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Interconnect
Monday, April 02, 2012
Reidel Introduces Suite Of AVB products For Artist Digital Matrix Intercom Platform
Reidel Communications has premiered a suite of AVB products for the Artist digital matrix intercom platform. AVB allows for transporting AES3/EBU audio in real-time with guaranteed bandwidth and quality of service (QoS) via Ethernet-based Local Area Networks (LAN). The new products were unveiled at the recent Prolight+Sound 2012 show in Frankfurt.
Based on official IEEE next generation Ethernet standards like 802.1Qav, P802.1Qat and P802.1AS, AVB allows risk-free utilization of AVB compliant facility or enterprise LAN infrastructure for intercom applications. This allows for new approaches in system and facility design providing significant savings in infrastructure investments.
Intercom applications for Riedel’s AVB products feature matrix-to-control panel connections via LAN, audio distribution via LAN, matrix-to-matrix trunking connections via LAN and distribution of digital partylines via LAN.
The Riedel suite of AVB products includes the AVB-108 G2 Client Card as well as the Connect AVBx8 panel interface.
The AVB-108 G2 card is a regular Artist client card to be used inside the Artist mainframe. It converts eight Artist matrix ports into AVB and vice versa.
The AVB-108 G2 client card communicates either with other AVB-108 G2 client cards in another Artist systems, e.g. for trunking, or with Riedel’s Connect AVBx8 panel interface.
The Connect AVB Panel Interface is a small unit, which converts an AES signal into AVB and
Connect AVBx8 converts eight AES signals to AVB and vice versa. Build in a compact 9.5”/1RU housing the device provides eight CAT5 ports to connect up to eight Artist control panels in one or two-channel mode to the matrix via IP-based LANs. Connect AVBx8 is the perfect team mate for Riedel’s AVB-108 G2 eight channel AVB client card.
Thirteen Audinate Dante-Enabled Products Introduced At Prolight+Sound 2012
At the recent Prolight+Sound 2012 show in Frankfurt, eight Audinate OEM partners announced a total of 13 Dante-enabled products.
“What is truly amazing is the wide range of different Dante products that were launched at the show”, says Lee Ellison, Audinate CEO. “It seems that no matter what you are looking for, there is now a Dante-enabled product on the market.” Lee also adds, “The market is dictating what it wants, and Dante seems to be the answer,”
Here is what was launched at this year’s show:
—Yamaha Commercial Audio Systems announced the launch of the new CL Series of digital consoles. The Yamaha CL Series is a Dante network–based console featuring remote I/O for a faster, more responsive Yamaha system solution. All three CL models in the Centralogic series, only differentiated by frame size and input capability, feature 24 mix buses, 8 matrix buses, plus stereo and mono outputs, and 16 DCAs. The footprint of all three CL consoles is small, yet powerful and has been developed specifically for sound reinforcement applications such as performing arts venues, theaters, houses of worship, touring, and remote broadcast.
—Symetrix announced the SymNet Edge, a significant update to its long line of DSP solutions packed with new features. Symetrix designed Edge to meet the I/O capacity requirements of the bulk of commercial sound installations. Edge features four configurable I/O card slots, up to 16 channels total of local I/O plus 128 (64x64) channels of Audinate’s award winning Dante network audio.
—Harman Soundcraft is extending its range of digital audio transport option cards for its Vi, Si Compact and Si1/2/3 Series digital consoles with all options expected to start shipping this year. Plans are also in place for Dante network cards compatible with Vi, Si Compact and Si1/2/3 Series consoles, following an agreement reached between Soundcraft and Audinate. “This agreement is a reflection of Harman’s responsiveness to its customers who want Dante networking,” adds Ellison.
—NEXO announced the new NXDT104 Dante audio plug-in card for the NXAMP, enabling NEXO loudspeaker systems to explore the many benefits of a high-performance AVB-ready digital networking solution. The NXDT104 will distribute digital audio plus integrated control data, automatically configuring its network interface and finding other Dante devices on the network.
—AuviTran, the networking specialists, is adding Dante technology to the Audio Toolbox product range. Audinate was chosen to accelerate AuviTran’s development towards new networking technologies. Since 2003 AuviTran has provided audio professionals with innovative networking solutions using EtherSound technology and now adds Audinate’s Dante networking to its networking solution portfolio.
—JoeCo Limited introduced the latest product in its BlackBox range of multi-channel live audio recorders and players in the form of the new BLACKBOX BBR64-DANTE RECORDER that can record or replay 64 channels.
—Four Audio, manufacturer of audio and loudspeaker management systems, announced the launch of the new DBO1, a Dante-enabled breakout box with 8 analog output channels. It allows the user to go from Dante to analog signals in a small 1U box. The DBO1 is operated by simply connecting it to a Dante network and configuring the easy-to-use breakout box via the Dante Controller.
—TEQSAS, a professional audio solutions company with core competencies in delivering high quality audio products, announced a new Dante AES breakout module for the cyberTEQ - m-family of digital audio interconnect solutions.
Understanding Specification Sheets: What Do The Charts & Graphs Really Mean?
The main purpose of specifications is to allow us to make sure that we have the right tool for the job. But what does this information really mean?
For the majority of humans, there is nothing simpler than listening to sound. You simply, well, listen.
When it becomes necessary to describe the listening experience analytically, however, a host of complex equations and diagrams are required to describe even the simplest of sonic events.
The benefit of mathematical analysis is that it can yield insights that are not apparent through intuition alone.
Acoustic signals are easily measured, and the audio components that produce them have characteristics that can be measured.
We do not expect specifications to tell us how a product sounds. This is what listening is for.
The main purpose of specifications is to allow us to make sure that we have the right tool for the job, and this information is most often presented in the form of charts and graphs.
But what does this information really mean?
Variables
The heart of understanding the specification sheets that describe audio products is the understanding of dependent and independent variables.
The concept is one that most people use every day, though often without realization.
An independent variable is one that describes a series that has a fixed value.
For example, the time of day in the city that you live in is an independent variable. Regardless of what happens tomorrow, time will progress like it did today.
What will change are your moment-to-moment activities. These events represent a dependent variable. They depend on time.
If you look at a page in your day planner, you are looking at a plot of activities vs. time.
Time is the independent variable. It is the same on every page of the planner.
The scheduled events are the dependent variables, because where you go and what you do depends on what time it is. Most graphs show the relationship between dependent and independent variables.
Now let’s look at a variation on the theme. Let time be the independent variable (it usually is) and let the loudness of the sound system during a show be the dependent variable.
The plot might look something like Figure 1.
Figure 1: In this example, time is the independent variable while loudness is the dependent variable (click to enlarge)
The horizontal axis represents time (the independent variable) and the vertical axis represents loudness (the dependent variable).
We will call the horizontal axis the x-axis and the vertical axis the y-axis, although any two letters would do.
The values on each axis are usually discrete, meaning that they are individual samples, points, or measured values called data points.
The fact that most graphs look like squiggly lines just means that after many data points were taken, they were joined with a line to make it easier to read.
Such two-dimensional plots are found on virtually every good specification sheet in existence. They simply answer the question “What is the value of y when the value of x is this?” Some examples of two-dimensional plots found in audio engineering include:
Y-Axis————————————————X-Axis
Amplitude—————————————- Frequency
Impedance————————————- Frequency
Directivity—————————————- Frequency
Phase———————————————- Frequency
Amplitude—————————————Time
Level————————————————Time
Each plot shows the value of y for a given value of x. Pretty cool. In math-speak, in each case it can be said that y is a function of x. (We sound smarter when we say it like this.)
From this example, it can be seen that frequency is a very common independent variable in the world of audio and acoustics. The y parameters are said to be frequency-dependent.
In audio and acoustics, almost all parameters that we care to know anything about are frequency-dependent. This means that the answer to virtually any question regarding any of the y parameters is “it depends.” Y depends on x.
An example of a frequency-dependent parameter is the setting of a graphic equalizer. In fact, it’s a really good example because it is basically an xy plot of the type that we have been describing.
The x variable is frequency, and the y variable is relative level. The y value depends on the x value.
When you look at the front panel of a graphic equalizer, you are looking at an xy graph, which is why it’s called a graphic equalizer.
What Time Is It?
Another common independent variable is time. Many parameters in audio and acoustics are time-dependent. Examples include loudness, temperature and background noise, just to name a few.
Note that Figure 1 just gives us values. It’s still up to us to know what they mean and how to apply them.
Graphs are valuable because they give us some visual feedback regarding trends in the data. For instance, a glance at Figure 3 (later in this article) shows that the loudspeaker’s on-axis directivity is increasing as a function of frequency.
This means that everyone in the room might hear the low-frequency events, like a bass guitar, but only those in front of the loudspeaker will hear the high-frequency events, like the crash of a cymbal.
It’s clear why we would want the directivity of a sound reinforcement loudspeaker to be “frequency-independent.” The directivity of such a device would be a straight horizontal line.
It’s also important to consider the resolution of the graphed data. The closer together we place the points on the x-axis, the less likely it will be that we missed a significant data point when we measured.
For example, we could take the page of a day planner and break the time axis down into hours, minutes, seconds, or even fractions of a second.
Obviously, there is a point of diminishing return on resolution. It must always be appropriate for the data being plotted.
If you were plotting the arrival time of the tweeter in the main array to the back of the balcony, then one millisecond resolution would be meaningful.
But that same resolution would be extreme overkill for plotting your daily schedule.
What time resolution do I need? Again, it depends!
Following are some examples of common plots found on data sheets, with plain English descriptions of what each one means.
After digesting each, download some data sheets from various manufacturers and attempt to interpret them.
Use them to form an understanding of the product, what it does, and how it might compare to a similar product.
Remember that to fully describe the performance of a product, and infinite number of graphs would be required.
Most “one-number” ratings in audio and acoustics have little meaning.
They usually over-simplify something that is much too complex to specify with a single number.
Unfortunately, many people base their gear-buying decisions on this meaningless data, and then wonder why the gear does not live up to their expectations.
A graph is much better, but even graphs can’t tell the whole story.
We live in an amazingly complex world!
Figure 2: The frequency response plot answers the question “What is the relative on-axis level change of the device-under-test regarding frequency?” (click to enlarge)
What’s The Frequency?
In Figure 2, the independent variable is frequency. The dependent variable is level. The frequency response plot answers the question “What is the relative on-axis level change of the device-under-test regarding frequency?”
For a device that produces the same level at every frequency, the plot would be a straight, horizontal line.
A real-world loudspeaker response is also shown. Some would consider a flat line response to be the best possible loudspeaker; however, a spectrum plot alone does not tell the whole story.
Now, let’s return to Figure 3.
Figure 3: At a glance, we can see that the loudspeaker’s on-axis directivity is increasing as a function of frequency (click to enlarge)
Again, the independent variable is frequency, while the dependent variable is the on-axis directivity.
The directivity plot answers the question “What is the ratio between the sound intensity on-axis to the total radiated sound intensity as a function of frequency?”
Q = 1 means that the device is omni-directional, where Q = 10 means that the intensity on-axis is 10 times the average radiated intensity.
Q = 100 means that the axial intensity is 100 times the average intensity.
Another way of describing the same thing is to use the directivity index, which is the Q rating converted into decibels with the formula DI = 10logQ.
It yields the same information in decibels, giving the loudness advantage produced by controlling the sound radiation.
DI and Q are often found on the same plot.
Turning our attention to Figure 4, once again the independent variable is frequency.
Figure 4: The impedance plot shows the opposition produced by the loudspeaker to current flowing from the amplifier as a function of frequency (click to enlarge)
The dependent variable is impedance.
The impedance plot shows the opposition produced by the loudspeaker to current flowing from the amplifier as a function of frequency.
A large peak on the curve means that less current is drawn at the frequency of the peak. This can happen at frequencies where the loudspeaker system is resonant, i.e. vibrates naturally.
Other frequencies require much more current to produce the same sound pressure level. Low spots on the curve represent frequencies where maximum current is drawn from the amplifier, i.e. where the amplifier is under a greater load.
The low values should be used when determining the required gauge of loudspeaker wire that should be used, or how many loudspeakers can be run in parallel.
Impedance is also required to calculate how much amplifier power is delivered to the loudspeaker, which in turn allows the loudspeaker’s power handling limits to be assessed. This is a good example of where a single number impedance rating (often called the nominal impedance) serves as little more than a guideline.
The impedance plot paints a much better picture of impedance and the other ratings that come from it.
Always remember to use specification sheets for what they’re intended – determining the suitability of a product for an application.
They are not a substitution for listening and measurement when evaluating products and should not be the final word in the buying decision.
A famous physicist once said, “The data on a spec sheet may be the best data they ever took or the only data they ever took!”
Pat Brown teaches the Syn-Aud-Con seminars and workshops. Synergetic Audio Concepts (Syn-Aud-Con) has been a leader in audio education since 1973. With nearly 15,000 “graduates” worldwide, Syn-Aud-Con is dedicated to teaching the basics of audio and acoustics. For more information visit their website.
Tour De Absurd: Unbound By The Fundamental Rules Of Reality
Everybody's dealt with horrible vendors from time to time, and Sully's got some tales from the road to which we all can relate.
I was just bitten by a dog.
Truthfully, not 20 minutes ago when I went to pick up a piece of gear at somebody’s house.
When I pulled into the driveway, an unholy spawn of a late night dalliance between Benji and the Geico gecko waddled over to me, growled, then bit me on the f***ing ankle.
I screamed like a five year old, which somehow triggered the garage door to open and spew a teenage girl carrying the gear I was there for.
“This yours?” she piped. “Your dog just bit me on the f***ing ankle,” I squeaked. “”Really? Sorry…” She froze with a look on her face that indicated she was now invisible and I should leave wondering where’d that girl go?
Needing more satisfaction I called the owner of the house.
“Hello?”
“Your dog just bit me on the f***ing ankle.”
“That dog’s 13 years old, he’s never bitten anyone.”
“Oh. Cool. Never mind then.”
“You sure?”
“Hang on a tic, let me make sure my portable morphine drip isn’t on high. Nope, machine’s good… the f***er definitely bit me”
The vicious dog attack left me sulking about the hound’s total lack of fear and respect for me. Then I got mad at myself for sulking about not being feared by an arthritic Chihuahua. Skillfully, I managed to cram in a 30-minute session of bi-polar self-loathing and admonishment in the time it took to drive from the scene of the assault to our bus.
It suddenly occurred to me, as I stared down at the dog sticking out of my jeans, that this was a fitting coda to the four-week tour de absurd that I and the rest of my crew had just endured. During the preceding month, 90 percent of the production vendors we had met had attempted to convince us they alone were not bound by the fundamental rules of reality.
To prove this point, they had taken our advance phone calls, listened carefully to our requests, sagely reassured us all would be well… then rolled us over and tried to bite us on the neck when we showed up. Same deal as the dog. They looked us up and down and figured they could take us.
Act 1 Me: “Hey, how wide is this box?”
PA prestidigitator: “205 degrees for the long throw, 365 degrees for the downfill.”
Me: (Knowing it’s general admission) “OK.”
Act 2 The setting: a large field with bands of disgruntled raisins milling arrogantly about.
A2: “We’re ready, my lord. We are prepared for you to communicate with the magic box and give us the array angles for the sound system.”
Steak sauce: “I have spoken with the machine. It gives no advice today. You must have done something to anger it. Go now, butcher the factory program and burn the fatted DSP as an offering. Leave me.”
A2: “My liege, the troubadours will be upon us soon… Can you offer no wisdom for us to assuage their FOH knight?”
Steak Sauce: “Tell him…tell him… the sound will emerge crooked if you angle the speakers. Tell him flat… Yes, flat is best. Threaten to rub petroleum jelly on him and burn him as a witch if he questions you.”
A2: “You are indeed the wisest in the land.”
Act 3 Production manager motioning to four speakers hanging from swing chain flown with the aid of two winches off the front of a quad runner.
PM: “What da hell is that?”
Local vendor: “EV X-Array.”
PM: “No it’s not.”
Local vendor: “Yes it is.”
PM: “No it’s not.”
Local vendor: “Yes it is.”
PM: “No it’s not.”
Local vendor: “O.K., no it’s not. I bought an X-Array box and copied it.”
PM: “You mean EV X-Line. You copied X-Line.”
Local vendor: “Yeah, the big EV box. X-Array-Line.”
PM: “O.K., just so we’re clear…you pirated something from EV and call it X-Array.”
Local vendor: “Yeah. Sounds great too. Wanna see our V-Disc wedges?”
Act 4 The three principal characters enter upstage center and proceed downstage in slow motion, their movements reminiscent of Apollo astronauts bravely approaching an ill-fated capsule.
Bonded by an invisible energy, their gaze begins tracking the seventy-five degree seating angle until at last their eyes settle upon the top seat, 600 feet aloft. One holds a laser range finder and whistles quietly at the data it yields.
Their attention is suddenly diverted to the single horizontal row of two EAW KF750s stacked neatly on the stage deck. A small man rapidly approaches the group.
He is equipped with a large black belt dubiously supporting a brick-like walkie-talkie with a solid three-foot antenna fully extended.
The effect is not unlike a remotely controlled Hobbit. A roll of gray tape used to seal air conditioning vents dangles from his meaty wrist, and he is thrusting an irate digit at the tiny speaker array.
Small Man With Big Belt: “I don’t want to hear it! Them speakers cover front row to top row perfect. They’re 70 degrees up and down so we don’t even need to tilt them. Sounds exactly up there like it do down here. I don’t want any of your smart-alecky talk about math. We done it this way for 10 years and it sounds great. Now, welcome and go away, I mix the opener tonight and I gotta make sure they’re happy”.
Act 5 A man stands beaten, his feet loosely clutching the prefabricated stage. His attention is captivated by the scene unfolding before his weary blue intelligent eyes…Men of ill-advised employment are hoisting a large-format console by attaching a 1/4-ton drape motor to its top-riveted session handles.
They stand under it, marveling at the graceful way it swings in the cool breeze. Our hero calculates that when the first handle lets go, the desk will swing low, hijack a stagehand at it’s nadir and force him to ride it bareback halfway to the rafters.
As the console reaches it’s apex and the second handle shears away, the desk will immediately divest itself of it’s passenger and enter a vertical spin, 25 feet off the ground, shortly proving wrong the load-out adage, “gravity is your friend”.
Quickly, without remorse, the sad man dispatches an intern to the balcony with a bin of economy popcorn and two video cameras. Word must reach the outside world of the transgressions that have transpired here…
Act 6 Me: “What version of the prediction software are you using?”
Them: “Ashly crossovers. They’re out front.”
I own a cat that hides behind the drapes when in trouble. It sits perfectly still, avoiding all eye contact, staring straight ahead looking like a paisley tumor respirating below the front window.
She is so convinced of her sudden undetectability that I have no choice but to accept the fact that the curtains have spontaneously evolved a tail and I should look elsewhere for her.
I marvel at her ability to gaze directly into the face of truth and maintain plausible deniability. Like the vicious miniature wolfhound noted earlier, the cat has eyed me up and come to the conclusion that she’s got my number.
I’d start dutifully working on a complex about my lack of respectability within the various animal phyla, but I know from experience, it’s not just me.
Many of the band guys I run into step off of the bus in the morning with dingoes latched to their ankles. They all have stories that somehow involve PA and lighting vendors avoiding eye contact and hiding behind backdrops with only their five D-MAG lights sticking out.
Sometimes I’ll look into their eyes, pat their dogs and smile with them, offering these words of solace: “get your sun block out boys, we’re goin’ to Hell.”
Finale Me: “Two horns are popping red and two are green. Which is correct?”
System provider: “Which is better?”
Sully is a veteran live sound engineer and really has no clever off-hand remarks for this space at this time.
Advanced Small Console Techniques: Maximizing The Available Feature Set
What about those times when, for whatever reason, a big console is not available? That's the time for ingenuity and some special techniques
Everyone loves a big console.
Even when they’re small in size, like modern digital consoles, we favor consoles that have everything we need to solve any problem that may come up.
But what about those times when, for whatever reason, a big console is not available?
That’s the time for ingenuity and some special techniques that maximize the usefulness of the available feature set. Some people call these “workarounds,” but the term I like is “tricks.”
You know, like a magician. Here are some of my favorite tricks.
The XLR “Y” Cable
The one-female-to-two-male XLR “Y” cable is a powerful problem solver that should be in every audio person’s bag of tricks. It’s most common use is splitting an input to two channels of a mixer, using one channel monitors, the other for mains. This allows the EQ and processing used on one to not appear in the other.
It can be especially important if you need maximum gain before feedback in the monitors for a vocalist, or the input channel EQ helps with an acoustic guitar that’s giving you feedback issues. This trick can also be used for “wet/dry” channels, or if you need some sort of crazy EQ or effect for one part of a song and then quickly need to change back to normal.
Channel Insert As FX Loop
When a console is lacking a function, sometimes it can be found in a piece of outboard gear. For example, most effects units have a built-in mixing feature. When the aux sends are all used up, but you still want one more special effect on one more input, it’s a simple thing to insert the device into the input channel, and use the effect’s wet/dry mix control to vary the amount of effect.
No, it’s not ideal, but under battle conditions, you do what you must.
Direct Out As FX Send
The input channel’s direct output is not just for multi-tracking any more! An input channel’s direct output is another way to get signal to an effect that is only needed on one channel. This trick works well for things like a chorus on an acoustic guitar, or a reverb on the lead vocalist.
It’s best if the direct out is post-fader. A pre-fader can also be used as a direct out too, but you’ll have to keep a watchful eye (ear) on relative levels.
Audio Group As FX Send
So, you’re using four auxes for monitors and two for vocal effects - how are you going to get reverb on the drums?
Sure, the same ‘verb could be used for everything, but where’s the fun in that?
One trick is to assign the drums you’d like to have reverb on (I often leave out the kick drum) to a sub-group, and use the sub-group output as an effects send.
If you have a 2-in/2-out effects box, assign the drums to two sub-groups and you have stereo drums with stereo effects. Use the group’s insert loop and the mixing feature of the effects box, and you have it without the need for an effects send or return.
Pretty cool, huh? This trick can also be applied to vocals, horn sections, or any group of inputs.
Uncommon Insert Hardware
Everyone is familiar with dynamics processors and equalizers inserted into input channels. However, these are not the only useful tools you can insert into an input channel.
For example, many smaller mixers do not include a variable high-pass filter (HPF). The advent of digital loudspeaker processors finds a great many analog crossovers sitting around gathering dust. Just connect an insert send to the input of an analog crossover, then take the output from the crossover’s “high” output, and there you have it, a variable HPF.
Many common analog crossovers have a frequency range both above and below the frequency of a console’s switched HPF, which is usually 80 Hz to 100 Hz. If you don’t have an analog crossover laying around, the glut of these kinds of units on the used market makes them very affordable.
That 4-channel, 2-way crossover pulled from the old monitor system can now be four channels of variable HPF, all in a single rack space.
Sub-Mixers
Any conversation about small mixers should include sub-mixers.
Sub-mixers get a bad rap, but they are an excellent solution to many small mixer problems, the most common being not enough input channels.
The logical use for sub-mixers is on things that can be grouped together - things like drum kits, horn sections, or backing vocals.
With a stereo sub-mixer, you can pan some inputs left, some others right, effectively giving you two sub-mixers. Sending the sub-mixers outputs to input channels on the main mixer transforms those input channels into additional sub-groups.
If your main mixer has sub-groups, you can also patch in your sub mixer there, and save the inputs on the main mixer for other things.
Insert To Bypass Mic Preamp
It’s becoming increasingly popular for artists or their engineers to carry around an esoteric front-end device. These devices almost always contain a microphone preamp as well as some combination of EQ and dynamics processing.
More often than not, the line inputs of an inexpensive mixer is the mic input padded down to line level. This means the fancy preamp is being hooked up to the console preamp, which is one preamp too many.
A specially constructed tip-ring-sleeve cable can bypass the mixer preamp completely by using the channel insert jack. The preamp end of the cable is wired tip/sleeve, the mixer end of the cable is wired ring/sleeve.
Please note that a cable constructed this way is for use with the most common unbalanced inserts that are tip send/ring return. For a mixer that’s configured ring send/tip return, both ends of the cable should have a tip/ring/sleeve connector wired tip/sleeve.
These cables, and any other cable wired in a non-standard fashion, should be very well labeled!
Many Happy Returns
Console returns are very often left unused. Most of us prefer the stereo input channels and/or the mic input channels, and for good reason. They have EQ and full routing capability, so what’s not to like?
Once again, for things like drum kit reverb where you are going for a “room vibe” type of effect that will be set and mostly left alone, the often overlooked console return works just fine. These are good places to plug in sub-mixers too.
Go Forth And Mix
These tricks are ways to squeeze big console performance out of a small console (or two). Again, some of them are not preferred practice, but they allow you to say “sure, no problem” instead of “I can’t do that.”
Now go forth and mix - big console or small.
Dave Dermont is a long-time working live sound mixer and a moderator of the ProSoundWeb Live Audio Board (LAB).
Universal Audio Now Shipping Apollo Audio Interface With Real-Time UAD Processing
Get more of the latest news from the 2012 PL+S show.
Universal Audio (UA) has announced the worldwide shipment of Apollo, a new high-resolution audio interface that combines UA’s analog design heritage with UAD powered plug-Ins in a sleek, elegant recording system for Mac and PC.
“Apollo is the culmination of 10 years of analog and digital audio development here at UA,” says Bill Putnam Jr., Universal Audio founder. “In many ways, it’s brought the analog and digital sides of our company together. With Apollo, we’re delivering the sound, feel, and flow of analog recording with all the conveniences of modern digital equipment, including next-generation Thunderbolt technology.”
Coinciding with initial product shipments, Apollo is also on display at the ongoing Prolight+Sound 2012 show (booth C68 in hall 5.1), with live demonstrations by Fab Dupont (Jennifer Lopez, Mark Ronson, Santigold) and vocalist Liza Colby.
Apollo is now available in both DUO CORE and QUAD CORE processing formats with estimated street prices of $1,999 (DUO processing model) and $2,499 (QUAD processing model).
In addition, Apollo’s Thunderbolt Option Card will be shipping in summer 2012, with pricing TBD.
Note that Apollo is compatible with Mac OS X 10.6 and 10.7; Windows 7 support coming in summer 2012.
Connecting Unbalanced Outputs To Balanced Inputs—And Vice-Versa
Two key issues: different signal operating levels between consumer and professional equipment, and making connections while avoiding ground loop problems
Based on my years of helping customers solve interfacing problems of all sorts, connecting unbalanced outputs to balanced inputs, and vice-versa, certainly ranks among the most common and confusing of tasks for system integrators.
Basically, two issues must be dealt with. The first involves the different signal operating levels between unbalanced (consumer) and balanced (professional) equipment. The second involves making the actual connections to transfer the signal while avoiding “ground loop” noise problems.
Signal operating and reference levels are significantly different for consumer and professional equipment. The consumer reference level is -10 dBV or 316 mV rms, while the pro reference is +4 dBu or 1.228 V rrns. Therefore, a voltage gain (for consumer outputs driving pro inputs) or loss (for pro outputs driving consumer inputs) of 3.9 or about 12 dB is theoretically required.
On the consumer to pro side, a fair question might be “Why not use a step-up transformer for this gain?” Several commercial products do, but I don’t recommend them. Let me explain. Transformers simply reflect impedances from one winding to another - they do not have an intrinsic impedance of their own. Assume we use a transformer with a turns (voltage) ratio of 1:4 to get 12 dB of gain. This unavoidably (laws of physics) makes the transformer’s impedance ratio 1:16, the square of its turns ratio.
Therefore, the impedance of the pro input will be reflected back to the consumer output as 1/16 of that. Since a typical balanced Input has an impedance somewhere between 10 kiloohms (kohms) and 40 kohms, it will be seen by the driving consumer output as 625 ohms to 2.5 kohms. Virtually all consumer outputs are rated to drive a “10 kohms minimum load.” That’s because their internal or “output” impedance (usually unspecified) is typically 1 kohm or more.
Therefore, actual gain will not be 12 dB but only 3 to 8 dB because of the low load impedance on the consumer output. Worse yet, the consumer output will experience a serious headroom loss, up to 8 dB, causing premature clipping. Since most consumer outputs use coupling capacitors designed for a “10-kohm minimum load,” the severe loading will usually result in poor bass response, too.
Usually, specs relating to these issues are conspicuously absent from manufacturers’ data sheets. However, gain is seldom an important issue because pro equipment inputs generally have at least 12 dB of additional gain “reach.” If we eliminate the signal gain requirement, unbalanced to balanced interfaces become fairly straightforward,
On the pro to consumer side, operating level differences are an important concern.
Because consumer inputs rarely include input level controls, the consumer equipment is easily overloaded by pro signal levels.
Again, since the professional reference is +4 dBu or 1.228 V rms and the consumer reference is -10 dBV or 316 mV rms, a loss of about 12 dB is required.
Obviously, the output of the pro equipment could be reduced by 12 dB, but then its level metering would be nearly useless and signal-to-noise performance would be degraded.
Signal attenuation is required for these interfaces.
In most cases, noise rejection is a far more important issue.
For consumer to pro interfaces, the widely used hookup of Figure 1 uses shielded single-conductor cable and an RCA to XLR adapter or ready-made adapter cables built as shown.
Figure 1 - Using an unbalanced cable with an adapter results in zero noise rejection.
Unfortunately, it has 0 dB of ground noise rejection - and wastes all the potential noise rejection of the balanced input! Sadly, the availability of such adapters or cables leads many unwary users to create this noise-prone connection. Performance is especially poor when cables are long, since the entire interface is unbalanced, allowing both audio and ground noise to flow in the cable shield.
A far better hookup shown in Figure 2 uses shielded twisted-pair cable to take advantage of the noise rejection available from the balanced input stage.
Figvre 2 - Using balanced cable wired as shown results in at least 30 dB rejection.
Because ground noise now flows in the shield conductor rather than one of the signal conductors, noise rejection is improved by about 30 dB when the input is a typical “active” differential-amplifier type. If the equipment’s balanced input is truly high-performance, using an input transformer or the InGenius IC, rejection is improved by about 80 dB. [Reference 1]
Figure 3 shows noise rejection for various consumer to pro interfaces. The top plot at 0 dB represents the simple adapter and 2-conductor cable connection.
Figure 3 - Noise rejection in unbalanced to balanced interface. Top to bottom: cable of Figure 1, cable of Figure 2, cable of Figure 2 plus output transformer, cable of Figure 2 plus input transformer.
The plot at -30 dB shows the improvement due to the simple 3-conductor hookup. The next plot shows the effect of an ordinary isolator using an output transformer (no internal Faraday shield!). It improves 60 Hz hum by about 20 dB, but has little effect on buzz artifacts around 3 kHz. A high-quality isolator using an input transformer (with an internal Faraday shield) increases rejection to almost 100 dB at 60 Hz and about 65 dB at 3 kHz.
For the best possible noise rejection, use a 3-conductor cable (wired as in Figure 2) from the unbalanced output to the isolator input. The plots here were measured with a 600-ohm unbalanced output and a 40-kohm balanced input.
For pro to consumer interfaces, rejection of ground noise is also very desirable.
Figure 4 (below) shows noise rejection for various balanced to unbalanced interfaces, The upper plot at 0 dB represents a direct connection, such as for an adapter or adapter cable.
Direct connections are problematic because various types of balanced output circuits are used in equipment each with its own peculiar limitations.
Some, such as the one in the schematic, can be damaged if one of its output terminals is grounded. Outputs stages using either transformers or widely used “servo-balanced” outputs, must have one terminal grounded in order to produce a proper output signal at the other. But the “servo-balanced” output can oscillate or become unstable if the ground connection is made at the far (receive) end of a cable. [Reference 2]
Figure 4 - Noise rejecfion in balanced to unbalanced interface. Top to bottom: direct wiring, connection with output transformer, connection with input transformer as in Figure 5.
Therefore, a cable that works with one piece of equipment may not work with another. Fortunately, adding a transformer allows the interface to work with any output stage. The middle plot shows that an output transformer reduces 60 Hz hum by about 50 dB and buzz artifacts around 3 kHz by about 20 dB. A high-quality input transformer, increases rejection to over 105 dB at 60 Hz and to nearly 75 dB at 3 kHz.
When this transformer is a 4:1 step-down type, the 12 dB level difference problem is neatly solved as well. Figure 5 shows the system schematic of this “universal” pro to consumer interface.
Figure 5 - A step-down transformer works with any balaoced output.
Bill Whitlock has served as president and chief engineer at Jensen Transformers for more than 15 years and is recognized as one of the foremost technical writers in professional audio.
REFERENCES
[1] Whitlock, B., Interconnection of Balanced and Unbalanced Equipment, Application Note AN003, Jensen Transformers, Inc., 1995.
[2] Hay, T, Differential Technology in Recording Consoles and the Impact of Transformerless Circuitry on Grounding Techniques; Audio Engineering Society, 67th Convention,1986, Preprint #1723.
Yamaha Launches New CL Series Digital Consoles At Prolight + Sound 2012
Get more of the latest news from the 2012 PL+S show.
Retaining essential features and functionality that have become standards over the past quarter of a century, the new Yamaha CL Series of digital mixing consoles offers an evolved experience in accessible mixing. The CL Series was just unveiled at the ongoing Prolight+Sound 2012 show in Frankfurt.
The line-up comprises three consoles, the CL1, CL3 and CL5, ranging in scale from 48 to 72 mono plus 8 stereo inputs. All feature 16 DCAs and 24 mix/8matrix output buses.
Founded on the proven Centralogic interface, the CL Series incorporates multiple innovations and refinements, including enhanced Select Channel functions and User Defined knobs.
The CL Series EQ and effects have been vastly expanded. They include an Effect Rack with VCM analog circuitry modelling technology, as well as a Premium Rack that includes the Rupert Neve Designs Portico 5033 equalizer and Portico 5043 compressor/limiter, developed in close cooperation with Rupert Neve.
Two new rack-mountable I/O units, Rio3224-D and Rio1608-D, can be used in a variety of combinations and configurations, communicating via a scalable Audinate Dante digital audio network.
Up to eight I/O rack units can be connected to a CL Series console, while multiple CL consoles can share control of the same I/O rack unit. A new Gain Compensation function adds the ability to combine front of house and monitor control via a single network, for comprehensive digital live sound integration.
For live multitrack recording and virtual sound checking, CL consoles are equipped with dedicated recording control capabilities for use with Steinberg Nuendo Live DAW application running on a Windows or Mac computer.
The new Yamaha CL Series line-up. (click to enlarge)
Further Details
User interface. The Centralogic user interface ensures that the consoles will be immediately familiar to many thousands of live sound engineers.
It has evolved considerably on the new consoles, incorporating a new generation, highly responsive color touch screen and an array of user definable rotary encoders and buttons.
Newly designed faders offer optimum feel, visibility and accuracy and are freely configurable to allow control of any combination of inputs, outputs or the 16 DCA faders.
The control surface also provides editable, back-lit channel name displays above each fader, with assignable color bars. The CL1 and CL3 also feature the option of an external meter bridge.
The new Yamaha CL1. (click to enlarge)
Audio quality. Audio quality and character were top priorities in the development of the new consoles. In addition to featuring newly-designed mic preamps and delays on every input channel and output port, the range debuts a prestigious line up of studio quality processing.
The Effects Rack provides the equivalent of eight SPX2000 effects processors, along with a range of VCM EQs and dynamics, while two further virtual racks provide access to up to 32 channels of graphic EQs.
However, probably the most exciting sonic innovation is the introduction of the Premium Rack concept. Developed by Yamaha’s Dr K (Toshi Kunimoto) and his team, the Premium Rack provides a range of extremely high quality, dynamic processors and EQs.
Yamaha has collaborated closely with Rupert Neve to incorporate the acclaimed Portico 5033 EQ and 5043 compressor as key elements in this new concept. These processors are included as standard in the CL series, eliminating the need for any plug-in management.
“For the first time we have the capability of bringing Rupert Neve sound into the live audio field, entirely due to Yamaha VCM technology. I believe that it is indistinguishable from the original analog sound,” says Neve.
Scalable solution. A key factor in making the CL Series so flexible is the pair of accompanying I/O racks, the Rio3224-R and Rio1608-D, and the fact that consoles are the first to feature built-in Dante networking as a standard feature.
A scalable system is easily constructed by simultaneously attaching up to eight I/O racks via Dante, providing up to 256 input sources. Pairs of CL consoles can also be cascaded to handle larger mixing requirements.
Rio rack-mountable I/O units. (click to enlarge)
Connection of basic systems is easy, using the console’s auto-configuration facility. Two or more consoles can share the inputs from one set of I/O racks without fear of unexpected level changes due to the inclusion of Auto Gain Compensation within the I/O racks themselves.
With the new Dante 32-bit mode of operation, gain compensation can be provided without audibly affecting the dynamic range.
“We are extremely excited to be collaborating with Yamaha on the extraordinary new CL Series and I/O racks,” notes Audinate CEO Lee Ellison. “Dante provides a flexible, low latency, highly scalable, plug and play networking solution to connect Yamaha networked systems, Dante Virtual Soundcards or any other Dante networked device.
“We believe the combination of technologies integrated in this new platform will provide an unsurpassed digital experience.”
The Rio3224 also includes four stereo AES-EBU outputs, keeping signals in the digital domain right through to the amplifier. The new consoles offer three MY card slots on the rear panel, maintaining compatibility with every existing audio format as well as newer cards like the MY8-Lake speaker processing card and the MY16-Dugan auto mixing card.
The three card slots also allow for additional I/O alongside the consoles’ onboard eight mic inputs and eight line outputs.
The new CL5. (click to enlarge)
Software control. CL Editor is a new standalone online/offline editor which runs on both Macs and Windows PCs.
Featuring all the functionality that users will be familiar with from other Yamaha Editor software, it does not require Yamaha Studio Manager as a host.
Further new software applications include a new version of StageMix for iPad, which offers comprehensive wireless remote control and has been expanded to include new features such as channel naming, DCA fader control and tap tempo.
Crucially, both CL Editor and StageMix can be run simultaneously, allowing very flexible options for engineers, sound designers and system technicians alike.
Meanwhile, Yamaha’s new File Converter software has been upgraded to allow straightforward exchange of console files between the CL Series, PM5D, M7CL and LS9.
Recording. Every CL console customer will receive a copy of the new Steinberg Nuendo Live recording software, which has been designed specifically for live recording applications.
Further aspects include StageMix for iPad, the MY16-Dugan auto mixing card, and Nuendo Live. (click to enlarge)
Available from July, it includes unique features not found in any other live recording software and will be tightly integrated with CL consoles to provide optimum ease of use.
When combined with Audinate’s Dante Virtual Soundcard (also included with every CL console) engineers can easily use the Dante network to record up to 64 tracks of audio to either a Mac or Windows PC.
“Nuendo Live is not only ultra stable and easy to use, but it integrates seamlessly with the latest generation of Yamaha live consoles,” says Steinberg managing director Andreas Stelling. “This is another successful example of the strong ongoing relationship between Yamaha and Steinberg.”
In addition, basic stereo recordings can be done via a convenient 2-track USB recording and playback function.
CL5 console plus Rio I/O. (click to enlarge)
With the advent of the CL Series, the company is confident these products strike the right balance between innovative functionality and sound investment potential.
“The CL Series ideally answers today’s needs using today’s technology” says Kazunori Kobayashi, general manager of the Yamaha Pro Audio Division. “The collaboration between Yamaha, Rupert Neve Designs, Audinate and Steinberg has made it possible to deliver a sublime balance of sound, performance, and features that results in uncompromised overall mixing capability and quality. After a quarter century of evolution, the CL Series represents a momentous new chapter in the history of Yamaha digital mixing.”
The CL5 and Rio3224-D are scheduled to be released in the spring of 2012, with the CL1, CL3 and Rio1608-D available in the summer.
Soundcraft Extends Digital Console Networking Backbone With AVB, Dante & BLU Link Options
Get more of the latest news from the 2012 PL+S show.
Soundcraft is extending its range of digital audio transport option cards for its Vi, Si Compact and Si1/2/3 Series digital consoles with all options expected to start shipping this year.
Harman is one of the pioneers of AVB IEEE 802.1 and an active AVnu Alliance member, and with the standards now ratified, work has commenced to make AVB audio networking available on the Vi, Si Compact and Si1/2/3 Series consoles.
Plans are also in place for Dante network cards compatible with Vi, Si Compact and Si1/2/3 series consoles, following an agreement reached between Soundcraft and Audinate.
“This agreement is a reflection of Harman’s responsiveness to its customers who want Dante networking,” adds Lee Ellison, Audinate CEO.
To further enhance integration between Harman’s other Professional audio brands, option cards are also in development to provide a simple on and off ramp between Vi, Si Compact and Si1/2/3 Series consoles and the BSS Soundweb London digital audio bus system BLU link.
The BSS digital audio bus is integrated with the majority of BSS Soundweb London series products and with the PIP-BLU is available for a number of Crown amplifiers.
Existing expansion cards already available for Soundcraft digital consoles or stage boxes include Cirrus Logic, CobraNet, MADI, Aviom A-Net, AES, Digigram Ethersound, Riedel Communications RockNet, Alesis ADAT, SD/HD SDI and Dolby E.
Soundcraft product manager Richard Ayres states, “Connectivity is an important part of everyone in pro audio’s future; new products must integrate with legacy systems and infrastructure whilst existing products must adapt to emerging technologies. With these announcements we are making it clear that by investing in a Soundcraft console you are assured it will remain future-proof and core to an integrated system.”
Lynx Offers Thunderbolt Driver For Expansion Chassis Usage
Lynx Studio Technology has released a new Thunderbolt Driver which provides Thunderbolt connectivity for the company’s PCI Express audio cards.
When used with third party expansion chassis products from Sonnet Technologies and Magma, the new Lynx driver allows the AES16e. AES16e-SRC and AES16e-50 professional audio cards to utilize the Thunderbolt connectivity on newer Apple computers.
“This is our first step in supporting Thunderbolt interface technology,” states David A Hoatson, Lynx co-founder and chief software engineer. “These upcoming expansion chassis allow many of our existing and established products to be easily added to a Thunderbolt equipped computer.”
The new Lynx driver has been fully tested with the Sonnet Echo Express PCIe 2.0 Thunderbolt Expansion Chassis and Magma’s ExpressBox 3T for the Lynx AES16e PCI Express 16 channel AES/EBU interface models. Both units are expected to be shipping soon.
The Lynx Thunderbolt driver is available on the Lynx Studio Technology website.
When connecting loudspeakers: whatever the approach, impedance is the thing
Multiple loudspeakers can be connected in series or parallel to the output of the amplifier.
In either case, the current drawn from the amplifier is determined by the total impedance of the load as presented to the loudspeaker terminals.
Impedance is the opposition to the flow of current.
As the load impedance is decreased, the load on the amplifier is increased, because it must work harder to supply the demand for current. In similar fashion, an automobile trying to maintain its speed uphill is under a greater load than on flat ground.
A “no load “condition means that nothing is hooked to the amplifier, so no current flows and no power is transferred.
The opposite condition, a dead short between the amplifier “+” and “-” terminals, represents the maximum load possible, and current flow is limited only by the resistance of the wire making the connection.
So, the lower the impedance the greater the load - a bit counter intuitive but nonetheless true.
HOLE IN THE BUCKET
An example will clarify this. Imagine a bucket full of water.
Two loudspeakers in parallel require twice the current of a single one, just like two holes in a bucket offer one-half the opposition to water leaving the bucket as a single one. (click to enlarge)
Assuming watertight construction (a good thing for a bucket), there will be no water leaving the bucket (analogous to current flow), and the pressure against the sides of the bucket (analogous to electrical voltage) will be constant.
Now, let’s put a hole in the bucket. Water will now leave the bucket at a rate proportional to the size of the hole.
The hole represents the connection of a loudspeaker - current now flows from the bucket (amplifier) through the hole (load).
If we keep the hole relatively small, the pressure will be similar to the water-tight condition. If we replenish the bucket continually, the flow can continue indefinitely.
Next, let’s put another hole in the bucket, identical to the first.
Water is now exiting the bucket at twice the previous rate (more current is flowing), so the supply to the bucket would have to be increased to maintain the water level.
The second hole is analogous to a second loudspeaker connected in parallel with the first to an amplifier.
The load impedance (the total opposition to water leaving the bucket) is decreased, meaning that the replenishing supply must work harder to keep up.
The bucket represents the amplifier, the holes the load, and the replenishing supply is the AC cord that plugs into the wall. Since water is flowing in one direction only, the current is DC (direct current).
The same principles hold true for AC (alternating current), which in this example would mean that water is alternately flowing in and out through the holes in the bucket.
GETTING THE FLOW
Series connection means that the current flows through one voice coil before it flows through the other.
A look at series connection. (click to enlarge)
The applied voltage will divide between the two in proportion to the magnitude of their impedance. If their impedance is the same (the most common case), then the voltage will divide equally across the two. The same current will flow through both.
Series connection is usually accomplished by connecting the amplifier “+” to the “+” of the first loudspeaker, and the “-” of the first loudspeaker to the “+” of the second, and finally the “-” of the second to the “-” of the amplifier.
The total impedance of loudspeakers in series will be the simple sum of their individual impedances, so adding more loudspeakers will decrease the load on the amplifier.
In other words, the higher the load impedance, the lower the current demand on the amplifier, and the less power delivered to the load.
Series connection is rarely used in multiple loudspeaker systems since adding a loudspeaker will change the power flow (and loudness) through all of the loudspeakers. If one loudspeaker opens up, the feed to all of the loudspeakers is lost.
SAME ACROSS EACH
Parallel connection means that the amplifier output current flows through both of the voice coils simultaneously (a current divider), in proportion to their impedance.
If they are the same impedance (the most common condition), the current through each will be the same.
The output voltage of the amplifier will be the same across each voice coil, since all “+” terminals are connected together and all “-” terminals are connected together.
The load increases (the total impedance gets smaller) as more loudspeakers are connected in this fashion.
Parallel connection is the preferred method for configuring a multiple loudspeaker system, because adding additional loudspeakers does not change the power flow (or loudness) through the existing loudspeakers.
The sound level from existing loudspeakers remains the same as additional loudspeakers are added.
Parallel connection, the preferred method for multiple loudspeaker systems. (click to enlarge)
Care should be taken to avoid overloading the amplifier - a condition that occurs when too many loudspeakers are paralleled. This produces a total impedance that is too low and draws excessive current from the amplifier.
When loudspeakers are “daisy-chained” they are being connected in parallel.
The interconnecting cable buses all of the “+” loudspeaker terminals together and all of the “-” loudspeaker terminals together.
This is often confused for series connection, but it is not since the current does not need to flow through one loudspeaker to get to the next.
Series-parallel is useful within loudspeaker enclosures to allow a target impedance to be achieved with multiple devices. (click to enlarge)
An open voice coil in one of the loudspeakers will not produce a level change in the remaining loudspeakers, making this configuration ideal for distributed ceiling loudspeaker systems.
Series-parallel connection combines both of the above and is useful within loudspeaker enclosures to allow a target impedance to be achieved with multiple devices (i.e., dodecahedron loudspeakers, guitar amplifier cabinets, etc.).
It is sometimes employed to achieve an impedance value that would otherwise be too high if series connection alone were used, or too low if parallel connection alone were used.
This method works fine when used internally in a loudspeaker box (like a six 10-inch-loaded “guitar cabinet”), but should be avoided if the loudspeakers are to be distributed around a facility.
Such a system would be difficult to expand and difficult to service due to the non-standard method of connection.
Pat and Brenda Brownown and operate SynAudCon, conducting training seminars around the world.
QSC Audio Q-Sys CCN32 CobraNet Audio I/O Card Now Available
QSC Audio Products has announced that the new CCN32 CobraNet Audio I/O Card is now available, expanding the offerings of the Q-Sys system for audio system control and signal transport.
The CCN32 enables system bridging between Q-Sys and a CobraNet legacy platform. The card can provide 32 I/O channels when installed in a Q-Sys Core and 16 I/O channels when installed in an I/O Frame.
Designed specifically for those customers who have an existing CobraNet system installed, the CCN32 provides a very cost-effective and seamless means for a phased approach to bridge between the legacy system and the advanced technology Q-Sys platform.
Q-Sys, the complete integrated system platform that encompasses everything from the audio input to loudspeakers, provides all the audio routing, processing, control and monitoring necessary for any facility.
It is utilized worldwide in a variety of applications, including both large and small, including stadiums and arenas, attractions and theme parks, performing arts venues, transportation hubs, legislative and judicial chambers, hotels and casinos, bars and restaurants, houses of worship, corporate campuses, and educational facilities.
Percussion In The Dark: Nothing A Couple Of LEDs Can’t Fix
Bringing together technology to help foster a new and creative concert experience
Ever wonder how to cue an ensemble in the dark? Just grab a few 220-ohm resistors and some light-emitting diodes and you’re halfway there.
Shenandoah University in Winchester, VA made a mark on innovation and ways to experience music as an audience member with a recent percussion ensemble performance of “Pika-Don” by American composer James Tenney.
The piece is a musical representation of a percussion ensemble paired with prerecorded voiceover tracks of writings taken from the scientists at the first atomic bomb test, and from the survivors of the Hiroshima bombings.
Professor Earl Yowell, the director of the ensemble, came to adjunct professor Mike Sokol with the desire to perform this piece, but lacked the technical knowledge to accomplish the task.
“When I discovered Pika-Don (“Flash-Boom”) by James Tenney, two thoughts came to my mind,” Yowell explains. “First, I have to do this piece with the SU Percussion Ensemble, and second, I have no idea how to make the technological demands work. When the piece was written the technology was very different than now. Instead of, for example, a [Apple] Logic editing program running on a computer, there was a 4-track analog tape.
click to enlarge
“The composer suggested that the performers use four synchronized stop watches,” he continues. “The process of coordinating the quadraphonic tape, stopwatches and the performers seemed clumsy at best. Deciding to use an audible click track was the first option, but the players also needed to be able to hear the pre-recorded voices.
“So, Mike Sokol came up with the idea of an (LED) light click track. Instead of an audible click, the performers would be cued visibly with a flashing LED. The timing for the flashing would be provided by a fifth audio track that contained some kind of sound that would be used to trigger the LEDs.
“I chose a gong for that sound because its sharp attack would light the LEDs with a definite on-beat every five seconds and the gong sustain would fade out the LEDs gradually over a second. Mike and his class did a wonderful job building, soldering and bringing together all of the technology that made for a flawless performance and gave the audience a new and creative concert experience.”
click to enlarge
Sokol took his light click track idea to the live sound practicum students in the Music Production and Recording Technology program at Shenandoah University, and they were able to build a system to not only cue the four percussionists in the dark auditorium of Shenandoah’s concert hall, but also have the pre-recorded tracks running from a Logic session on a MacBook Pro computer at the same time from four different loudspeaker locations surrounding the audience in the auditorium.
The setup was borderline Frankenstein’s monster, but it went over without a hitch. The power source for this monster was two Crown Audio Power Base power amplifiers wired to four Community CPL monitor wedges. The wedges were faced toward the auditorium walls so that the voiceover tracks were diffused before they reached the audience, which lent itself to the overall effect of how the bombing survivors felt during their experiences.
The Logic session was run on a MacBook Pro via Firewire feeding a MOTU 828 I/O box whose outputs 1-4 were routed with TRS/DB-25 connections through a Whirlwind 5.1 PA Precision Attenuator. The signal was then routed down to the power amplifiers in front of the stage and finally out to four cables sending signal to the monitor wedges. The Whirlwind 5.1 gave the ensemble director—who was sitting in the middle of the audience—the overall gain and volume control of the voiceover tracks so that they were balanced throughout the hall, but did not affect the LED cueing mechanism in any manner.
The ensemble needed to be cued hundreds of times at 5-second intervals throughout the piece, but Yowell wanted to maintain the dark, solemn atmosphere while they played. Enter the LEDs. Sokol’s prototype proved to be what was needed for the performance to run smoothly.
The practicum class jury-rigged four female XLR connectors with 220-ohm resistors soldered to pin 1, the LED soldered to pin 2, and a solder joint connecting the resistor and LED in series. Since an audio signal is AC (alternating current), the polarity of the LED hookup doesn’t matter at all for this application. (Please contact Mike Sokol via e-mail .(JavaScript must be enabled to view this email address) with any comments or questions about this technology.)
Creating the LED cue light was half the battle.Getting the LEDs to work as an actual cueing system was the next step.
The ensemble director created an isolated audio track in the Logic multi-track session that played gong hits, which became the LED visual cue signal at 5-second intervals for the players.
click to enlarge
This “visual” cue track was internally mapped to the channel 5 output of the MOTU 828, which bypassed the Whirlwind attenuator by running directly into a Marantz MA-5 Esotec 30-watt power amp.
The Marantz amp output was then connected to a Whirlwind 1x6 Parallel Line Splitter via a male XLR “kludge” cable connecting the amplifier output to pins 1 and 2 of the XLR, and the cue signals were run to the percussionists’ cue LEDs via XLR cables coming from the Split-6.
The LEDs were gaff taped onto the players’ stands below their music so that they had view of the LED cue light, but the space to move freely around their stations.
Sokol has also designed and built a “stereo” version of this same LED cue system that could be driven by two audio channels with different color LEDs over a common XLR cable.
He notes that this LED assembly can be driven directly by a 30-watt power amp, or any decent headphone amp. A headphone amp might only be able to drive a single LED, but a 30-watt power amp is capable of driving dozens of these LED assemblies in parallel.
click to enlarge
This system would allow the director or sound tech to patch two audio channels into the same LED cue box, with, for example, one color LED for the snare channel and the other for the kick channel.
That way, if desired, other musicians in sonically-challenged performance spaces could be visually synchronized to a live drummer or recorded click track. However, for Pika-Don the director thought that a single red LED would be sufficient, which did indeed prove to be the case. Simple is beautiful.
All that was needed to begin the performance was for the director to press the play button on the laptop and sit back as the percussionists watched their LEDs for the cues.
He then tweaked the volume levels of the quad playback tracks to match the live percussionists’ volume.
The technology behind a piece should never take precedence over the piece itself. The sensitivity and solemnity of “Pika-Don” was the greatest concern when creating this monster of a contraption, and was not overshadowed due to the musicians LED cueing system.
Plus, having four channels of pre-recorded dialogue synchronized and merged with four live percussionists was totally awesome.
The director was free to watch the score while the gong track gave the LED cues, and the audience had no distraction from the piece since there were no obvious hand cues being thrown around by the director. The technology behind the piece was a success and the integrity of the piece was maintained, which is all one can really ask for when creating a new performance medium.
click to enlarge
According to Sokol, “We must always remember that any special stage gadgets we build are but tools for musicians and directors to present the best possible performance to the audience. Sometimes technology overshadows the art form, but in this case it appears that we hit a home run.
“It was so seamless as to be practically boring to operate, and the technology just melted into the background. That’s exactly how it’s supposed to be.”
Author Courtney Roebuck is a junior Music Production and Recording Technology major at Shenandoah University [Winchester, VA]. She has experience with both studio and live sound settings of recording, but she finds a special interest in live sound and innovative ways to record in a live setting. H.G. La Torre of Fits & Starts Productions provided editing; photos were supplied by Madalyn Newell, and graphics by Mike Sokol.
Sommer Cable Unveiling Cardinal DVM120 Series Audio Tools At PL+S 2012
Get more of the latest news from the 2012 PL+S show.
At the Prolight+Sound 2012 show in Frankfurt, Sommer Cable is presenting the new Cardinal DVM120 Series, which includes seven audio tools for professional live and studio applications.
Devices include:
DVM-120-TDI Tube Direct Box. The combination of a very high impedance input and a high-class buffered valve stage takes care of very sensitive signal sources in an optimal manner. So, for instance, the high-Z input will prevent the overloading of the pickup output.
The input has two bridged 6.3 mm jack sockets plus a 1:1 valve-buffered output jack socket. Parallel a balanced and electrically isolated low-impedance signal is output at the rear XLR socket via a Lundahl transformer. With the gain switch, the gain factor is selected. A ground lift switch for the input section and a ground-on switch for the XLR output help to prevent ground loops.
DVM-120-HZDI High-Impedance Dual Direct Box. A simple way of adapting sources to microphone inputs. It includes quality Lundahl transformers with 22 dB or 28 dB attenuation, which are suitable for high input levels while offering low distortion and neutral response - even down to sub-low frequencies.
On the input side, each channel has two bridged jack sockets; if necessary, the input and output grounds can be linked with the ground lift switch. The output signals are on male XLR connectors.
DVM-120-DDI Dual Direct Box. For easy, reliable matching of medium- and high-impedance sources to microphone inputs. The Lundahl transformers offer 17 dB attenuation combined with low distortion and neutral response.
On the input side, each channel has two bridged jack sockets; if necessary, the input and output grounds can be linked with the ground lift switch. The output signals are on male XLR connectors.
DVM-120-DLI Dual Line Box. The Lundahl 1:1 line transformer used in this dual-channel device was custom-developed for applications with very high levels. So, for instance, it is suitable for the electrical isolation of active PA components and high-level line sources.
Input signals are fed either via 6.3 mm jack or female XLR sockets. Both channels have discrete ground lift switches to disconnect input and output grounds as required. The output signals are on male XLR connectors.
DVM-120-HPA Variable Phones Amp. A dual-channel amplifier with XLR inputs, tone control and level control provides the basis for the main path. There is an additional auxiliary input with an XLR/jack combi socket which can be mixed to the main signal using a dedicated control.
The stereo switch is used to switch between stereo and mono modes. In stereo mode, the sub/pan control is used for setting the balance of the main signal. In mono mode, the aux and sub inputs are mixed with the main path via their own controls – turning it into a compact three-channel mixer.
Headphones can be connected on the front via two 6.3 mm jack sockets. All inputs are electronically balanced and can be switched between +4 dBu and –10 dBV nominal level. DVM-120-SPS Remote Controlled Speaker Selector, DVM-120-SPR Remote Monitor Controller. These dual-channel units serve to connect two loudspeaker pairs with one monitoring output – a typical studio situation. The switching of the line level signals before the active loudspeakers or power amplifiers is done in the balanced signal path using high-quality signal relays.
In addition to the DVM-120-SPS Speaker Selector, the DVM-120-SPR Monitor Controller has a level control via an incremental rotary encoder; the adjustment is done using a display which indicates the level dB-perfect. The functions can be controlled using a remote control (connection via Cat-5/Cat-6 standard patch cable).
All devices offer a rugged housing design made of an extremely solid 3mm extruded profile. The front and connection plates are recessed for added protection.
All active devices are equipped with an internal power supply, suitable for operating voltages between 90-volt and 240-volt (50/60 Hz). The mains supply is provided via a high-quality combined IEC switch/fuse unit.
The high input impedance of the Lundahl transformers—about 560 kohms and switchable to 280 kohms allows direct,noise-free connection to the mixing console or preamplifier.
Recording the music in your worship services on a two-track recorder, such as a CD recorder, seems like it would be simple. But if your mixes sound terrible, here’s how to do it the right way.
Most mixing consoles have a set of connectors labeled “tape out” or something similar. And while you may be tempted to simply plug your CD recorder into that output, there’s probably trouble on the way.
Here’s why. Let’s suppose you have an electric guitar or drum kit on stage. It’s probably so loud all by itself that you don’t have to add much, if any at all, of these instruments into the main PA loudspeakers. But you’ll probably also have a vocalist who will need to be fully amplified in the main speakers to be heard at all.
This may mix together nicely in the room, but if you record this PA loudspeaker mix directly to your CD recorder, it will sound all wrong. There will be overpowering vocals and almost no electric guitar or drums at all, save that which is picked up by any open mics on stage.
That’s because a “console recording” doesn’t include anything that was loud on stage to begin with. Essentially this recording is the inverse what actually happened on stage and bled into room. What’s a sound tech to do?
Just use a spare auxiliary (aux) send to develop a separate recording mix. Using a post-fader send for this will automatically track any fader moves. Note that the aux send for each channel will be the opposite of the fader position. That is, if you have a fader pushed most of the way up to amplify a vocalist, you’ll probably need to turn that aux send down quite low.
On the other hand, an electric guitar (or other loud instrument) will have its fader most of the way down, so you’ll need to turn up that particular aux send. Remember not to pull the fader down all the way, or its sound won’t go to the aux bus at all.
And you can always un-assign that instrument from the main loudspeaker by using the bus selector switches if need be. Note that adding even a small amount of e-guitar in the main PA speakers helps even out a room.
This, by the way, is also the mix that you want to send to your remote speakers in another room.
If your console has a stereo aux send bus, you can also do this recording in stereo. Many Mackie boards have stereo aux sends, as do several consoles in the Allen & Heath product line. But most of the time, a mono mix will be sufficient.
Monitor this mix with a good set of sealed headphones from the CD recorder’s headphone-out jack to get the proper balance between the vocals and drums or guitars. Then you’re on your way to making great recorded mixes everyone will appreciate, especially the drummers and guitar players.
Happy mixing!
Mike Sokol is the chief instructor of the HOW-TO Church Sound Workshops. He has 40 years of experience as a sound engineer, musician and author. Mike works with HOW-TO Sound Workshop Managing Partner Hector La Torre on the national, 36-city, annual HOW-TO Church Sound Workshop tour. Find out more here.
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