars and keyboards, and MIDI triggers for electronic drums (1/4-inch TS).
—Microphone inputs on portable audio recorders (3.5mm TRS or TS) and some entry-level audio equipment (1/4-inch or 3.5mm TRS or TS).
—Mic or line level I/O from PCs and laptops (3.5mm TS or TRS).
—Patch bay connections in audio and telecom applications (standard, long frame or TT/Bantam 1/4-inch TS or TRS).
—Audio + video output on some consumer electronics devices such as camcorders and portable DVD players (3.5mm TRS or TRRS).
—Headphone or headset connections on cellular phones and mobile devices (3.5mm TRS or TRRS, occasionally 2.5mm TRS).
At this point, you may be wondering: what about the cable it’s wired to? Glad you asked. As stated earlier, the tip, ring and sleeve conductors are wired differently depending on the cable’s intended use.
Here’s a wiring guide:
It’s also important to know that not all 1/4-inch cables are created equal! Even though the connectors on two cables may look identical, the cable type may not be.
For example, guitar cables use a braided shield around a center conductor, and loudspeaker cables use two shielded wires with no braid. These cable types have different impedances, tolerances and other specifications that make them uniquely suited for their intended purpose.
A guitar cable plugged into the output of a power amp pushing enough wattage can melt, and even start a fire! Always make sure the 1/4-inch cable you are using is the right one for your application.
Church Audio Video specializes in the design, installation and support of high-quality and affordable custom audio, video, lighting, broadcast and control systems for worship facilities. For more information, visit their website.
Posted by Keith Clark on 10/26 at 09:08 AM
Monday, October 24, 2011
Gepco Launches RunONE Powered Loudspeaker Cables
Combine audio, power and optional data under one jacket
Gepco International has introduced RunONE powered loudspeaker cables, which combine audio and power, along with optional data, under one durable yet flexible jacket.
Each RunONE cable combines one channel of power with two, eight or 12 channels of 110-Ohm balanced audio for line level, mic level or digital AES audio signals and can be used with self-powered loudspeakers or in DMX lighting control.
Additional configurations include two channels of Category 5e cable that can be used for data drops in remote power and audio applications.
Snakes with optional data can also be used for digital audio transmission while running power to Front of House for remote locations.
Shielding around the power channels eliminates power noise from interrupting the audio/data signal, ensuring high-quality performance.
Terminated with industry-standard connectors, RunONE cables offer the option of Edison, IEC and Neutrik powerCON connectors for the power channel; 3-pin XLR, 5-pin XLR (for DMX lighting), TRS and Neutrik convertCON connectors for audio channels; and RJ45 and Neutrik etherCON connectors for optional data channels.
RunONE cables are available in pre-defined and custom configurations.
“The RunONE cables are a great solution for anyone looking to save time,” says Joe Zajac, market development manager for Gepco Brand products. “With up to 12 channels of audio combined with power and the optional two channels of data, the RunONE cables will also provide for much cleaner set-ups.”
Powersoft Joins AVnu Alliance Focusing On AVB Standards
Part of company’s plan to expand its presence in install market, as well as to produce future AVB-compliant products
Powersoft has become a promoter member of the AVnu Alliance, the industry forum that aims at establishing and promoting the IEEE 802.1 Ethernet Audio/Video Bridging (AVB) standards.
The Italian professional audio manufacturer specified that the move is part of the company’s plan to significantly expand its presence in the install markets, as well as producing future interoperable, AVB-compliant audio networking products.
“Powersoft firmly believes in standard protocols as a way to help our clients,” says Claudio Lastrucci, Powersoft R&D manager and one of the company’s founders. “We have been working in this direction for a while, as technical contributor to the AES, and now as a promoter member of AVnu alliance.
“On this basis we think we can also successfully contribute to the efforts to ensure interoperable AVB products,” he adds.
Luca Giorgi, Powersoft pro audio business unit manager, states, “Our clients demand true plug-and-play, affordable devices that can effortlessly recognize, and talk with, other devices in their network. We at Powersoft fully support the mission of AVnu Alliance, since we believe that open standards are the only way forward to provide customers with inexpensive, user-friendly solutions for their networked applications.”
“We welcome Powersoft to the growing number of AVnu Alliance members who are committed to interoperability and AVB standards,” says Lee Minich, AVnu Alliance Marketing Workgroup chair and president of Lab X Technologies. “Powersoft’s collaboration in AVnu Alliance will better serve their customers and grow the entire market.”
AVnu Alliance is an industry forum dedicated to the advancement of professional-quality audio video by promoting the adoption of the IEEE 802.1 Audio Video Bridging (AVB) standards over various networking link-layers. The organization creates compliance test procedures and processes that ensure AVB interoperability of networked A/V devices, helping to provide the highest quality streaming A/V experience.
The alliance also promotes awareness of the benefits of AVB technologies and intends to collaborate with other organizations and entities to make use of this work in their respective efforts to provide a better end-user A/V experience.
Wednesday, October 19, 2011
Symetrix Grows SymNet Solus Product Line With Debut Of New Solus 16
Developed for small to mid-sized installations not requiring I/O expansion
Symetrix announces an addition to the SymNet Solus product line – the Solus 16.
“The Solus 4 and Solus 8 already provide two of the most popular form factors requested by integrators,” said Trent Wagner, senior product manager at Symetrix. “But input counts run higher in many types of installations, and we received a barrage of requests for a higher input form factor. The Solus 16 answers that request without requiring a jump to networked DSP or separate expansion I/O devices maintaining the high value for which the Solus line is known.”
Solus is powerful SymNet DSP hardware, developed for small to mid-sized installations not requiring I/O expansion.
The entire family of SymNet hardware, including Solus, is configured using open architecture SymNet Designer software.
System designers have the option to use or modify Solus DSP design templates for basic projects, or, to create unique designs entirely from scratch.
The three Solus hardware offerings differ only in their audio input and output counts:
—Solus 16 with sixteen inputs and eight outputs;
—Solus 8 with eight mic/line inputs and eight outputs;
—Solus 4 with four inputs and four outputs.
Ethernet, ARC port, RS-232 port, two control inputs, and four logic outputs complete the control feature set.
To simplify set-up, a front panel LCD displays system settings. Solus supports Symetrix ARC wall panels, third party control systems, and SymVue, a SymNet end-user control panel application.
Harman Pro Offering HiQnet System Architect And AVB Training Course In November
Will take place at Harman Signal Processing in Salt Lake City on Nov 17-18
The Harman Pro System Development and Integration Group (SDIG) has announced it will be offering a training session for its HiQnet System Architect software, as well as AVB networking.
The training will take place at Harman Signal Processing in Salt Lake City, Utah on November 17-18, 2011, led by Harman product development specialist Emilian Wojtowycz, an expert in the implementation of System Architect for installed sound applications.
HiQnet is a communications protocol that enables all the compatible devices in the audio signal path, from mixing consoles to loudspeakers, to seamlessly communicate with each other.
System Architect is the software used to set up and configure a HARMAN HiQnet and AVB system.
Among the topics to be covered are the following:
· Design workflow
· Overview of Ethernet AVB technology
· Advanced design workflow
· AVB networking
· Day-to-day operation
“Our latest version of System Architect, version 3, is ideally suited to handle the unique challenges of sound system design and reduce the complexities of networked audio routing,” notes Adam Holladay, market manager, Harman SDIG. “This training course will provide attendees with invaluable, practical, time-saving knowledge to help acquaint them with the capabilities of HiQnet System Architect and AVB.”
Master Class In Fiber Optic Audio To Be Presented By Optocore Founder Marc Brunke At AES
A discussion of the discuss the technological underpinnings as well as real-world case studies of complex and multi-faceted broadcast, studio, and live performance applications
Optocore founder and chief engineer Marc Brunke will host a presentation entitled “The Fundamentals of Audio and Data Networks over Fiber Optics and Cat-5 Cabling” at the 131st Audio Engineering Society (AES) Convention.
Specifically, the program will begin at 9 am on Friday, October 21, at Room 1E08 on Level 1 (lower level) of the Jacob Javits Center in New York, site of the AES Convention.
Brunke will discuss his views on the direction the industry is taking, the constantly expanding role of fiber optics in entertainment technology systems, and he’ll also be available to answer questions.
Different approaches and different ways of dealing with synchronization and jitter problems will be described; each point will be supported with an example from real life applications, with all present known technologies taken into consideration.
Brunke will also cover topics such as component technology advancements and their cost/performance ratio as it relates to recognized AES industry standards such as MADI, along with other current (and future) platforms.
The program will run for 90 minutes, providing ample time to discuss the technological underpinnings as well as real-world case studies of complex and multi-faceted broadcast, studio, and live performance applications.
Brunke has almost 20 years of experience in fiber optic transport disciplines, and their influence on commercial design challenges.
For further information go to www.optocore.com. In North America, please contact Brandon Coons at the Optocore North America office at 1-(416) 287-1144.
Tuesday, October 18, 2011
Audio? Confusing? Learning Is A Life-Long Process
Consider the input types that may exist on a mixing console. I found all of these on units sitting around the shop...
Almost every Syn-Aud-Con seminar has attendees from other technical fields that need to learn about sound systems and audio.
These fields include networking, telephony, lighting, electrical and others.
Many tell us that audio is the most confusing thing they have encountered in their technical careers – and it is no wonder.
Consider the input types that may exist on a mixing console. I found all of these on units sitting around the shop.
There are nine (9) analog topologies and twelve (12) digital topologies. Each serves a purpose. Each works fine for its intended application. Each is defensible from a technical and practical point-of-view. Each will likely remain in use as other connector types and topologies emerge.
Consider also that some of these have several variations, such as the polarity convention on an XLR connector or AES3 on a DB25 connector.
It is ironic that digital I/O is often touted as making things easier, yet there are more digital connector types than analog! Add to this the confusion caused by the need to configure digital I/O for the correct sample rate, bit-depth, etc.
It’s no wonder that noise and distortion remain the weak links of most sound systems. They often result from feeding the wrong signal to the wrong jack.
We have all heard a DVD player over-driving a microphone input. Yes, you get sound, but in audio the presence of sound does not necessarily mean that you hooked it up right.
A modern digital mixer may be able to convert between any of these formats.
The signal may come in as some form of analog and go out as some form of analog or digital.
The user must often choose based on the required cable length, input options on the next device or some other criteria.
So not only does the audio tech have to understand the connectors and interface topologies, he must also know the characteristics of the devices at the other end of the cable.
The interconnect is the easy part. Consider the knowledge and experience required to configure a DSP for a 3-way loudspeaker.
Technical complexities aside, the most perplexing part of audio for non-audio people is the artistic side. While some levels can be set with voltmeters and analyzers, many other adjustments are based on subjective criteria – you just turn the knob until it sounds right. But what is right? There can be any number of “rights.” How confusing is that!
The really good audio people have strong theoretical and practical backgrounds. They have “must have” tools in their toolbox that you can’t buy anywhere. They have the ability to diagnose many system problems by just listening to a speech track played over the system. They can often make it sound way better by turning one knob a little bit.
Their personal study time may be divided between Sound System Engineering, product manuals and Einstein’s papers on relativity, and they understand that all three are completely relevant to what they do.
Most importantly, they know that they will never know it all.
Yes, audio is confusing. Background in another technical field can help, but learning audio is a life-long process.
Pat and Brenda Brown own and operate SynAudCon, conducting training seminars around the world in addition to providing in-depth web-based training.
Posted by Keith Clark on 10/18 at 01:01 PM
Monday, October 17, 2011
Going Mobile: The 2-Pound, 72-Channel Wireless Console
Inside one of the transitions taking place in the world of mixing consoles
Not so long ago, mixers had six channels, round knobs, and green paint. Next came a parade of large-format mixing desks, weighing hundreds of pounds.
Later, the first digital consoles appeared, also weighing hundreds of pounds. (Is there an echo in here?)
Finally, second generation digital consoles emerged as smaller, lighter, and less expensive versions of their predecessors, some with surprisingly advanced capabilities including really useful on-board effects, choice of EQ types, optional plug-ins, and much more.
Whether you love, like, or hate them, digital consoles have forever changed the way that business is conducted in pro audio.
Today a similar technology transition is taking place that’s likely to become a full-fledged revolution in the not-too-distant future. While we strongly suspect that some of the leading console companies are already working along these lines, it’s no secret that a small innovative company based in Las Vegas, called RML Labs, is leading the thrust into this new frontier, in much the same way that other companies took the leap of faith to transition from analog consoles to digital formats, some twenty years ago.
RML Labs, the creator of the Software Audio Console (SAC for short), is a software development company that has turned a Windows PC into a powerful virtual live mixing environment.
“It all started because I was tired of humping heavy consoles into venues and I decided to do something about it,” explains Bob Lentini, inventor of SAC and owner of RML Labs. “As I began working on the initial concept, I quickly realized that a host of other advantages would become even more important than simple weight reduction.”
Lentini came to this realization quite early (“bleeding edge” being close to the mark), having introduced SAC as a proof-of-concept way back in 1992 at an AES convention in New York City. Now, nearly 20 years later, SAC has proven its worth on thousands of shows mixed by a regiment of front of house and monitor engineers, who weren’t afraid to try something out of the ordinary.
“First and foremost, this new approach, centered on digital technology, would have to sound exceptional - not just good - with respect to the best analog consoles of the day,” Lentini notes. “Making great digital sound takes meticulous care in programming, and that doesn’t happen overnight. The variables are immense.”
After two decades of work, he contends, “We’ve been able to achieve sonic properties that match, or perhaps exceed, the offerings of even the largest and most well-funded companies.”
While researching this story, the author spoke with numerous users who are as passionate about SAC’s sound quality as Lentini himself. One is Steve Emler, long-time front of house mixer for Tesla. Elmer told us, “While the other aspects of SAC are useful and intriguing, if it didn’t sound as good, or better, than anything else I’ve heard, it wouldn’t fit my needs.”
That says a lot. Elmer’s comments indicate that it’s not just about convenience, as valuable as that may be, but about sonic excellence which is something that all conscientious professionals continually strive to achieve.
Bear with us. A key attribute that enhances the sound quality of a SAC system is it utilizes linear integer processing with hexi-decimal extensions that eliminate fractional values when digitizing audio.
With other digital audio formats, their mathematical calculations of audio are summed primarily in fractional values - and these must ultimately be interpreted to 1 or 0 (binary code). Such interpretations are made randomly and result in inherent inaccuracies.
SAC, on the other hand, does not round-off ones and zeros arbitrarily. The greater accuracy of these proprietary algorithms result in audible improvements that not only emulate analog, but perhaps even improve upon it, in terms of reduced noise floor, lower distortion, and increased resolution.
That’s a mouthful, to be sure. For those of us who do not design A/D and D/A chip sets for a living, or spend our time writing hexi-decimal code, the simple explanation is that SAC provides a level of audio purity that goes well beyond the norm.
Users passionately agree that Lentini’s algorithms provide an audible improvement that’s not only significant when analyzed scientifically, but clearly audible as well. It seems that the topology and component selection of the analog portion of the pre-amps plays a smaller part than might be expected, when the digitalization is so well executed.
To the uninitiated, this may well seem like a complete reversal of the established standards that have prevailed since the dawn of analog electronics. Let’s explore further.
Of significant importance, SAC is written in assembly language. A declining art in these modern times, where C++ and other high level languages are the norm, assembly language is low-level and close to machine code - and therefore highly efficient.
Events happen extremely fast in assembly language because processing overhead is very low. An old 486 processor running assembly code can beat a new multi-gigahertz CPU that’s running a high-overhead language. What’s fascinating, and very important, is that the results can be heard, not just benchmarked.
A screen shot of the current Software Audio Console (click to enlarge).
While a SAC license can be purchased directly from RML (and a free demo is available at www.softwareaudioconsole.com), software alone does not a system make.
And while a small-business software developer needs to stay very focused on, well, software, which is what Lentini loves to do, numerous other bits and pieces are needed to make a real-world functional system…and they must be carefully integrated with expert knowledge and care.
This is exactly why Value Added Resellers, or VARs, play an essential role in completing the picture. One leading VAR interviewed for this article assembles a full working system (pun intended) to ensure that all aspects are sorted out and ready to use “right out of the box.”
Based in the beautiful mountains of Lake Tahoe, where “getting air” is a common event during snow season, the principals of AIR Consoles (a.k.a., Audio Integrated Research) have an intense vision of their own: changing the world of large-format mixing desks as we currently know them.
Their version of getting air has to do with the mobility of the control surface, not just jumping off mountain tops… which they also excel at.
AIR Console’s president Damon Gold, and vice president Glen Campbell (no relation to the singer), share a mission - along with a handful of other VARs and integrators – to supply the hardware, education, and services that give SAC the fuel that gets the job done.
An AIR console consists of a one or more banks of microphone pre-amps (usually 8 channels per bank that normally include output channels), plus one or more PC tablets for remote control. Options include a touch screen controller (up to a 32-inch screen with 50 touch points), and all other supporting hardware that brings the whole system together.
An AIR console can be as small as 8 x 8, or as large as 72 x 72 – and 128 x 128 will soon be available for mega-events.
Presently, various commercial preamps can be utilized - and often are - but the company is headed in the direction of developing its own optimized I/O product that will talk directly to the assembly language code that SAC is based upon, thereby providing what might be accurately called a “super-interface.”
SAC working with an AIR package. (click to enlarge)
The range and depth of possibilities that an AIR Consoles package brings to the table is considerable. Much like the introduction of the early digital consoles – long after the world became comfortable with analog desks – an AIR console is best utilized when the engineer fully embraces the new approach that AIR is capable of bringing to the workflow - rather than trying to merely emulate a physical console of the past.
A good example of this new workflow is the way that multiple monitor mixes are handled. While traditional methodology demands a large number of Aux sends to provide a large number of individual mixes, an AIR console takes a nearly opposite approach.
At first blush, AIR’s six stereo Aux sends would seem to fall far short of the number needed to accommodate today’s big productions. But delve a little deeper, and you’ll quickly find that there are 24 additional virtual consoles, each with their own six stereo Aux sends (yes, that’s 24 virtual consoles).
So instead of tweaking Aux sends all night long to keep the performers happy, each monitor mix has its own virtual console. The Auxes can now be utilized as effects sends, instead of the backbone of the various monitor mixes.
What’s more, none of the virtual consoles interact, in any way, with other virtual consoles, except to share the same head-amps.
It would be entirely feasible to have 24 separate operators - each concentrating on a single monitor mix – if that’s what’s needed for a given situation.
The same scenario can be used for other applications as well, such as when many different zones require different balances.
Consider a complex AV system that encompasses numerous rooms, or even numerous geographic locations. Each location might have sources that are both local and remote. In this scenario, all virtual consoles would share all inputs from all locations (or just accept stems, if that’s preferred). The individual location engineers can then balance the global inputs (or stems) against their individual local sources, thereby providing the appropriate mix for the room they’re serving.
While this is a concept that could be executed with conventional hardware, the complexity and cost would be prohibitive for all but the most exalted events, bearing the highest possible budgets, and unlikely to happen except in the most rarefied of circumstances.
A recent medical conference, taking place in several countries simultaneously, did exactly that. Orchestrated by Lee Pepper, who could accurately be called a super-user of Lentini’s original Software Audio Console, the convention included real-time surgery performed on live patients in numerous locations at the same time. SAC provided the ability for the attending doctors to interact with one another, via Internet connectivity, during these mission critical procedures. It’s a brave new world.
AIR president Damon Gold deploying SAC on a touch screen.(click to enlarge)
Freedom To Move
Though there’s no requirement that an AIR console be controlled by a wireless tablet or wireless laptop, (you can always use an Ethernet cable, if you like), it can be highly advantageous to do so. The ability to roam freely throughout a venue and tweak zone levels and zone EQ on each of the 72 outputs, gives the engineer a whole new palette of capabilities. For the first time, he/she will really know what’s going on throughout the venue during the show, instead of having to rely on comments from others.
Freeing up valuable seating space is another attribute, and one that’s bound to make promoters happy. Not only is it possible to reclaim seating space taken up by the console itself, but eliminating the security barriers and security guards that are normally needed to surround the house mix position, can represent a significant cost savings.
An AIR console footprint is insanely small: two shoes walking. And because audio never passes though the laptop or tablet, the remote devices merely communicate control signals. Therefore, a failure in the wireless link will not bring the system down - it will always remain where it was last set.
And in lieu of tablets and laptops, AIR makes various sizes of touch screens available – up to 32 feet - for those who wish for more tactile control, or who may not benefit from the ability to move throughout the venue. Shows such those that take place in TV studios, or a theatrical environment, are two of many such examples.
One comment that came up repeatedly while researching this article: performing artists absolutely love having their monitor engineer walk on stage and stand next to them, tablet in hand, to adjust the stage monitors to their individual taste and satisfaction.
And from the monitor engineer’s perspective, he/she can finally hear and experience exactly what the artist herself is experiencing. As hard as you try, second guessing the effect of a Marshall Stack behind the guitar player, is a guess at best. Hand signals are no longer needed, so sound checks move along faster and with far better results.
While cabling is still required to connect microphones to the head amps, which typically will be located at the side of the stage adjacent to the monitor engineer (if a monitor engineer is present), there is no longer a need for expensive splitters and multi-core cabling. This aspect alone saves hundreds of hours of labor over the course of a tour; reduces the size of truck space (or even the size of the truck itself); and eliminates tens of thousands of dollars of rental or purchase expense.
Band On The Run
Every AIR console comes equipped with Bob Lentini’s award-winning SAW (Software Audio Workstation), which works in the background to capture the inputs, the groups, the outputs, audience reaction mics, or whatever else you tell it to, up to 72 tracks.
The resultant recording can be used for a virtual sound check at the next gig, or it can become a full-fledged album release in its own right. You can overdub flubbed vocals at a later date in the studio, use an auto-tune plug-in, or leave it as raw as you wish. And speaking of plug-ins, the SAC platform supports VST, which means that a vast range of effects can be applied to one, or as many, virtual consoles as desired.
While the band’s en-route to their next destination, the engineer and producer can be busy mixing-down last night’s performance for Internet release, archival purposes, tomorrow night’s virtual sound check, the next day’s radio broadcast, or any other artistic or commercial requirement.
Those who desire physical faders instead of virtual ones, can interface an AIR console to a wide selection of commercially available control surfaces. Up to 256 faders, buttons, and rotary controls can be mapped to inputs, outputs, Aux sends, EQ and so on. Future AIR products will include dedicated fader banks that can be combined with PC tablets, bringing the best of both formats to users.
Though initially it can take some effort to get one’s head wrapped around the new concept of an AIR console, the payoff is likely to be worth the learning curve. The issues are no different than when the first wave of digital consoles originally entered the market.
Leading companies like Yamaha, Digidesign, DiGiCo, Soundcraft, and others, quickly realized that the key to selling their products was to help users become as comfortable as possible with the new paradigm of digital workflow, in as short a time as possible. AIR consoles realizes this as well, and has built its plans and has followed suit.
“Training is of paramount importance,” AIR’s Glen Campbell notes emphatically. “In this early phase we still provide hands-on assistance to every customer. This is essential for us to help our customers get up to speed as rapidly as possible, as well as for us to learn exactly what’s needed among our clientele, while we seek to broaden our scope of activities.
“In the near future, we’ll be introducing a variety of educational media, while we continually add product specialists and support personnel to our staff. Our customers must have a clear path available to them, in order to realize the full potential of this unique technology,” Campbell states. “This is essential for them to take full advantage of the unique capabilities that AIR consoles provides,” he concludes.
Ken DeLoria is senior technical editor for Live Sound International, and at one point during his highly accomplished career in pro audio, he served as worldwide director of live sound to launch Avid’s VENUE digital console.