Wednesday, September 25, 2013
Church Sound: Are You Blinding The Congregation With Your Mix?
Escaping personal mixing bias
My eyes felt like they were as big as dinner plates and any sunlight was blinding. Welcome to having an eye exam. During the exam, your optometrist will insert eyes drops that widen each pupil to the size of a grapefruit. This enables him to check the back of your eyes for signs of diseases and medical conditions.
The problem with the eye drops is they keep your pupils in that WIDE OPEN state for about, let’s see, it’s 3:42 now, so….about six hours. During this time, sunlight is your enemy. It’s your kryptonite. It’s blinding.
Mixing audio, you have a natural bias in your mix style. You might mix bass-heavy. You might mix with a strong tendency towards boosting high frequencies. You might push an instrument a bit too loud a result of your personal preferences. To you, it seems natural but to others, it’s BLINDING!
This isn’t to say all mix biases are bad. I’ve heard song covers that were better than the original. I’ve heard re-mixed songs that brought new energy or a different emotion to a song.
However, it takes a lot of talent to be able to do that and you need an audience accepting of “something different.“Today let’s consider if your mix bias benefits the music or blinds your congregation.
Two Signs Of A Blinding Mix
1. You get regular complaints from different people about the same problem.It’s one thing to get an occasional complaint about volume (overall or instrument-specific) but if you are getting a weekly dose of complaints from different people about the same thing, then you are blinding them with your mix. You must reconsider your mix and remember your purpose in mixing.
2. The congregation doesn’t engage in worship as when someone else mixes. You might think the other tech’s mix is sub-par to yours, but if the congregation is more engaged in worship whenever they’re mixing, then it’s your mix that’s sub-par.
Please know that comparing mixes isn’t a competition. It’s not about who is better, it’s about who is doing what’s best for the congregation, the room, the music, etc.
Are there other signs of a blinding mix? Probably. But these are the two which should give you immediate cause for consideration.
What Can You Do?
1. Take it personal. I mean this in a bad way.
You could think things like, “the congregation doesn’t appreciate the quality of my work” or “I don’t care what the congregation thinks.” Ummm, too bad for you, if you do. You have missed the point of church audio production. If you are at this place, please read this article.
2. Take it personal. I mean this in a good way.
You might mix on weekends for a country band and create a phenomenal mix. But that same mix might not be best for your church congregation. It’s different people, it’s different music, and it’s a different purpose. Accept that you have a mix bias and learn to adjust your mix style for the congregation.
For a month:
—Listen to how the other church techs mix.
—Listen to worship music via iTunes or Spotify (or whatever works for you).
—Listen to how volumes are different.
—Listen to how individual instruments are mixed.
—Listen to how the overall mix is shaped.
—Listen for similarities in how you mix and listen for the differences.
The Take Away
Working behind the mixer, you can be as creative as you wish to be. The problem is you can be blind to your personal mix bias. That, in turn, can be blinding to your congregation. Be aware, mix smart, and let the Holy Spirit be the only thing that shines on your congregation.
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, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.
Church Sound: A Step-By-Step Process For Optimizing Stage Monitors
“Oh it sounds much better. Very musical and pretty much exactly what I expect to hear.”
A few weeks ago I was having a discussion about the sound quality of our monitors with our sax player.
He’s a very discriminating guy who really knows sound. If he says he needs 185 cut 3 dB, he needs 185 cut 3 dB.
As we listened to his wedge, it was clear that the sound he was putting out of his horn was not being faithfully reproduced by the wedge.
I knew it would take a while to remedy this, but I set aside a day to go through and fix the wedges.
First, here is our signal chain: Each mic gets plugged into a splitter (it’s passive, not transformer isolated, which was a bad choice, but not mine).
One split goes to an M7 in monitor world. The M7 mixes up to seven wedge mixes.
The omni outs of the M7 go to two Klark Teknik 9848 4x8 processors set up in 4x-biamp mode. The KT 9848s feed a rack of QSC amplifiers, which in turn feed EAW SM12 monitors.
Aside from the split, it’s a decent system. It took me a little bit, but I finally got my Mac talking to the 9848s (using Parallels, XP and a RS232-USB converter). I should mention that KT’s tech support was very helpful and quick in getting this running.
Once I had that going, I had a nice, graphical interface with which to adjust the settings. I positioned a wedge in the middle of the stage, and placed our Earthworks M-30 measurement mic right about where a musician would stand.
I took the following approach: When EQ’ing monitors, you really aren’t worried much about the room as it’s really a near-field monitor.
The only real boundary is the stage itself, and my goal was a pretty linear system; that is, flat and set up so that what goes into the board comes out of the monitors.
In the past, I would have started running pink noise through the system and looking at the response on an RTA. But that amount of noise (I measured at 94 dB SPL-A) gets annoying really fast.
I’ve been learning more about more modern forms of measurement including swept tone and FFT, so that’s how I went about this process. I’ve found swept tone gets me a lot closer a lot faster than pink noise, without the grating noise.
I started off with a great little program I found called FuzzMeasure Pro 3 (from SuperMegaUltraGroovy Software, the best software company name ever). FuzzMeasure uses swept sine wave deconvolution to report frequency response.
If that sounds complicated, don’t worry. All you need to do is hook the mic up to a USB interface and press measure.
The software emits a quick click that is used for impulse measurements (great for setting delay, but that’s another post), then a swept sine wave from 20 Hz to 20 kHz (or in my case, 50-17 kHz; it’s user-adjustable).
The resultant frequency response is shown on a graph. Here’s where we started:
(click to enlarge)
Keep in mind that each light grid line is 1 dB. So we started off with the low end some 13 dB below the mid- and upper-range; not so good. I made a 12 dB adjustment on the gain for the low channel (actually I cut 6 off the top and added 6 to the low).
(click to enlarge)
Now we start getting a little closer. But it’s still way off. After spending some time with the parametric EQs built into the KTs, I ended up with a sweep that looked like this:
(click to enlarge)
It might look a bit wonky, but realize that this time, the heavy grid lines are 1 dB. So the response could be called flat, ±1 dB from 80 Hz-17 kHz (the right edge of the graph is 17 kHz, as set in my prefs)—that’s not too shabby.
Now, because I was getting more complaints from some musicians than others, I decided to drag another wedge over and take a measurement.
I was surprised (well, not that surprised) to see a significantly different trace, even with the same EQ settings as the first one.
So I decided to tune each monitor individually. In my new setup, each monitor has a number, and it will always be used with the matching monitor send.
Thus, Speaker 1 will be plugged into Monitor 1 on the patch panel. That ensures that all monitor mixes are basically the same, even though production variations give each wedge a slightly different response curve. I’ve applied custom EQ to each one.
The final step was to tweak it a little closer using another cool program called Spectre from Audiofile Engineering.
Spectre has a a compare trace FFT function which allows you to look at the signal coming out of the board and the signal coming back from the measurement mic at the same time.
With some gentle pushing and pulling of the EQ curves, I was able to get that line almost completely flat. It’s a little easier to see in this view, where the purple is the output of the M7 pink noise generator and the green is what’s coming back from the measurement mic (which is nearly completely flat from 20-20 kHz).
(click to enlarge)
What’s fun about this view is that if you add say, a 8 dB bump at 1 kHz on the output EQ of the monitor mix, both curves show exactly 8 dB of bump, in a bell curve that looks just like the graphic on the EQ display.
I had now reached the point of it being a linear system; what goes in comes out. The next weekend, I asked our sax player how his wedge sounded.
Without telling him what I had done, he commented, “Oh it sounds much better. Very musical and pretty much exactly what I expect to hear.”
Mike Sessler is the Technical Director at Coast Hills Community Church in Aliso Viejo, CA. He has been involved in live production for over 20 years and is the author of the blog, Church Tech Arts . He also hosts a weekly podcast called Church Tech Weekly on the TechArtsNetwork.
Friday, September 20, 2013
No Shock Zone: Understanding And Preventing Electrical Damage (And Worse)
The fundamentals of Ground Fault Circuit Interrupters (GFCI)
Every year, hundreds of musicians and audio techs suffer serious electrical shocks while on the job, and these incidents have the potential to be fatal. Therefore, it’s imperative to have a basic understanding of electricity to avoid possible electrocution, and further, this knowledge can help protect your equipment from going up in smoke.
No it’s not the name of an insurance company or a European sports car, GCFI is an abbreviation for Ground Fault Circuit Interrupter or G-F-C-I. They’ve been required in many localities for electrical outlets located near sinks and outdoors for the last 10 years or more. The two types of GFCIs you’ll encounter are either built into the power outlet itself (Figure 1, left) or inside the circuit breaker at the power panel (Figure 1, right).
Both do exactly the same thing: they watch for electricity that’s going someplace it shouldn’t in an electrical Circuit by way of a Fault to Ground and then Interrupt the flow by tripping the circuit breaker. Rearrange the letters and you get G-F-C-I for Ground Fault Circuit Interrupter. That’s how the name is derived.
Why Do We Need A GFCI?
The heart muscle of humans is very sensitive to electrical shock. While it takes around 8/10ths of an amp (800 milliamperes) of current to power a 100-watt light bulb, it takes about one percent of that same current (10 milliamperes or so) to send your heart into fibrillation, causing death by electrocution.
Figure 1. (click to enlarge)
That’s why the NEC (National Electrical Code) now requires a special type of circuit breaker for damp locations that can tell the difference between the normal currents feeding an electrical appliance and the currents accidentally flowing through you to ground. And while a GFCI sometimes trips unexpectedly, it’s really there to save your life and the life of your appliances and other electrical components.
How Does A GFCI Work?
It’s a pretty ingenious system that uses a small current transformer to detect an imbalanced current flow, so let’s use our water pump analogy to review the typical current path in a standard electrical circuit.
Figure 2. (click to enlarge)
As you can see from Figure 2, we have our pump and turbine system again. And let’s imagine the pump at the top is pushing 7 Gallons Per Minute (GPM) of water current around in a circle that our little turbine at the bottom is happily using to spin and do some work.
I’ve added flow meters at the bottom left and right of the illustration so we can keep track of these currents. Now since our pipes have no leaks, the current going out of the pump from the black pipe will exactly equal the return current coming back in the white pipe. And this will be an exact balance since no water is lost in this closed loop. That is, if 7.000 GPM (Gallons Per Minute) of water are flowing out of the black pipe, then 7.000 GPM will be returning to the pump via the white pipe. There are no water losses in this perfect system.
Keeping In Balance
Let’s add an extra meter in this system so we can keep track of the water flow a little easier (Figure 3). Notice there’s now a center meter that will show you the difference in flow between the other two meters. If the left and right meters show exactly the same water flow, the center meter will show zero GPM of flow by centering its needle.
Figure 3. (click to enlarge)
This is exactly what should happen in an electrical circuit that’s working properly. That is, if a light bulb has exactly 1 amp of current flowing out from the black (hot) wire, then exactly 1 amp of current should be flowing back in the white (neutral) wire. And an electric griddle that has 10 amps of current flowing out the black wire should have exactly 10 amps of current flowing back in the white wire.
If there’s nothing wrong in the light bulb or griddle circuit, this electrical current balance will be pretty close to perfect, out to at least 3 decimal places. That is, 10.000 amps of current flow going out will equal 10.000 amps of current flow coming back in.
Out Of Balance
Now I’ve added a leak in the black outgoing pipe via the red pipe sticking out to the left (Figure 4). You can see from the red pipe’s meter that 5 GPM of water is flowing out onto the ground. And since only 7 GPM of water is coming out of the black pipe on the pump, there can be only 2 GPM of water returning into the white pipe on the right.
Figure 4. (click to enlarge)
Those 5 GPM of imbalance show up in our center balance meter, which alerts us to the fact that there’s a leak somewhere in the system. Now, we really would like to know about small leaks as well, so that center meter will tell us about an imbalance down to very small drips, say less than 1/1000 of a GPM.
The same is true of our electrical circuit where we’re interested in currents in the 1/1000 of an ampere range (1 mA or 1 milliampere). That’s because just a few milliamperes of misdirected current flow is close to the danger level for stopping your heart.
In an electrical system, a similar type of detector is used at the center of the circuit which is acting like a balance beam. So if 7 amps of current shows up on both sides of the balance, then the beam will be exactly level.
Figure 5. (click to enlarge)
However, put 7 amps of current on the left side and 2 amps of current on the right side, and that 5 amps of imbalance will tip the scales, just like the teeter totter ride you took with your dad when you were maybe 50 years younger and a 150 pounds lighter (Figure 5). In our GFCI circuit this is a much more sensitive balance beam that only needs 5 mA (5 milliamperes or 0.005 amps) of current imbalance to tip over rather than the 5 GPM we’ve shown in the water pump illustration.
The reason for needing this much sensitivity is that our hearts can go into fibrillation from just 10 mA of AC current flow, so we would like to detect and stop that flow before it stops your heart.
Putting It All Together
So here’s where it all comes together. Notice that our guy is unwisely touching a hot wire with a hand while his foot is in contact with the earth (Figure 6). And while the electrical outlet might have been supplying 7.000 amps of outgoing current to an appliance with exactly 7.000 amps of return current, there are now 7.005 amps going out and only 7.000 amps coming back.
Figure 6. (click to enlarge)
Those extra 0.005 amps of current (5 milliamperes) are taking a side trip from his hand to his foot via the heart. And the current balance circuit inside the GFCI is sensitive enough to recognize that imbalance and trip the circuit open with as little as 5 milliamperes of current flowing someplace it shouldn’t be going.
The click you hear when a GFCI trips is its spring loaded contact opening up and interrupting the current flow in the circuit before it causes electrocution. That’s the entire GFCI’s reason for existence, to save you from electrocution and keep the electrical system safe from damage. Pretty cool, eh?
Also note that the GFCI doesn’t really need a direct ground connection via the ground wire to do its job. Yes, one is required to properly “earth” the entire circuit, but the current balancing act is only between the black and white wires going to the outlet. If the current flow in the white wire exactly matches the current flow in the black wire to within 5 mA (milliamperes), the circuit stays activated. If the current flow is unmatched by any more than 5 mA, say by someone touching a live wire and the earth at the same time, then the trigger circuit inside trips a little switch and the current flow is stopped. It’s that simple.
All this means you should install GFCI breakers where required, and don’t remove or bypass them if there’s false “nuisance” tripping. That so-called false tripping hints there’s something else wrong in your electrical system that’s leaking out current to someplace it doesn’t belong. And fixing that electrical leak is important since if you get your body in the middle of the current leak it can shock or even electrocute you. Just a quick definition of the word “electrocution.” That’s death from being shocked. So none of you readers were “electrocuted” last night at a gig and reading this now. Unless you’re zombies, and that’s an entirely different thing…
Mike Sokol is the chief instructor for the HOW-TO Sound Workshops (www.howtosound.com) and the HOW-TO Church Sound Workshops. He is also an electrical and professional sound expert with 40 years in the industry. Visit www.NoShockZone.org for more electrical safety tips.
Posted by Keith Clark on 09/20 at 02:30 PM
Wednesday, September 18, 2013
Properly Cleaning Mixing Console Faders
Cleaning a fader is not brain surgery, but it takes practice and a lot of care. Here's how to go about it - successfully.
Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.
I’ve cleaned a lot of faders over the years and suppose I’ve gotten a little bold when it comes to tearing a fader apart and giving it a good bath.
And I’ve learned the hard way just how much punishment a fader can take before it breaks.
In some cases, a certain amount of brute force is required to crack open a fader, but then a certain amount of gentle finesse is needed to clean its individual parts.
I recommend practicing with old junk faders – without experience, it’s all too easy to ruin a good one.
Cleaning a fader is not brain surgery, but it takes practice and a lot of care.
Before we getting into a total fader rebuild, let’s talk about quick cleaning. Much of the time, if the gear has been well cared for and the faders are not too dirty, then a little routine maintenance is all that is likely required.
Besides, keeping faders clean is always a good idea, preventing dirt from becoming embedded deeper inside where it can cause more wear and tear.
Keep in mind: with relatively new faders in particular, do as little as possible in order not to undo the original lubrication. And overall, don’t go any further with this process than you feel you need to.
The first step is to use compressed air to blow as much dirt as possible out of the fader.
Figure 1: Start by blowing one end, and then the other. (All photos by Alex Welti)
There is usually “dust bunnies” in the fader that will come out easily, and this might be all that needs to be achieved in terms of cleaning.
Move the fader carriage to one end and blow air into the slot aiming away from the carriage so that dust can escape through the slot. Then move the carriage to the opposite end and blow air aiming the opposite way.
Skip this step and compound the laziness by spraying some off-the-shelf cleaner-lube into the fader, and it’s likely that the dust bunnies will be matted down and stick in the corners.
Laziness can lead to temporary improvement but later, the dreaded “dust bunnies” in the corner syndrome.
A fader might seem to work better for a while, but this won’t last and might lead to the need for a more substantial (and time consuming) cleaning effort.
Note that the compressed air must be clean and dry.
I do a lot of cleaning, so I’ve invested in a $100 air compressor and then added an air filter / dryer unit for about $40. To this I’ve added a dryer cartridge that contains silica beads for about $5.
If an air compressor isn’t available, cans of aero-duster will work, but they don’t last long.
If the plan is to clean a couple/few consoles, an air compressor is a worthwhile investment, and it helps do the job right because you don’t need to be worried about running out of air.
In addition, the compressor will offer higher pressure.
Most canned air provides about 60 psi, with this dropping as the can is used.
With the compressor, I’m able to set pressure at a consistent 80 psi, which works very well. (And I found out the hard way that 100 psi will blow some faders and switches apart!)
If the initial “blowing out” process didn’t offer the desired results, it’s time to move on to use of chemical contact cleaner.
Figure 2: Contact cleaner outfitted with a nozzle that adds precision and cuts waste.
Some faders have lubricating grease applied by the manufacturer, while others employ a self-lubricating Teflon-type of plastic.
If used sparingly, chemical contact cleaner shouldn’t impact the self-lubricating type, but it will invariably wash away lubricating grease.
The goal is to avoid adding any more lubrication than is absolutely necessary - dust tends to fall away from dry surfaces, but it sticks to oily surfaces.
Figure 3: “Snap together, snap apart.”
After spraying contact cleaner, exercise the fader and then quickly blow out the excess cleaner.
This helps to spread the cleaner over the entire fader surface, while the excess cleaner carries away additional loosened dirt.
I’ve tried several types of contact cleaner since canned Freon was banned from the market.
There are a lot of good choices – my preference is Contact Cleaner II made by Techspray. It’s about $30 per can and worth the price. Note that I also invested another $30 for a screw-on trigger nozzle so that I can be precise and cut waste.
The fader is still feeling a little rough? Time to try a little lubrication. The key word is “little” – use as little as possible.
Did I mention not to use too much lubrication? Third time’s the charm – lubrication collects dust, so don’t overdo it!
Depending on the type of fader, I use a precision dropper to place just a few drops of lubrication in the fader, or give it just a quick squirt.
Exercise the fader and then blow away the excess with compressed air. Again with the compressed air?
Figure 4: The basic parts of a typical fader, and where they’re located.
Seriously, this helps spread the lubrication into a thin film and gets rid of any excess.
I’ve had good results with a spray lubrication called Tefrawn, made by Rawn. It’s Teflon-based and beneficial to the self-lubricating type plastics noted earlier.
Also, it smells like bananas, not that it matters!) Caig also offers products of this type.
Lesson learned the hard way: some oils react with plastic, causing it to break down. If there’s any doubt, test it out on a spare fader first before applying.
Figure 5: Be careful not to damage the wiper, which can ruin the fader.
Also, certain faders use thicker grease that results in a “smoother” feel, and these may actually feel too loose after lubrication.
If this proves bothersome, use silicon or petroleum grease (but not bacon grease!). I’ve found this step to be more trouble than it’s worth - if “feel” is that important, buy new faders.
Time to reiterate: “Air > Cleaner > Air > Lubrication > Air” About 30 seconds of effort for each fader.
Figure 6: Under and around the rails, but don’t touch the carriage.
Some of the more expensive faders are designed to be easily taken apart for cleaning.
If less expensive faders can’t be cleaned using the steps already outlined, it may not be cost-effective to go any further.
Consider replacement, but if it’s an emergency, keep in mind that you’ll be dealing with tiny parts that are easy to break and lose.
A total fader rebuild should take only about 5 to 10 minutes, after going to the trouble of taking apart the console to get to the fader.
If I take a module out for repair, I go ahead and clean its fader at the same time. Otherwise, I do fader rebuilding as part of a larger console-cleaning project.
There are several different types of fader construction. Higher-cost faders are literally a “snap” to take apart; that is, they have a “snap together” design.
A much more pleasant use of a dental pick than usual.
The main parts of a typical fader include the element that carries audio on conductive tracks, the carriage that holds wipers against the tracks, and the rails that guide the carriage.
Be extremely careful with the wipers - they’re easy to damage, and once bent, the fader is toast.
After opening the fader, first blow away the loose dust. There might be dirt wedged in at the point where the carriage and rails meet, so use a dental pick to loosen this up, and then blow it out.
Again, blow the loose dirt out.
Use a strip of clean cloth dipped in isopropyl alcohol to clean the rails, pass the strip under and around each rail.
Gently wipe the surface of the conductive element with a clean cloth dipped in alcohol or contact cleaner. Be gentle, and do NOT go under the carriage with the cloth. This can damage the wipers!
Top it off with just a dab of lubricant. Caution: a little goes a long way!
Apply just a few drops of lubrication to the rails and exercise the fader. Blow away any excess lubrication with and reassemble the fader
And that’s it. With a little practice and patience, anyone can make old faders feel like new again!
Editors Note: For more information on console maintenance, check out Zen & The Art Of Mixing Console Cleaning & Maintenance.
Alex Welti is vice president of research for Creation Audio Labs, a service facility in the southeastern U.S. He served for a decade as service manager of Soundcraft, and prior to that, worked as a technical supervisor for Westlake Audio.
Monday, September 16, 2013
Make It Stop! There’s No Excuse For Loud, Bad Sound
Are we part of the solution or part of the problem?
Preserving our hearing is critically important, but that’s not what motivated me to write this piece. Really, I just can’t stand bad sound!
To a certain extent, it used to be somewhat excusable if a live show was loud and didn’t sound all that great.
Sound reinforcement systems have come a long way in the past decade alone, but ‘back in the day’” good clean power was harder to come by, and loudspeakers just were not designed to produce high-fidelity audio.
In fact, it was generally considered that you either had reliable speakers OR ht-fl speakers, but not both.
The late Albert Lecesse’s oft-quoted list of priorities goes like this: “Make sound. Keep making sound. Make good sound.”
And for legions of engineers, this has been the mantra, i.e., that high quality sound is not the first priority. And I agree with this for the most part.
However, with in-ear monitoring, processor-controlled loudspeakers, line arrays, plenty of good, inexpensive power, good wireless systems, higher quality microphones than ever before, etc., etc., the equipment has ceased to be the problem. In fact, it’s become part of the solution.
But what about us, the operators? Are we part of the solution or part of the problem?
I’ve encountered the resistance to using better microphones in a lot of places: “This warhorse mic has been good enough for the last 30 years so why change now?” Indeed.
But when was “good enough” ever really good enough? Those standard mtcs became a standard for a reason: they were the best thing available at the time. Are you using the best tools and techniques available to you right now?
Instead of just pointing the finger and expecting to get your middle finger in response, I thought it might be useful to cover some of the things, I think, that are fairly easy to implement and that can make a huge difference in the quality of our product, i.e., the sound of the events we mix.
Previously I’ve talked about some general topics related to listening, thinking and related audio issues. This time, let’s get more specific.
HEARD IT BEFORE
Gain Structure. This has probably come up more times than any other concept in sound reinforcement, and for good reason. It’s the basis of how the signal goes from one device to another and within devices, and understanding it fully goes a long way towards at least giving us a fighting chance at producing good sound.
Here are the main points: Avoid excess noise and maintain enough headroom to keep signal from distorting.
First, it’s not a good idea to be combining consumer, semi-pro (whatever that means) and pro equipment in a single system. But sometimes it’s unavoidable, particularly when you need to pipe in some CD material or feed a mix to a video camera.
However, understanding that not all ‘line level’ signals are alike is a good place to start towards integrating these devices, For example, there is roughly a 14 dB difference between the output level of consumer gear and that for pro gear!
But let’s say that you have gain structure knowledge under control and levels are carefully managed throughout the system, yet the system has a fairly high amount of harmonic distortion. Wait, let’s back up: how would you know? I’ve run into a lot of sound engineers that aren’t readily able to identify modest levels of distortion, or if they can, they still can’t identify which type of distortion is predominant.
In music school, students are required to take “ear straining and sight screaming” otherwise known as ear training and sight reading. Let’s concentrate on the first part.
There are CD sets for sale, such as the Golden Ears library, designed to serve as ear training materials for audio engineers. And these can be very effective.
I also suggest finding ways to set up your own experiments. For instance, set up two stereo channels on the console, fed from a CD player - one with really good gain structure and the other with some problems (for instance, bring the faders down to -20 and bring the input gain up so that the output to the mix bus matches that from the “clean” channel).
How far do you have to go to push the problem channel into distortion before you can hear it? Then adjust the EQ on the two channels - does it make the distortion on the “bad” channel worse at better? And what EQ settings make the problem worse?
Related to this, and hopefully part of what you either already know or what you might discover with the above experiment, is that we do not hear all frequencies equally. Fletcher and Munson at Bell Labs were able to determine a set of ‘equal loudness curves’ pointing to the fact that we hear midrange frequencies (3 - 5 kHz) the most easily, and our response drops off from there.
As such, very high frequencies and very low frequencies are more difficult to hear. Thus, for distortion or other audio problems, this set of response curves is very important. If there is distortion in the midrange, then we will be more likely to hear it than if it is in the bass or the high treble.
Equal loudness contours. (click to enlarge)
For this reason, loudspeaker manufacturers (in particular) spend an enormous amount of effort to reduce distortion in this ‘region. This range of frequencies also .happens to be an important component of vocal sounds, primarily to distinguish between consonants… Ever wonder why it’s so hard to tell the difference between B, T, D and E over the telephone?
Several years ago I heard a demonstration by ServoDrive, a subwoofer company with a product that used servo motors instead of voice coils to drive the cones.One point made has stuck with me: For a mix to sound solid, the midrange doesn’t need to be loud, only the bass.
The company proceeded to show this by playing a mix that was very heavy on the bottom, but light in the mids. Those of us attending the demo could talk to each other, yet the system seemed to be putting out a HUGE sound. I was impressed!
Since then I’ve always wondered why many mixers seem to push the mids into distortion…
IN THE DRIVER’S SEAT
Another thing that affects system headroom, and thus distortion, is signals in the mix that don’t need to be there. This is one reason why a lot of “old school” mixers know to use the EQ to cut only, never boost. I was taught that the first thing to do when considering how to start EQ’ing a mix is to begin cutting the low mids.
First, boost the EQ enough so that you can hear the effect, and then sweep the filter to find the most obnoxious frequency. Then cut that frequency 3 dB or so. By doing this on the majority of channels, I’m always surprised how much better the mix sounds.
Frequency response of a telephone. (click to enlarge)
On vocals, anything below 80 Hz isn’t needed (unless you’re mixing Barry White, and maybe not even then). Same for acoustic guitar. The only instruments that really need much below 100 Hz are kick drum, bass, maybe floor tom, keyboards, and perhaps electric guitar.
Not only is it important to keep most of the instruments out of the subwoofers, but it’s important to try and keep lots of low frequencies out of your mains. Tom Young does a great job here of describing how to incorporate subs via an aux channel. I used to do this when I was touring: Put the upright bass and kick drum mic into the subs, but nothing else. And it worked. Really well.
WRAPPING IT UP
My main point of all this is that we don’t have to make loud, bad sound, so please don’t. There are creative, clever ways to use almost any equipment better than we’re using it now. And there are better tools available for much lower prices than there used to be.
So then, making good sound or not becomes a choice. Learn more about the equipment you have, how to use it, and what new gear might solve some of the problems.
Get your ears tuned up by listening to acoustic music, quality recordings and quality show mixes. Do some ear training to learn to identify different frequencies, different kinds of distortion and other aspects.
It’s easier than ever now to make sound and to keep making sound. Now it’s time to make good sound!
Karl Winkler is director of business development at Lectrosonics and has worked in professional audio for more than 15 years.
Wednesday, September 11, 2013
Is There A Better Way? Thoughts On Sound System Security Measures
Rather than control, is fostering a different atmosphere a better approach?
I’m not big on physical security for audio systems. Locked sound rooms and doors. Locking rack cabinets. Security Torx screwheads. Warning signage and sticky tape threats (Do not touch this knob!).
And we’ve all seen the locking rolltop desk that covers the entire sound system.
Why do folks do this? In a word – Fear. Fear of damage. Fear of loss of control. Fear of the unknown (if someone twiddles with the knobs – they’ll totally screw up the sound for this Sunday!!!)
Yes, a sound system is a significant investment by the organization, and it’s true that its misuse by untrained personnel could result in lots of financial exposure.
But it’s also true that the sound system is not a nuclear reactor. It is not rocket science or brain surgery. Running a sound system is a lot more like driving a car. It is a skill that can be learned with some training, observation, and experience.
It drives me crazy when I go into a sound room and see pieces of masking tape attached to the mixing board, the CD player, the video camera, or other gear with threats or warnings in black magic marker. Here’s some recent examples of ones I have seen: “NEVER TOUCH,” “DO NOT MOVE,” “ASK BOB BEFORE TOUCHING THIS,” “MAKE SURE FADER IS AT 1/3.”
I have to be honest here – if your sound room contains such messages, it is an indication of bad leadership. The sound techs running the console shouldn’t need those kinds of signs because 1) they’ve been trained, and 2) they know what those knobs do and what their optimal settings should be.
Here’s a wild thought. What if instead of trying to control everything and everyone, we instead fostered an atmosphere of freedom, learning, mentoring, encouraging, and trust? What if instead of investing in security hardware, we invested in the training and development of those who have an interest? What if our book of rules, administered by the cranky head deacon or defensive facilities guy were replaced by a much smaller set of rules, such as:
1. A single sign - “We only allow trained and approved sound techs to run our system. To sign up for training, please see John Doe.”
2. A requirement - All personnel who might have reason to enter the sound booth (music director, programming guys, theater operators, drama director, techs in other disciplines, maintenance staff, musicians, singers, etc.) must first go through audio training. This models the right thing to others, gives everyone a consistent baseline of learning, and prevents “accidents” from people doing ignorant things.
What’s the best way to learn how to run a sound system? The same way you learn to do anything else! Here is a great little system described by John Maxwell in the book, Developing The Leaders Around You:
1. I Do. You Watch. We Talk.
2. I Do. You Help. We Talk.
3. You Do. I Help. We Talk.
4. You Do. I Watch. We Talk.
Then you repeat these steps with someone else. This is how you build a team that is confident and capable.
Are there any circumstances where security hardware is prudent? Sure. The system processor is a part of the system that should not be adjusted except by a trusted professional. A security password or security cover is completely appropriate for this piece of gear.
Jeremy Carter is a veteran of the pro audio industry with extensive experience designing and operating church audio, video, and lighting systems.
Coming Up: Compact Loudspeaker System Demo In Conjunction With WFX In Dallas
The Live Sound International Compact Loudspeaker Demo at the upcoming WFX Conference and Expo in Dallas provides you with the opportunity to directly listen to, evaluate, and compare more than a dozen compact loudspeaker systems.
The first two iterations of the demo have drawn a total of more than 1,500 church sound and production personnel from around North America.
The demo presents a unique controlled environment demonstration designed to provide side-by-side listening opportunities for you to evaluate compact loudspeaker systems from around the industry, in addition to getting further technical details and pricing information from qualified representatives of each company participating in the demo.
Industry leading loudspeaker manufacturers participating this year include Renkus-Heinz, L-Acoustics, Adamson Systems, Martin Audio, Danley Sound Labs, D.A.S. Audio, Bose Professional, EAW, Alcons Audio, Electro-Voice, PreSonus, Line 6, and Elipsis Audio.
The demo will be held October 2 and 3. The site is an expansive exhibit hall in the Dallas Convention Center, adjacent to the WFX show, where each day, three 1-hour demo sessions will be held.
During each session, all systems will be played, using identical tracks. Listeners will move from system to system, evaluating what they hear and also observing each system’s scale, components and other important details.
Each participating compact loudspeaker company will also be offering a dedicated 15-minute exclusive demo session for interested attendees on both days of the event.
Demo sessions are open to all attendees of WFX as well as other groups and individuals interested in audio and loudspeaker systems.
In addition, an “A-list” of technology sponsors are also involved with the demo, including Yamaha (consoles), Shure (wireless microphone systems), Link (digital signal transport), Mega (lighting), and Dicolor (LED displays).
Don’t miss this opportunity to gain valuable insights on what’s available in the world of compact loudspeaker systems in this comprehensive event!
Go here to register (FREE) for WFX.
In The Studio: Editing To Tighten Up Performances
Not just because you can, but because it sounds better
I’m currently working with a band and after each part is recorded I’m spending a good amount of time editing to tighten up the performance.
It’s not because they can’t play well, they’re actually really good, I could probably get away with not doing any tightening of the performance at all. I’m tightening the performance not because I can, but because it sounds better.
To me, the effort and time spent is worth it. It makes the song sound more polished and another step closer to a professional result. I do this kind of editing on everything I work on; I won’t mix a song that has a sloppy performance.
This editing to tighten things up is often called “pocketing.” It takes time and it’s not very much fun, but it really makes an difference. In this project I’ve edited the drums, bass and guitars to be perfectly in time. Some people like the bass a bit behind the beat by a certain amount and have different ideas about where things should be.
I do everything right on the grid and it sounds right to me. I have not yet had a complaint that it sounds too perfect or too rigid.
You don’t have to do every 16th note of the performance, you can do much less than that. I usually start by lining up all elements at the start of each section of the song.
Then go through in finer detail if anything sounds off. Often I’m adjusting down to 8th notes especially if parts are double tracked, with two performances panned left and right of the same part.
If those parts aren’t tight you get a distracting bounce between each ear. I really hate that.
DIs Really Help
When I record guitars I always like to have a DI track. I don’t usually ever need to actually listen to the DI track, but it’s useful for editing because the transients are much clearer than that of a mic on a distorted guitar amp.
Of course, the DI is useful as a safety if you end up hating the guitar tones from the mics.
Cutting & Warping
There are two methods to pocketing, cutting or warping:
Cutting is separating the recording at each section or note if needed and moving the start of each piece into time. Then filling gaps and crossfading between each piece. In Pro Tools you can do this automatically with Beat detective or manually.
Warping would be using a function like Pro Tools Elastic Audio or Logic Flex editing to stretch and shrink the notes within the recording without making any separations. Warping is faster but there is a sound quality loss.
I do this kind of editing a lot. In general cutting works best for drums. Warping is usually fine with bass and guitars and vocals but not always, sometimes it can really degrade the sound.
Jon Tidey is a producer/engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com.
Real World Gear: Recent Microphones For Live Vocal & Instrument Applications
New mics in a range of formats
Expanding on the basic technologies of dynamic and condenser transducers, manufacturers continue to develop new microphones to meet the needs of musicians and sound engineers in a range of formats, sizes, and price points.
In more recent introductions, we’re seeing a mix of innovative development, refinement of previous technologies, specialization of mics to specific applications, and changes in manufacturing.
Higher end condenser vocal mics are truly amazing devices, and even most of the relatively inexpensive models are durable, reliable, and sound good. However, one microphone designer recently lamented to me that some companies are becoming more focused on lowering manufacturing cost to meet competitive pricing points, with the result being a compromise of quality.
Looking at the latest offerings, companies are going with their strengths, and expanding on their unique technologies. For example, Countryman’s latest offering takes advantage of expertise with miniature, high-fidelity condenser mics, resulting in the H6 hypercardioid headset. Sennheiser continues to add dynamic and condenser models to the Evolution (“e”) Series, and beyerdynamic extends the TG Series with a ribbon model.
In addition, both DPA and Earthworks are building upon measurement mic backgrounds to introduce roadworthy condensers for vocals. These projects were motivated when touring engineers came to appreciate the audio qualities of these precision devices, experimented with them in the field, and then approached both companies with requests to develop them into a format for vocalists.
DPA made another step with the modular design of the d:facto II vocal microphone, where the transducer can be easily moved from the handheld format to a wireless adapter. Thus, the same mic can be used in either a wired or wireless application, as needed.
Several ribbon models have been introduced relatively recently for both instrument and vocal reinforcement. These are rugged enough to survive the rigors of touring, with robust shock mounting and internal windscreens to permit high SPL applications. Both take advantage of neodymium magnetic structures and active electronics to boost output levels. Instrument models are available from Shure, Audio-Technica, and Royer, while beyerdynamic offers the TG V90r in a handheld vocal format.
Beyond the traditional handheld mic and pencil condenser, other forms have found greater favor for live sound – especially for miking instruments. Small-format condenser capsules are used on wind instruments, stringed instruments ranging from mandolin to acoustic bass, and are even found mounted above piano soundboards.
A variety of dynamic and condenser mics in more “studio style” formats are used on drum kits, percussion, brass, and guitar amps. One manufacturer points to greater creativity from engineers in the field, who are experimenting with mics across many applications and in non-traditional ways.
Extending or modifying a popular model is another way to bring out new features or specialized applications. Neumann raised the low-end response curve of the KMS 104 to create a vocal mic tailored to female vocals – the KSM 104 plus cardioid condenser. Electro-Voice builds upon its classic RE20 Variable-D design with the RE320 that offers a dual-voicing switch to select from two response curves, and at an approachable price point. Audix has created microphone “packs” that offer a combination of vocal and instrument mics in a convenient case, and several other manufacturers offer drum mic kits with different models for kick, snare, toms, and overheads.
Introduced within the past couple of years, Neumann KMS mics are also available in digital versions, with A/D converters within the mic itself. Going from analog to digital right after the transducer gives the benefits of extended dynamic range, an integrated peak limiter/compressor to control clipping, and immunity to electromagnetic noise induced into the mic cable. Specialized software allows remote control of some microphone settings. A-T, Blue, and others have digital mics with USB connectors for direct connection to computers for recording, so further advancement of this technology is likely.
Line 6 provides microphone modeling within its handheld wireless transmitters, approximating the audio response characteristics of a variety of microphone models with the push of a switch. Will some variant of this technology become available in a phantom-powered wired microphone? The miniaturization of circuitry and powerful DSP algorithms could make this possible. Blue’s enCORE 300 handheld incorporates specialized preamp circuitry to stabilize the transducer’s response across all frequencies and SPL levels, further demonstrating that more sophisticated and complex circuitry can fit within the mic handle.
As the rest of the live sound signal chain – from console to loudspeaker arrays – has become more sonically accurate, it’s easier to discern the difference between a good and a great mic, and investing in a better model can make a live show sound more like the recording. Better controlled stage monitor coverage patterns, in-ear monitors, and more precise equalization have made it more feasible to use studio-style condenser mics on stage and still achieve the necessary levels without excessive feedback.
Of course, there’s always room for more innovation and invention – the quest to take it from very good toward that unreachable “perfect.” And some manufacturers, at least, will welcome the creative requests of sound engineers for new models that better solve particular problems.
Perhaps my “blue sky” imaginary miniature mic embedded in the singer’s front tooth – a daydream while sitting at my desk long ago while with Electro-Voice – will become a reality some day, solving the positioning problems inherent in lavalier mics, and the obtrusiveness (at least a couple decades ago) of headsets. Though where to place the mute switch – on the right cheek, left ear lobe, or…?
Enjoy our Real World Gear Photo Gallery Tour of recent dynamic and condenser microphones.
Pro audio writer Gary Parks has worked in pro audio for more than 25 years, including serving as marketing manager and wireless product manager for Clear-Com, handling RF planning software sales with EDX Wireless, and managing loudspeaker and wireless product management at Electro-Voice.
Tuesday, September 10, 2013
How Do I Get My “Big Break” In This Business?
Attitude is the key that will open the most doors
I recently received an email from a reader asking how to get started in the audio business. I had to think about it for a while, hope this helps someone.
I was drafted…several times. I wasn’t even trying to get into the business. I just found a world I wanted to be a part of. Someone told me once to “put your body where your calling is.” Basically, if everything within you craves working in live sound or a studio, do it.
If, however, you’re determined to be there and get paid for it…there might be a problem. Have you ever heard the old phrase “It’s not what you know, it’s who you know?” That’s pretty much it.
If I’m working a show and you walk up wanting to mix for me, not gonna happen. I don’t know you. Your word about your skills and your experience mean nothing. I’m not taking your word. Your word means nothing. Now, if someone I know and trust walks up with you, it might be a different story.
You have to know people, people have to know you. Nobody touches my gear unless I know them. You aren’t even rolling cable for me if I don’t know you. You’re definitely not touching my board during my show. Just asking tells me you’re a knucklehead and I should call security. So your fantasy world that lets you imagine walking out of tech school and getting handed a new console, big check and adoring fans is a joke. Exactly 99.9 percent of engineers didn’t start that way.
How did most of us get in? As volunteers. We found somewhere we had to be. We found people we wanted to be with. We wanted in so bad that we worked for free. Just to be there. Just to be with them. Not for cash or glory, because we just wanted it.
If you want to work in a studio, here’s your fantasy. Walk in, get the job, show everyone the RIGHT way to do everything, mix a perfect album, get rich, etc. Close? Here’s reality. Walk in, clean the bathroom, make coffee, clean the kitchen, make coffee, sweep the halls, make coffee, roll a hundred cables, make coffee, change a light bulb, make coffee, clean the bathroom again. If anything in you panics at the sound of that, go ahead and get over it.
When I first met one of my primary mentors, Larry Howard, I just wanted to hang out with him. I helped him move. I helped clean his house. I eventually moved up to cleaning the studio. Then I was promoted to coffee maker. Demoted after the first pot. Retrained. Re-promoted. After a while I got to roll cables and setup mics. I met a lot of great artists doing that. People who now know who I am. I ended up as second engineer to some legends. I worked with him in three different studios, including one we built together, over 15 years. I NEVER MIXED A SINGLE TRACK IN ANY OF THEM. Did you get that? Never.
I wasn’t there to show him what I could do. I was there to help him. I was there because I wanted to be there and learn. I already knew everything I knew. I needed to learn what he knew. Was it frustrating? Where there times I got mad about it? Did I ever want to just quit? Yes to all. But I got mad when I lost focus and forgot why I was there.
I had a great engineer as a teacher in school. He took us to some amazing studios he worked with and exposed us to things we never would have imagined. He told all of us, clearly, he never used students during sessions. Don’t even ask. So, I never asked. Not verbally.
He took us to visit a legendary studio outside Phoenix. We had a class, toured the studio, looked over the gear and hung out for a demo of what they did. I was in awe. Spectacular. The control room sent chills down my spine. I needed to work that room. So, guess what I did the whole time? While everyone sat around between tours and demos, I made coffee. I cleaned up the kitchen. I rolled cables. Never asked for anything. Just did it. Not my first rodeo.
Near the end of the day, when everyone was getting ready to leave, he came looking for me. I was in the kitchen again, cleaning up coffee cups. He looked me straight in the eye and said thanks for helping out today. He also told me that he didn’t have any sessions scheduled that month, but wanted to use me in the studio when he did. Why?
He never heard me mix. He never read my impressive resume. He never saw my name in a magazine. But, he knew me. He knew what kind of person I would be around his clients. He knew that I didn’t mind doing the dirty work and making him look good. He saw everything he needed in an assistant. Attitude is the key that will open most doors. He liked my attitude.
As for the live sound, I got in like many techs…church.
I was working as a volunteer for a youth ministry in 1992. We did live shows, small concerts, dramas and stuff like that. Standard youth ministry stuff. I knew just enough about sound equipment to be dangerous. When the room needed a sound upgrade, I did it. We hung speakers, wired the stage area, set up a brand new, antique mixer and cranked it up. It was loud. Not crisp or properly tuned, just loud. I loved it. That got me over to the tech side of the ministry.
When we moved, I ended up working at that church too. I was asked to help out with some events and ended up taking over after a few years. Again, all of this was unpaid time, purely volunteer.
I met hundreds of performers and engineers during my time with all these guys. They opened doors for me. They introduced me to the paying gigs. They helped me get on the crews that paid the good money. They got me out on the road. The people who I gave my time and life to made stuff happen for me.
So. What do you really want? Do you want to work in pro audio or do you just want a check? You can get the check anywhere. Flip burgers or flip houses for the check. Work wherever you have to when you start out. The business will probably not pay the bills for a while. If you expect it to, you need to find the closest soup kitchen. You’re going to need it.
But if you just have to work in this world, prove it. Work it for the work. Get to know people in the business. Be humble. They know more than you. You need to learn from them. Once you get that opportunity, don’t blow it. Don’t lose focus. Guard your attitude. The guys with basic tech knowledge and crappy attitudes will more likely end up as roadies. Not techs. The attitude is what moves you up or down in the business. Good luck.
NOTE: Erik’s new book, The Art of the Soundcheck, is now available. It’s here.
M. Erik Matlock is a 20-plus-year veteran of pro audio, working in live sound, install, and studios over the course of his career, as well as owning Soundmind Production Services. Erik provides advice for younger folks working (or aspiring to work) in professional audio at The Art Of The Soundcheck—Random Stories and Wisdom from an Old Soundguy. Check it out here.
Exploring Converging Techniques For Tuning Line Arrays
Viewpoints from "in the trenches" versus the ivory tower of a "factory marketing geek"
A few years ago, I was asked to make a presentation about equalizing line arrays at a concert sound seminar.
My specific assignment: discuss the equalization of line arrays in concert settings, in combination with Robert Scovill detailing his approach towards the same. Since we’d never formally met, and Robert’s background is from an “in the trenches” viewpoint while mine is largely from the “ivory tower” of a “factory marketing geek,”
I was a little apprehensive that we might have clashing techniques. This proved to be anything but the case.
Robert’s presentation was very informative. His first request to touring companies when he’s mixing on a line array system is to “please load the factory specified presets into the digital loudspeaker processor.”
This was music to my ears – I’ve been preaching that “rolling your own” settings is an almost impossible task with line arrays. Even with a measurement system, a line array’s response is different at every distance. And changing crossover frequency and delay settings away from factory presets can also cause unexpected polar-response changes that can come back to bite later, and it’s difficult to even know the cause of the problem.
Based upon his experiences, Robert is aware of these pitfalls and thus always starts with the factory presets, knowing that manufacturers have spent countless hours optimizing the basic settings for the best overall results. Given this solid starting point, equalization modifications to account for room coupling at low frequencies, as well as the number of boxes and the splay between each, can both be determined. How, then, is this to be done? (More later…)
My presentation kicked off with an explanation of how long an array needs to be to act as a true line array at a particular frequency, and that many so-called “line arrays” are simply vertically oriented loudspeaker clusters that sound nice because the factory preset is well done.
Next, I demonstrated how a particular frequency will generally follow the -3 dB per doubling with distance slope within critical distance from the array, but will oscillate at a unique period. This means that at a given distance from the array, adjacent frequencies may be up or down several dB relative to each other, and that this can be the opposite at a different distance.
Therefore, line array equalization MUST be done using an average response, comprised of several measurements made at difference distances from the array.
With this in mind, what was Mr. Scovill’s approach to equalizing line arrays? The same, accomplished by averaging several Smaart measurements at different distances from the array. (This warmed the cockles of my factory-geek heart!) He even insisted on PZM’ing (pressure zone miking) the microphone on the floor for each sample measurement, something I’ve been recommending for quite some time.
One of Robert’s key points is that the closest measurement should be about one-third of the way into the seating/coverage area, not using the array’s response in the first several rows of seating.
Two reasons for this: first, the near-field loudspeakers used for front fill are not really part of the array due to their greater splay and minimized interaction with other loudspeakers in the array; and second, any array has gradient side lobes (top and bottom of a vertical array) which exhibit a radical, unique frequency response that should not be part of the array’s average response.
I learned to EQ an average response of an array from the guys at Duran Audio, who make the Axys Intellivox loudspeaker system, and who have been steering broadband straight-line arrays for 20 years.
They recommend logarithmic spacing of sample measurements, which, with a straight-line (non-“J” array) array, puts each measurement in roughly even vertical angular increments about the center of the array’s main lobe.
There too, I don’t use the “up-close” position of the front seating rows because the gradient side lobes are not typical of the array’s general response.
Starting at the one-third point in the house, logarithmic distances dictate only two to three measurements in most venues. For example, begin with a sample at 40 feet from the array, leaving only samples from 80 and 160 feet to be taken.
At the warehouse hosting the EV demonstration, there was a back wall at about 120 feet. We used four Smaart memories for measurements, taken at 40, 60, 80, and 100 feet.
DATA IN ACTION
Let’s have a look at the Smaart transfer measurements, made after the many cooks involved in the seminar came up with their own final EQ settings for the EV XLD-281 line array.
The XLD-281 uses two 8-inch woofers and two 2-inch-diaphragm/1-inch-exit compression drivers in a 3-way, band-pass configuration.
The measurements were made with the subwoofers turned off. Figure 1 shows the sample made at 40 feet, and Figure 2 was taken at 60 feet.
Figures 1 and 2
Figure 3 and Figure 4 were made at 80 and 100 feet, respectively.
Figures 3 and 4
Figure 5 shows the average of the first four samples. Be aware that when you look at the coherence in Smaart for a line array, it will not be very high at the short wavelengths/high frequencies, due to the fact that many of the sources are several wavelengths farther than the nearest driver.
Note in Figure 6, where all the samples are overlaid, that there is 10 dB to 12 dB of difference between individual measurements at particular frequencies.
Figures 5 and 6
This is proof that equalizing line arrays at one position (front of house comes to mind) by ear or measurement is a fool’s errand.
Looking at the average , 125 Hz is at “+6” with a fairly smooth roll-off to “-6 “at roughly 12 kHz.
With the exception of Figure 3, made at 80 feet, which has an unusually pronounced summation of high-frequency drivers from 5 kHz to 10 kHz, the rest of the measurements are substantially lower in energy in that range.
To benefit the other sample distances, I would be tempted to 3 dB boost to the 5 kHz to 10 kHz range, to see if the overall sound at various listening distances wouldn’t have a little more “air” and define the high end better.
I’d also be tempted to EQ the 3.5 kHz range down by 2 dB to 3 dB or so to get it more in line with the rest of the spectrum.
Once these changes were made, and the subs were added back in, one would have the typical “bass-haystack” added to a flat response that we all know and love!
John Murray is a 35-year industry veteran who has worked for several leading manufacturers, and has also presented two published AES papers as well as chaired numerous SynAudCon workshops. He is currently the principal of Optimum System Solutions, a consulting firm.
Church Sound: Have You Been Caught By These Five Production Surprises?
Don’t get caught without a plan!
Fly fishing Sunday night, I cast my new imitation blue dragonfly onto the water. A small fish, about five inches long, took the fly and the fight was on! OK, it wasn’t much of a fight. Not yet, anyway.
I pulled in my line, with the little fish in tow, when WHAM I got the surprise of the night. The WHAM was from a largemouth bass gulping down the fish on my line. No longer was I pulling in a tiny fish, I was now fighting with a 2- to 3-pound bass.
In my younger naive days, I would have fought with all I had and promptly snapped the light line. This time, with wisdom on my side, I let the bass take all the line it wanted because the only way I’d land it was if I slowly tired it out.
Production surprises, like my fishing story, can come at any moment. Whenever ANY type of surprise occurs, ranging from a mid-service equipment fail to a last-minute request for an extra microphone, the most important part of dealing with it is how you react…and having a plan makes all the difference.
I’m Talking Total Catastrophe
My church runs a digital system (mixer, snake, IEM system) all hooked into a couple of mix engines. If one of those mix engines fail, and they have failed, what do I do?
I could frantically run around and check every cable connection while muttering a few words under my breath. The congregation would notice my anxiety and the pastor would certainly notice it. Now, everyone is uncomfortable and they don’t know what will happen next.
Contrast that with a reaction in which I say there is an issue which will take about 5 or 10 minutes to resolve. I shut down our system, have the stage manager shut down the stage rack units, and then do a full system reboot, loading the saved settings into the system.
It’s OK that surprises happen. They happen to everyone. The difference is in how people react. Preparing your list of possible surprises and your reactions is a great way to be ready when the next one hits. The next time it happens, you will know exactly what to do.
Five Common Areas For Surprises
1) Last-minute requests. Conversations concerning last-minute requests can get into debating whether we “allow” these or not. For example, if a musician came up two minutes before the service and said, “instead of in-ears, I want to use a floor monitor.” Let’s push that conversation aside and talk about last-minute surprises that we have to accommodate.
A great example of a last-minute request with significant impact is that of adding a musician. Let’s say the worship leader comes up to you right before the service and says, “Chris is back from college and really wants to help lead worship today. Where can he plug in his guitar?” Let’s go with the assumption you have room on the stage and an available channel.
Plan on working through three areas; stage needs, monitoring needs, and mix changes
Start by evaluating the stage needs. You’ll need a stage drop for his guitar. Does he sing? Ask if he needs a vocal microphone. Next, where should he be on the stage? If he’s running through an amp then you don’t want that amp pointing at the keyboardist. You need to find the best setup given the existing stage arrangement.
Second, what about monitoring needs? If you have an in-ear system, do you have an available unit for him to use?
Given the last-minute arrangement, he doesn’t have time to set a great monitor mix in the in-ears. He could go up and dial in a close approximation…which you’d want him to do.
If you have floor wedges, consider an existing floor wedge monitor he can use with a mix close to his needs, such as that of another guitarist.
Finally, you’ll need to set his gain. If at all possible, do this before the service. If you don’t have the time, tell him you’ll do it during the first song. In that case, make sure other musicians don’t have his channel in their monitor as you don’t want to throw off their mix.
If you do set the gain during the first song, let him know when you are done so he can set his in-ears if he’s using the in-ear system, or bring him up in his floor wedge if he has his own. There are a lot of variables in this situation. These are variables known only to you.
I will say this about those last-minute additions; don’t go for ideal, go for good. This would be true of his monitor mix and definitely your own mix. Now you are mixing in another instrument that wasn’t in the original mix during the practice.
I’d go with fitting this additional guitar into the existing mix compared to re-mixing everything. Cut out a bit of frequency space and fit the guitar in there. But then, as you know, every band and every song is different, so do what is best in your situation.
2) Equipment failure. A few weeks ago, I was following a twitter thread on a church that went acoustic because of a critical equipment failure. As you can see, equipment failures can be drastic or they can be as simple as a dead microphone battery.
Simple equipment failures would be problems like a dead microphone battery or a cable that stops working. In these cases, have equipment in place that can be quickly pulled into use.
For example, you could keep a spare vocal microphone (wired or wireless) on a stand which could be grabbed if a microphone went out during the service. In the case of cables, have a few spare cables at the ready so they could be swapped if necessary.
You can go the emergency route and keep a wired mic on a stand with a DI taped to the bottom, with cables at the ready. In case of an emergency stage equipment failure, you have a wired mic, DI, and tested cables available.
What about massive equipment failures? Mixer failure? Amp failure? You should have a plan in place for dealing with such issues.
If your mixing console dies, can you pull in a backup mixer or the mixer from the youth room? Would it work with your main sanctuary equipment? What is the least that you could do to make it work? Consider this thought; all you need on stage is one microphone.
In the worst case scenario, as long as you can get one microphone working on the stage and playing through the system, you’ve got enough to make it through the service.
Outside of the idea of spare equipment, what can you do to minimize the impact of an outage? I caught an article the other day by Ken DeLoria that spoke on this very idea. One idea Ken put forth was in arranging amplifiers so they powered house loudspeakers across different frequency ranges:
“The amps can then be circuited so that one unit can power LF [low frequency subs] and HF [high frequency speakers]; another LF and MF; another MF and HF, and then repeat as needed. Doing so offers three distinct advantages: First, if an amp fails, only a section of the system will go dark, rather than an entire bank of LF, MF, or HF drivers.
“The second advantage is that the amplifier’s power supply, which is typically common to both channels in most two-channel units, will not be stressed as hard as it would be if both channels are only powering LF, because LF normally requires a lot more power than MF and HF. Headroom and maximum available power to the LF load will be improved, and with the lower overall demand on the power supply, reliability and longevity will increase.
The third reason is that you’ll only need one amp model for spares, rather than several. While there is a cost penalty to configuring a system this way, the collective benefits may very well outweigh the added expense.”
3) Feedback. Surprise! This is an area for being proactive and reactive. Being proactive means you are watching the stage and listening for the beginning of feedback.
If a singer is standing close to a monitor and starts to drop the microphone to their side, then immediately mute the channel. If you hear a bit of ringing as the start of a full-on feedback attack, EQ out the feedback by cutting back on the frequency that’s getting excited. While the feedback did occur, as you caught it early and dealt with it, I’m calling this being proactive.
There is a time to be reactive. When you get a sudden full-on feedback attack, pull your house main back immediately until it goes away. You now have a few extra moments to look at the location of instruments and monitors on the stage and to review your mixer settings.
Once you’ve done that, you should be able to identify the source of the problem and take appropriate action, such as turning down the gain on a channel or pulling a volume back. Preferably the latter so you don’t mess up your monitor mixes…but there is a time and place for everything.
You can plan for feedback issues by training the musicians in proper microphone usage, as a means of proactively preventing feedback. You can also plan for it by knowing what you’ll do when it happens, as I explained earlier.
4) Whoops, you forgot to… I’ve been guilty of this one. It’s usually a small mistake on the board but it has a severe impact.
For example, you have all of the band’s channels on and faders up but you forgot to bring them up in the subgroups, which is why you don’t hear them in the mains when they start singing. Most of the “I forgot” problems are related to being heard.
In rectifying this type of situation, also bring that channel’s fader down, make the correction, and then fade their volume back up. You don’t want a sudden jolt of sound. Yes, the congregation will likely know a mistake was made, but fading in the correction is less distracting then the sudden jolt of a full-on sound.
5) Grabbing the wrong microphone. Surprises happen when people on stage do something unpredictable. A perfect example is when a person grabs the wrong microphone. Don’t waste your time worrying about that fact that it’s the wrong microphone, just turn on that channel and make whatever EQ/effects adjustments are necessary.
For example, if a person walks up to read scripture and grabs a singer’s mic (the mic you set up with heavy reverb) instead of the speaking mic, then turn on the mic and turn off the effects. You can make EQ changes if needed but first get them that clear/non-distracting sound.
6) Snap, crack, POW! I’m ending the list here because you should remember you can’t control everything. I’ve had a guitarist unplug his rig at the end of the worship set when a song was still in the last a ccapella chorus…POW! Why he opted to unplug at that time is beyond me—he had never unplugged early before. And while I can watch the stage for some things, there are some things I (you) can’t catch because it’s completely unexpected.
The Take Away
You can’t plan for everything. You can’t control everything. That’s OK. But, there are surprises that aren’t really surprises at all because you know they are bound to happen. Plan for those. Prevent them proactively.
When they do happen, confidently and calmly deal with them according to your plan. And if you don’t have a plan…relax, think, and proceed. Anxiety doesn’t help anyone. Welcome to the world of live audio.
As a follow-up to the fish story, the bass finally won. Eventually, the fish made a fast run for deep water and snapped my line.
upon my line.
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, and can even tell you the signs the sound guy is having a mental breakdown. To view the original article and to make comments, go here.
In The Studio: The 440 Hz Conspiracy (Includes Video)
The back-story on the standard A note frequency, and an alternative
When it comes to tuning an instrument, we think of the standard A note at 440 Hz as the reference standard, but it wasn’t always that way.
Prior to 1940 there were a variety of standards, although A=432 Hz (also known as “Verdi’s A”) was the one most frequently used. It wasn’t until 1940 that the US adopted A=440 as the standard, with the rest of the world following in 1953.
But why did the world change in the first place? For one thing, A=432 is supposed to be a more “natural” vibration based on the fact that it’s divisible by 3, unlike A=440 which is only divisible by 2.
A=432 is said to just feel right, and when tuning without any pitch reference, trained musicians are said to automatically tune their instruments there, and the ancient Egyptians and Greeks have also been found to have tuned their instruments at 432.
The science of Cymatics, which is the study of visible sound and vibration, is apparently on the side of A=432 as well, as you can see from the graphic on the left.
The physical nature of the two frequencies are pointed out by Dr. Leonard Horowitz in his paper Musical Cult Control, but he goes even deeper into what he thinks are the reasons why we went from A=432 to A=440.
Horowitz claims that there was a conspiracy between the Rockefeller Foundation and Nazi propaganda minister Joseph Goebbels in changing the standard because"herding the populations into greater aggression, psycho social agitation and emotional distress” was necessary to create a war mentality. Supposedly the Rockefeller Foundation had strong financial interests in weapons of war at the time, and of course the Nazis had strong interests in, well, war.
But the ultimate test is listening, so here are two versions of the same piece—one tuned to A=432 and the second at A=440. See which one you prefer.
I liked the A=432 better, but then again, I believe that the guitar should be an Eb instrument because it just feels and sounds better.
Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blogs.
Pro Production: Learning Lighting Programming & Operation
Deriving original looks, concepts and chases and then bringing it to life
The automated lighting programmer must have many skills beyond knowledge of simple programming syntax. The position requires that one evaluate each situation to determine the right method of operation. Some productions hire a programmer to handle all aspects of the lighting, while others hire a programmer to bring a lighting designer’s (LD’s) vision to life.
Real-world experience with many productions is the only way an automated lighting programmer can become successful. Knowledge, speed, accuracy, people skills, and so on are all vitally important, but there is no substitute for experience.
There are many different levels of productions, each requiring specific types of people on the production staff. Understandably, there are several different categories of automated lighting programmers. Each holds an important position within our industry, by providing different levels of experience and knowledge.
First, there is the weekend warrior. This type of person simply programs lights for fun, but has another main profession. The weekend warrior has little to no interest in learning more about the profession.
Next is the amateur programmer. Amateur programmers program lighting when and where they can (schools, churches, clubs, raves), but programming is not their main source of income. They have an interest in the profession and strive to learn more about automated lighting programming.
The apprentice programmer is involved in the lighting industry and programs whenever there is an opportunity. Oftentimes apprentice programmers are hired to work on productions in positions other than lighting programming (technician, followspot operator, etc.). They have a desire to gain as much programming experience and comprehension as possible.
A professional programmer earns his or her living by programming automated lighting. The majority of this person’s income (80–90 percent) is from programming. Automated lighting programming is the chosen career of professional programmers, and they continue to study and improve their skills as much as possible.
Finally, the professional programmer/designer works regularly as either a lighting programmer or lighting designer. Oftentimes this person will be hired onto a production filling both rolls simultaneously. The income of this professional is split between programming and design. Continued improvement of knowledge and experience is continually sought by this career type.
Of course, like most jobs in the entertainment industry, some people will work on high-profile productions (award shows, Broadway, large tours, television shows) and get lots of press. Their names will be well known in the industry and they will be mentioned in magazines and on websites.
Others will do many shows, but will not be well recognized within the industry. Yet these individuals will still be successful programmers with a grand career.
The Hollywood Syndrome
Our industry is only a small part of “show business,” yet it still lives up to many of the clichés. Many students and apprentices of the profession expect a fast track to the “big” shows. They see the concert tours roll through town and the live television broadcasts and think, “I can do that.”
Oftentimes these people will begin working with a lighting company and will not understand why they are not going out on the next big U2 or Rolling Stones tour as the programmer.
There is a good reason you see the same programmer’s names on all the big shows: experience. While anyone can learn which buttons to press on a console, it takes many years of programming to learn how to get the most out of your fixtures, work with different types of productions and LDs, and handle any situation that is thrown at you.
Ultimately, the actual data in the console is not what is important, but rather the end result. If you are able to create the LD’s vision in a timely manner and write the cue so that it is repeatable, then the methods of the data creation and storage are not essential.
Recently, I was hired to program a live television special. The LD hired me because we have worked together before and he knows that I will come to the event and simply do my job. I have learned from my experiences not to wait for him on every detail of the show. He knows I will create original looks that fit with his style and work for the television camera.
In fact, he even told me that he is not worried about what patterns are in the fixtures, as he knows that I will “work my magic and create new visual images.” If I had only programmed one or two shows in my life, he would not have wanted to hire me for this production.
There is an extremely short amount of time from load in to taping, and he has no time to sit with a programmer to describe every bit of the show. So do not be in a hurry to jump into the “big shows.” Instead, take your time and work hard, and the large productions will come to you. You will learn more about lighting on every production you are involved with and you should enjoy all the new challenges.
Creativity And Consistency
Usually lighting programmers are hired not only to assist in the creation and storage of an LD’s vision, but also to share in the creative process. It is extremely important that an automated lighting programmer is both left- and right-brained. It is often said that one side of the brain is technical and the other creative.
A programmer must be able to derive original looks, concepts, chases, and then utilize the tools at hand to bring these ideas to life. Whether you are a highly creative person or not, there are many books and exercises on the subject of creativity. I strongly recommend exercising your brain as much as possible.
The technical side of a programmer’s brain must contain the data needed to properly use a console and fixtures. In addition, regular, consistent routines should be used in console setup.
For example, if you always number your colors or positions in a particular order, then no matter what show or console you are using, you will know that color 3 position 5 equals “down center stage red.” Do not just randomly lay out your console with each show. Of course, there will be aspects specific to each production, but if your basic building blocks are the same, then your programming will be much faster and efficient.
Learning To Program
Once, while I was in Tokyo, Japan enjoying fine food with friends, we discussed the puffer fish (fugu). If the puffer fish is not properly prepared,
then it can lead to tetrodotoxin poisoning, which has a 50 percent mortality rate. In Japan, only specially licensed sushi chefs are allowed to prepare and serve this dish.
In fact, the U.S. Food and Drug Administration (FDA) allows properly prepared portions of the puffer fish into the United States only two to three times a year. The FDA’s agreement with Japan states, “Experience has shown that the best method for obtaining a product which will not cause illness or death is the highly specialized training and knowledge for product preparation.”
Although an extreme analogy, automated lighting also should not be taken lightly.
Luckily, the mortality rate for improper programming of automated lights is extremely small—although I know some LDs who have wanted to kill their programmers! However, lighting programmers must practice their craft and continue to learn. Consoles are always improving, new fixtures are released, and creative visions change. There are many resources to help you learn how to program, but practice makes perfect.
Look For Opportunities
Instead of waiting for the next gig to hone your skills, you need to find other avenues. The first place to look is your local lighting shop. Many companies will be willing to allow you to come to the shop and use a desk. While they may or may not have fixtures for you to plug in, at least you can use the console.
And, as a bonus, when you are hanging out at the shop, you might just be offered a gig. If you can get your hands on a desk, build a practice show from scratch with cues and everything. Do not just sit in front of the desk and poke around. Put yourself into a real-world situation and complete the required tasks.
If you do not have access to a console, you can always make use of offline editors. Most automated lighting consoles have applications for the personal computer (PC) that emulate the desk. Using this software you can practice the syntax and procedures of the console.
In addition, many of the offline editors now either include visualization or work with popular visualization software on the same machine. This means that you can sit at home and program virtual lights on a virtual console with your real computer.
The main reason to exercise your programming skills is so that the console functions become second nature, allowing you to spend more time being creative.
When you do not have to think about how the console works, an amazing ability comes through. You find yourself simply commanding the fixtures to create the desired looks without thinking about how to enter the data into the console. Of course, there will be times that you will be challenged by the console, but the more comfortable you are with the programming syntax, the better.
There are many types of exercises you can do, and I will suggest my favorite. Put yourself into the following scenario. You have been hired to program the lighting for a small 2-day business meeting using about 20 fixtures.
Late in the first day, the client surprises you by informing you that a band will play during the lunch break the next day (a 1-hour period). The client wants you to “do lighting” for the band. It is now 7 pm, and you can only be in the venue until 10 pm. So you have 3 hours to program lighting for a band that you know nothing about (not even what type of music).
The exercise is to program 20 fixtures for 2–3 hours to prepare for this surprise. Then ask a friend to grab a mixed collection of compact discs (CDs). Have your friend randomly select a CD and song and play it for you.
Alternately, you could use a random playlist on an MP3 player; just ensure that it plays various types of music. Play back your programming to the various music types. Then have your friend randomly select another CD and song and play back to that one.
Keep doing this for about an hour, and you will find out whether you prepared yourself (and your desk) for anything that might come up. I find this exercise to be very consistent with real-world situations where you have to program and operate lighting for acts you have never seen or heard.
Explore Your World
A large part of being a good programmer and operator has nothing to do with the console. Your timing, rhythm, listening, visualization, and many other skills are just as important. Many of us often can’t help but imagine lighting cues while listening to music, but how often do you really listen to the beat, changes, and so on?
Instead of trying to visualize the look of the actual lighting, try just thinking about when to trigger the different cues. Learn to anticipate changes in
the music and recognize musical elements. Listen to all types of music, not just what you like. Even though your production may not contain musical elements, these skills will come in handy in most situations.
You can also exercise your mind by trying to think of ways to recreate natural lighting conditions. Pay attention to how the quality and color of light changes during a sunset or sunrise.
One day watch a sunset on the horizon for 20 minutes, and then the next day watch a sunset on the side of a building or a tree. Sit in a dark house during a lightning storm, paying attention to Mother Nature’s lighting chases. These exercises will pay off even if you do not recreate these actual situations on stage, because they might inspire you to create an effect in a different manner.
Never Stop Learning
If you think you have mastered a console, think again. There is always something new to learn. Talk with others to see how they accomplish certain functions.Also, try doing things in different ways. If your desk has a very strong effects package, try building a simple 30-step chase “old-school” style. You will find yourself someday in a situation where an LD wants an exact look that cannot be created using effects.
For example, while I was working on an ice skating show the LD asked for a very specific chase. I thought I could build it with effects, and he thought it would have to be programmed as a chase. We were ahead in our programming schedule, so he gave me the time to try to create it with the effects. He was correct—it was not possible.
I then quickly built the chase as a 90-step cuelist and it did just what he wanted. Luckily, I had the experience and knowledge to create this monster chase in a hurry.
Be An Artist
There is a true art to programming automated lights. It is a skilled craft that requires many years of experience to fully master the possibilities. Because every production has its own unique challenges and requirements, the programmer must be fully confident in his or her abilities with the console and fixtures.
Yes, we are part of the creative arts, but we also perform a highly technical job. Just as a fine sushi chef must train for years to perfect the slicing of a puffer fish, we must maintain a high level of craftsmanship for our profession.
Go here to acquire The Automated Lighting Programmer’s Handbook, 2nd Edition, published by Focal Press. Use the promo code FOC20 during checkout to receive a 20 percent discount.
Brad Schiller has more than 20 years experience in the lighting industry. He has worked as a technical director and lighting designer, as well as an automated lighting programmer, and has experience with various types of productions including theatre, television, concerts, film, architectural, dance, and industrials. Project favorites include: The 1996 Academy Awards, the Capitol of Puerto Rico, The Sydney 2000 Olympic Games Opening and Closing Ceremonies, The Crystal Method, and Metallica.
Pro Production Video Tech Tip: DVI & HDMI—Making The Connection
Accurate distribution of the image information
The plain truth is that digital signal distribution is the best way to go for image fidelity.
Part of the reason why analog-distributed images often appear deteriorated stems from the fact that there can be significant line loss in the transmission of the signal. The nature of digital signals allows accurate distribution of the image information, regardless of how many times you split and route the signal.
Of course, there are many other factors to be considered, but they all have to be founded on the signal itself being a clean reference data stream. It is time to make the switch to digital.
The Digital Visual Interface (DVI) standard was introduced to provide a digital path for the graphics world. It has since expanded to allow other image sources to provide a digital connectivity medium to displays. Now we can find DVI on commercial and pro equipment as well. There are also digital cinema products that support the DVI standard.
In all, there are three types of DVI signals: single, dual, and twin. First, let’s go over what they have in common.
DVI uses a digital RGB transmission format. All signals are based on TMDS (transition minimized differential signal), which preserves the quality of high-speed data. All DVI formats use the same connector, but in some cases the connector may not be fully populated with connection pins.
This is the most commonly used DVI type. DVI transmission is based on the RGB ratio design. A single link DVI transmission can handle up to 165 MHz signals at 8 bits per color, red, green and blue. This means each pixel has 24 bits of color depth.
Single link (above) and dual link DVI connectors.
Dual link adds a second set of red, green and blue data paths. This allows signals up to 330 MHz in bandwidth. As you see from the picture below, an additional six pins in the center populate the dual link connector and differentiate it from the single link connector.
Twin link allows bit depths of greater than 24 bits per pixel. This is achieved by having two cables running in parallel. They can either be two single link cables or two dual link cables.
Twin single (above) and Twin dual connectors.
A pair of single link cables will provide greater bit depth for signals up to 165 MHz. A pair of dual link cable will provide greater bit depth for signals up to 330 MHz.
HDMI is based on the DVI standard and adds several more features to the DVI transmission protocol and the connection type. The main goal of HDMI is to simplify connectivity in the consumer market.
There are two types of HDMI signals: Type A and Type B. First, let’s go over what they have in common.
All HDMI cables support the transmission of video, audio, and control, all in the digital domain. This design is intended to minimize the number of interconnects that are required. HDMI can transmit RGB, YcrCb 4:4:4, and YcrCb 4:2:2. The default is RGB, to assure HDMI provides full compatibility with the DVI standard.
The Type A HDMI connector supports up to 165 MHz of video bandwidth at 24 bits per pixel. The audio format can support stereo, surround sound, and up to eight channels of one bit audio. Devices connected via HDMI can communicate with each other and be controlled via the CEC (Consumer Electronic Control) protocol, as long as they are designed to be compliant with the CEC standard.
Type A (above) and Type B HDMI connectors.
The Type B HDMI connector includes all of the signals of the Type A connector, but also adds another set of RGB signals, creating dual link. This supports up to 330 MHz of video bandwidth at 24 bits per pixel. In practice, one will very rarely find this connector in a current application.
Dual link is primarily used for very high resolutions, beyond 1920 x 1080, or active 3D. In these cases, the DVI connector is preferred because it provides more rugged construction.
This article provided by Digital Projection.
Posted by Keith Clark on 09/10 at 04:09 AM