So… What is the Difference Between Mic and Line Level?

How many times have you plugged in your PA, only to find that you just barely have to crack open the volume knob on your mixer, powered speaker, or “other”, and you’re either getting crunchy distortion from the iPod you have running, or blasting feedback from an attached microphone?  Or maybe you have the opposite problem. You find you need to turn the input and/or output volume all the way up to get an acceptable audio level, (or maybe it’s still not loud enough, and… “Where in the world is that unusually loud hiss coming from!?!”)

Many times my mind starts running down the laundry list of gear that could be broken

–      Bad cable?

–      Mixer? 

–      Power amp? 

–      Microphone? 

Sooner or later, after all components are verified as working and the issue remains, there is one other cause that has literally been staring you in the face (if you’re looking at the mixer).  Welcome to the topic of this blog entry: Proper Gain Staging in audio systems.  “WHAT is THAT!?” you say?  I am so glad you asked. ;)  Here we go:

Most people that work in the A/V industry know that you can’t just plug a microphone into a rack of power amplifiers and expect the attached speakers to make noise loud enough to be heard by anybody.  Why is that?  It’s usually an XLR cable connection on both ends, so why doesn’t it work?  And likewise, most people with even a little experience in this kind of work also know that you can’t take an output from a power amplifier, and connect it to a mixer input without frying the mixer… or power amp… or both.  Thank goodness (nowadays) that most manufacturers have made the respective cable/connector types on those two pieces of equipment so incompatible, that it would be hard to make that particular scenario happen.  If you don’t take anything else away from this little essay, here is the point I want you to remember:

There are three “operating levels” when it comes to electronic audio signal transmission.  Listed here, in order of “quietest to loudest”: 

1) “Mic” level  (usually measured in millivolts),

2) “Line” level (usually measured around one volt),

3) “Speaker” level (usually measured in multiple volts).

 

Speaker level voltage can vary greatly, all the way from a few volts for a small speaker/amplifier combination, to almost 100 volts for an extremely large speaker/amp combination.  Quite possibly you have heard these terms thrown around, or seen them silkscreened next to an input or output jack on a piece of gear.  Steps 1 through 3 above are the basic steps that an electronic audio signal needs to go through in order to achieve the necessary voltage amplification to move a speaker cone.  And here’s how it goes through those steps:

           

When a microphone picks up an acoustic sound wave, it generates a very small voltage waveform.  This is the “raw” voltage level created by the microphone.  And, as noted above, is usually best measured in microvolts.  Over the years, this type of audio voltage level has been given the nickname “Mic Level”.    Mic levels are very, very small.  That is why in nearly every case I have ever seen or heard of, that signal needs to be amplified to a larger level. 

Once it is amplified to a workable voltage level, it can be routed, mixed, and distributed.  Enter the “Preamplifier”.  This piece of electronics is a critical component in almost any audio system in existence.  It can be a no-frills circuit with a set voltage gain that is built into a piece of equipment that you never even notice, or a several thousand dollar stand-alone rack mountable piece of premium gear, 

Hey man nice preamp!

(with hundreds of options in between).  High-end preamps are usually used in recording studios, or possibly on the lead vocal mic in larger scale live concert rigs.  Most often though, this is the rotary knob that you see at the top of a channel strip on just about any mixing console surface, small or large.  Even the digital consoles will have this component located somewhere on or in the control surface.  With a digital console it may either be a software knob, or hardware knob, but whatever form it takes, it will still be there.  The microphone preamplifier has been given several different nicknames over the years, and different manufacturers have labeled it with their own favorite terminologies.  For instance, Yamaha prefers to call them “head amps”, as seen in the silk-screening on the console surface or console software, which is typically abbreviated “HA”,  (if you have ever wondered what the heck “HA” has to do with it).  You will also hear people refer to them as a “mic pre”, “pre”, or “mic trim” (trim is sometimes a different knob from the preamp control knob on fancier mixing consoles, however it is also what other manufacturers, like Mackie

have termed their preamp control).  I will say more about the nuts and bolts of dialing in mic pre’s in paragraphs to come, but the overall design objective of the preamp is to get the voltage signal generated by the microphone to a “loud” enough level that it can be manipulated without any further voltage degradation, often the phrase used here is “signal loss”.

XLR

Let me take a paragraph now to discuss the physical connections on the back of an analog mixing console, and also introduce another important electronic circuit.  Quite often you will have two physical input choices for every channel on a mixer.  One is an XLR jack, and the other is a ¼” (quarter inch) jack.  Typically, the silk screening next to each input jack will say “mic input” for the XLR, and “line input” for ¼”.  Aside from the physical difference in connectors, there is an important electronic difference as well.  The XLR jack on the back of a mixer is generally equipped to accept signals from microphones, because they need quite a bit of preamplification to work properly.  The ¼” jack is designed to accept signal sources like CD players, ipods, and computer audio outputs.  The console is expecting that the voltage levels from these devices will to be larger than that of a raw microphone, and because of that, they do not need nearly as much pre-amplification.

1/4"

May I now introduce the “pad” circuit?

This is an attenuator circuit that is sometimes built inline to the ¼” connector’s signal path before it even gets to the preamp circuit.  Midas, for example, builds in a 10dB pad into the ¼” inputs on their Venice consoles.   This is why a CD player with a hot output signal may overload an XLR input, but when plugged into the ¼”, will then be at a workable level.  On a more full featured console, this may also be an additional circuit that can be inserted into the signal path with the push of a button. This button is almost universally physically located next to the preamp knob on an analog console.  Different manufacturers will design different degrees of attenuation into these insertable pad circuits; anywhere from 10dB to 30dB, but it seems that 20dB is the most common.  The pad circuit is extremely useful for knocking down the input level on a very “hot” signal, which would otherwise overload the preamplifier and cause noticeable (and unacceptable) distortion.

This is where we finally get to the heart of gain staging.  Between these two electronic components, the preamplifier and the attenuator pad, we now have the flexibility to accommodate any level of voltage signal presented at a mixing console.  The overall goal of these two components is to get all of your input signals, as discussed above, to a workable voltage level so they can be routed, mixed, and distributed as needed within the mixing console, and beyond. 

Ideally, we want to get each and every one of our input signals to be as strong as possible without distorting.  This is because there is a certain amount of inherent noise in every electronic component inside a mixing console, and the entire audio system as a whole.  It is important to keep the signal voltage well above this inherent “noise floor” throughout the entire audio signal path so the resulting “signal to noise ratio” at the very end of the signal path is as high as possible.  Why is this a big deal?  I will give you an example from personal experience:

In the fall of 1997, I was getting my audio rig for “Anything Goes” ready to hit the road for a national tour.  We loaded into the first theater, and I started to rough in volume levels for the actors 25 wireless systems, plus all the mics from the orchestra pit.  As our workday ended and the theater quieted down, I couldn’t help but notice a hiss in the idle system while the pit orchestra mic faders were live.  Being mostly familiar with the equipment, I knew that this hiss noise was not something that should be this audible.  After a short bit of troubleshooting, I realized that the issue was the way I dialed in my preamps for all the pit orchestra mics.  I neglected to add enough gain to my preamps on each of my channel inputs for those mics, so the console was adding system noise as the signal made its way through each of the various mixing stages.  I went back the next day and added more gain at the mic preamps.  I was then able to dial the outputs of my main EQ back a bit, (thereby keeping the same apparent loudness to the audience) and was able to maintain a much more acceptable noise floor.

This kind of issue is exponentially increased with each signal you add to a mixing console.  It is cumulative.  Perhaps one mic might not be such a big deal, but when you start adding the noise of each channel together into a single, summed output, it can be very noticeable even to the casual listener.  The opposite problem can also be an issue as well, when many very hot inputs are summed together, it can actually overload a mix or output buss on an audio desk, and cause it to distort.  This is most evident in concert audio situations, where input levels can get hotter as the concert goes on, because each of the performers start pushing harder throughout their performance.  A good engineer will plan ahead for this, and leave themselves a little bit of headroom in the system inputs, (and outputs) so they will not have to back everything down (or “hit a ceiling”) later in the performance after distortion issues start to appear.

           

Let’s return to a more practical application for the average A/V tech.  Let me give you examples of common gain staging issues I see in small audio setups on a day-to-day basis. 

-Quite often I will see systems that use ceiling speakers permanently installed in a facility (hotel meeting or conference room, etc). 

-Someone has plugged a small Mackie mixer plugged into an XLR input jack on the wall of the room, and I hear a noticeable hiss coming out of the speakers. 

-Even though the master output level on the Mackie is turned way down, the audio sounds a bit “crunchy” when I speak into an attached microphone. 

-Almost every time I run into this situation, it is because the mixer output has been patched, (currently outputting a line level signal), into an XLR jack on the wall that is expecting a mic level input, and the signal is overloading the mic preamp attached to the XLR wallplate.  It just so happens that most small Mackie mixers (and other manufacturers as well) have designed an attenuation pad switch into their output stage to accommodate this scenario.  It is a small button usually located next to the output jack on the mixer.  Sometimes the silkscreening will say “Line/Mic” or “+4dB/-50dB”.  If you cannot find a line level XLR input to the house ceiling speakers, the next best thing is to take advantage of this switch and turn the mixer output into the mic level signal that the XLR wallplate jack is expecting to see.

I see very similar situations with many smaller events using powered speakers mounted on tripod stands.  I will see a mixing console output run to a pair of powered speakers, and the XLR cable from the mixer is patched into the mic input on that speaker, instead of the line input, (or the switch on the powered speaker is set to “mic” input).  This creates the exact same hiss and distortion situation as in the paragraph above, except there is a different, and better solution.  In this case, it is best to take the time to re-patch the cable into the line input XLR jack on that speaker (or flip the switch, like on a JBL Eon).  This is the ideal solution because you are now able to keep the voltage level on the XLR cable as high as possible, and any noise that might get picked up in between the mixer and the powered speaker will be “drowned out” by the larger signal voltage available to us in a line level signal.  This way the powered speaker will not have to re-amplify the mic level signal coming from a mixer that is set to output a mic level. 

Most audio mixing consoles are designed to output a line level voltage.  This is because once the signal leaves the mixer, it may have to travel a very long distance before it interfaces with any other equipment, be it an equalizer or a power amplifier, etc.  The longer the run of cable, the greater the chance that it is going to pick up some type of interference along the way (usually Electro Magnetic Interference or Radio Frequency Interference, but that is the start of another topic en.wikipedia.org/wiki/Electromagnetic_interference); so it is best that the actual signal we are interested in is as powerful as we can make it in order to “drown out” any interference that might get picked up along the way.  To reiterate what has already been stated, some mixing consoles have the flexibility to pad the output signal back down to mic level, although you should avoid doing this to keep as much unwanted noise out of the system as possible, unless it is necessary to interface properly with the downstream gear (e.g. house patch).

So far we have discussed mic level signals and line level signals, and the appropriate uses for each, but there is a third operating level that I mentioned briefly at the beginning of this article, and that is “Speaker Level”.  This voltage level is the largest of the three operating levels, and for a couple of reasons will probably require the least explanation.  “Speaker level” is the voltage you get after line level voltage has been run through an audio power amplifier.  As the audio world is evolving, we are seeing fewer stand-alone power amps, and consumers are starting to take for granted that this component will be built into the back of any modern speaker cabinet – both small and large.  One of the nice things resulting from this trend is this: The possibility of connecting a speaker level output to a piece of equipment that is expecting to see a mic or line level input is becoming harder and harder to do, and that is a good thing, because as stated at the beginning of this article you will destroy something by doing this.

Cheesy but effective audio level diagram

I have not really even begun to discuss the actual voltage standards and references associated with each operating level in an audio system, but it is worth also noting at this point, that there are actually two standards of line level audio.  One is the “professional” standard (also referred to as “+4dBu”) and the other is the “consumer” standard (-10dBV).  I will not get into reasons for the specific nomenclature associated with each standard in this article, but the important thing to remember is that a “+4dBu” signal is hotter than a “-10dBV”, even though they are both commonly referred to as line level.  Most CD players, ipods, and laptop computers use –10dBV as their operating level, and most professional audio equipment uses (and outputs) +4dBu as an operating level.  This is mostly important if you may be trying to connect a piece of professional audio gear to a semi-pro or consumer level piece of equipment.  You may end up with a situation similar to that of the line level mixer feeding a mic level input, and end up overloading the destination.  If that is the case, there are several gadgets on the market that you can insert in the signal path to change it to the correct standard of line level.

As you can see, this issue of proper gain staging can become complex in a large audio rig, but a working knowledge of the three operating levels in the world of audio signals is fundamentally important to ensure a good end product; even if all you're doing is connecting one microphone attached to a single speaker.

Bob Conley

Lead Audio Engineer

J&S Audio Visual Houston Branch

Bob is a Berklee College of Music graduate (Boston, Massachusetts), with a degree in Music Production and Engineering.  The majority of his professional experience comes from mixing Broadway tours world wide for about 15 years, with occasional forays into summer music festivals, club venues, and audio for houses of worship of all sizes (which is his passion), during theater down time. Bob started freelancing for J&S in 2006, worked on staff in the Dallas show services division for a short while in 2008, and is the currently working as the lead audio engineer for the J&S Houston division."

Digital Console Innovations

Analog is dead…  That is something you would not have heard coming out of my mouth a year and a half ago, because waaay back then I don’t think it was true.  However, within that time frame a number of improvements to digital consoles have come out that have rendered analog all but obsolete.  Yes, I have analog consoles here at J&S, and for the most part they collect a lot of dust.  Occasionally we get into a bind with all of our digital mixers being gone and they hit the road accompanied by the moans and groans of the mixing engineer on that particular show, and every once in a blue moon (these instances are rapidly getting fewer and fewer) someone will special request an analog desk.  This rapidly dwindling bunch of engineers will tell you “analog just sounds better”!  I agree, it does sound better, and the 3 audio guys in the room for the concert/convention or whatever purpose you may be mixing for can all get together and bask in the warmth of those analog waves, because nobody else in the room will be able to tell the difference.  Here’s the thing Digital consoles not only offer more substance in a much smaller FOH footprint and truck pack, they have become a bridge between the live audio and recording industries.  “So what?” you ask. “After all don’t we do live productions not studio work?”

Imagine this.  You’re getting ready for the opening session of a several hundred thousand dollar show.  There is an opening entertainment number involving singing jugglers and lots of spandex, the CEO will then welcome everyone, a keynote presenter is up next, and a six-person panel discussion follows him.  The room will then break for a half hour for people to get their drink on, and come back to a live party band playing their favorites.  That is a lot of information, and your job is to not only managing the chaos, but also to make everything sound good. 

Enter virtual sound check.  Not only can today’s digital consoles record each channel to a separate track, but with the push of a button or two you can play back the tracks you just recorded, over the PA through the exact channels that were used to make the initial recording.  This means that the 5-minute rehearsal where each of the 6 panel participants said three words before ripping off their mic and running to the next breakout session, was just recorded onto your computer.  Now you can play back each voice separately from one another to your hearts content! You can dial them in on the channel they will using during the show as if they were on stage speaking live, until their mics are EQ’d, compressed, gated, auto mixed (pick one, or pick ‘em all, it’s digital!) perfectly and conveniently.  It’s the same with the band. Have them play a song or two, and as they go back to their day jobs, you can annoy the lighting guy with your endless playback and… more importantly, create a perfect mix.  

Here’s the rub.  Some companies are putting these pieces together better than others.  Surprisingly enough, how well they work does not necessarily have to do with how big your piggy bank is.  Here are some quick talking points on two of the consoles that really stood out from the pack this year.

Avid Venue Profile

Avid makes Pro Tools, which has been an industry standard recording program for years.  They are the ones who pioneered the virtual sound check idea, and thus far, have managed to stay ahead of the curve.  The Venue

series ranges from the large and versatile D-Show to the much more affordable but limited SC48.  Here are several things Avid seems to be doing better then most others right now:

1.  A very intuitive work surface.  Especially for those who started in the analog world (which is still the majority).  The consoles workflow is based on analog principles, and thus, is much easier to wrap your brain around.

2.   Very user-friendly multi track recording interface.  Since Avid owns Pro Tools, they designed the Venue consoles around that recording platform, (which comes free with the console).  Meaning the console and computer used for recording will sync together easily and relatively seamlessly.  Because their recording platform is so solid the virtual sound check feature is second to none.

3.  They have the most comprehensive plug-in package available on the market.  Avid has essentially welcomed an open market for companies to create their own effects, compressors, gates, and whatever else your audiophillic (not a word I know, but it should be) heart may desire, to load on the console and use rampantly.  Many of these plug-ins are designed around very expensive pieces of rack gear at a fraction of the price.  

The Venues have taken the place of the Yamaha PM5D as the preferred digital touring console, and for good reason.  The Profile, which is their middle of the road console, can be customized to accommodate 48 or 96 inputs and outputs, and the internal processing can be scaled from just-get-you-by, to give-you-enough-computing-power-to-run-a-small-country.   If you refer to the picture above, the processing is contained in the rack unit just below the console, which means that the console surface is simply for control (think keyboard and mouse controlling the actual computer contained in a tower enclosure).  That rack space-computing unit is where the mix engines are housed. (“mix engine” is how Avid refers to their processors).  The Profile comes stock with 3 mix engines, which is enough computing power for most mixing applications, but can be fitted with 5, which is just shy of making this thing sprout wings and fly.  The rack on the left is called the stage rack, and is typically stationed in A2 or monitor world.  It contains all of the pre-amps, and is connected to the console via BNC cables, which can reach up to 500 feet before needing a signal boost.  Each Stage Rack contains 48 in/outs and you can put up to 2 Stage Racks on each console, each on their own BNC run, which beats carrying around hundreds of pounds of copper.  There is also a Front of House Rack not pictured here which will allow you to plug in your computer, 360, talkback mic etc. at the FOH mix position.

All of these features come with a price tag of course, which (hope you’re sitting down) ranges from around $35,000 to $50,000 depending on how crazy you want to get.  If you want to dig a little deeper into the Profile and other consoles in the Avid Venue family, you can check them out here www.avid.com/US/categories/Live-Sound-Production

PreSonus StudioLive

Surprisingly enough,  I have found two of the most innovative offerings, are from companies that started exclusively in the recording realm.  The StudioLive is a good example of this.  The first great feature of this console is the price, 24 inputs and 16 outs for $3300.  The flexibility of more expensive mixers is not there, and there are no motorized faders.  All of this is done, however, in order to keep the price manageable.  Here are some of the highlights of this soundboard:

1.  There is complete, real-time, software control of the mixer, since the faders are not motorized you simply push a button to turn them off and mix virtually to your heart’s content.  

2.  There is an iPad app for StudioLive that connects wirelessly to the console via the connected computer, and makes it infinitely more usable.

3. The multi-track recording interface is easy to use with a variety of recording programs.  There is a virtual sound check feature with this console as well, and (I was told by an end user not a sales guy) it works really well.

The layout is easy.  If you take a look at the picture to the left you will see that it plugs in like an analog console, there is no option to re-patch one input to another, or assign aux 3 to aux 7 like you can on many other digital consoles, which can be limiting, but makes set up and programming much quicker.  Another small down side is that the only way for a computer to connect is via one of two firewire 400 ports, which of course means that you cannot connect a wireless router and walk around to tune the room with computer in hand.  The way they give the mixer wireless capabilities is through the iPad app mentioned above, which connects to the console through the computer that is hardwired into the firewire ports, in effect turning your computer into a wireless router.  All things considered the StudioLive sounds good, has a lot of functionality with the built in EQ, Effects and recording capabilities, and for the price that is a winning combination.  To check out the StudioLive in more detail go to: www.presonus.com/products/Mixers

There are more manufacturers who are doing “digital right” at varying price points.  No doubt that the longer the industry is immersed in digital, the better it’s going to sound, and the cheaper it will get.  Simply for the sake of time and space these are 2 console makers that are doing things well in their own markets.  Others to check out though are the Midas Pro Series (1 through 10), the Digico SD Consoles, and Yamaha with their new CL line and nifty Dugan and Dante options.

Until next time,

Live long. Mix well.

Nathan Clark

National Director of Audio

J&S Audio Visual

Show Services

Originally wanting to get into a recording career, Nathan started working in live sound directly out of high school in order to gain a better understanding of audio.  Employed in the field part time while working on a college degree in economics, live audio quickly became his passion.  Deciding this was the career path he wanted to follow Nathan initially hired on full time with Dallas Stage Right, then in 2004 struck out on his own as a freelancer.  Nathan came on board with J&S Audio Visual as the director of the audio department in 2007, and is excited about new technologies on the horizon, and looking forward to tackling the opportunities they present with the rest of the audio team.

LS9 Routing Techniques and Ideas - the ins and omni-outs

Hello again fellow audiophiles, ready for some more fun with the LS9?  I was hoping so.  In this installment we’ll tackle a bit of routing and show some possible solutions to situations that popup while mixing corporate events.  First we need to get our terminology on the same level.  Let’s start with some Digital Mixer vocabulary to get us all up to speed.

Now that we have some groundwork laid down, let’s start putting it to good use.  I’m going to layout some scenarios and give an explanation of possible scenarios that would provide a solution.

DISCLAIMER… I am only making suggestions here, so don’t get your faders in a bunch if I don’t do it EXACTLY the same way you do it, this is theory, not a rule book.  Please send all hate mail correspondence to…Chilloutman@itsnotthatserious.com

Thanks.

REMEMBER TO MAKE SURE THAT THE STEREO MASTER FADER IS DOWN BEFORE YOU RECEIVE SIGNAL FROM ANY SOURCE!!!!!!

Scenario #1

You’re mixing for XYZ Distribution Company at their semi-annual sales seminar series workshop. (Yes, that’s a lot of S’s and I did that on purpose.)  You’ve got 2 presenters on stage with 4 Q&A mics in the audience.  The presenters need to hear the questions but not each other.  You’re also taking pre-recorded questions from a video shot earlier in the day.

Possible solution:

Create a mix-minus that contains the 4 Q&A mics and the sound from the video feed.  Then route that through a Mix on the console, post fader so that any mutes or level changes you make will be reflected in what the presenters hear.

You can accomplish this by bringing up the MASTER FADER LAYER on the console and assigning a MIX to an omni out that is feeding the foldback speakers on stage.

After you’ve done that go back to the input fader level and double tap the button that corresponds to the MIX number you just setup.  (These are just off the left hand side of the display.)  See it flashing, this is called “flip to fader mode” one of my favorite modes on the LS9.  Make sure the flashing button is the same mix that you routed to the desired omni earlier and bring up the faders on the desired channels. 

Voila!! You’ve just created a fold back mix.  You also want to make sure that these are set to POST FADE, you can do this by touching the ENTER button while the mix button is flashing and using the wheel to highlight ALL POST in the top right hand corner of the window, then press enter again. 

Note: The LS9 defaults to pre-fade, so if you are starting your programming from factory default, you will need to set your mix to post-fade.

Here’s a picture to help you navigate this area.

Scenario #2

You’re designing a system for Conglomerates Unlimited in a 2000 seat dome in Minneapolis.  You need to cover an area 200’ deep and 130’ wide.  You have a line array for the mains, front fills across the edge of the stage and delay lines about 140’ back into the room.  You also need to route signal to subs for some rockin’ tuneage they have in a video playback for the CEO.

Possible Solution

I use the matrix in this situation, since you will not need a mix-minus, and it has a “set it and forget it” quality that leaves you with all of your MIX sends available for record feeds, green room playback and such.  I would also setup an independent omni out for each bank of speakers so you can exercise both delay and volume control over each “zone”.

Start by labeling your outputs in the MASTER FADER LAYER for MATRIX 1 through 3 (The sub is going on a separate fader so don’t get ahead of me!) Matrix 1 will be our Mains, 2 our front fills and 3 the delay speakers.  While you’re in there select a patch point, I like to use start with omni 1 for matrix 1 and so on.  Keep a notepad handy and write these down, it will make the next couple of steps a lot easier and faster.  That should take care of the primary speakers.  Now we turn our attention to an oft forgotten ally in the audio onslaught, I’m referring to the subs.  The method I use for feeding subs entails using the MONO fader on the MASTER FADER LAYER.  (It’s just to the left of the red stereo fader, says MONO on the labels underneath it.)  Select the Mono fader and set your patch point like you did for the Matrix section, for instance omni 4.

Now that we’ve got everything wired together it’s time to do some mixing.  From the MASTER FADER LAYER, double tap the MATRIX 1 button to “flip to fader” the sends.  I like to route through the stereo fader since you have an easy visual reference in input fader mode as to what is routed there, and then to your matrix.  So grab the Stereo Fader and set it to unity.  Hit the Select button on the Stereo fader and move that fader to unity.  Repeat this step for MATRIX 2 and 3, this tells the console that you want to send signal FROM the stereo bus TO the Matrix.  While you’re on the MASTER FADER LAYER go ahead and turn up the MONO fader to about -15.  Now navigate back to the INPUT FADER LAYER and let’s do some mixing.

Let’s start at the video playback that I mentioned earlier. Select the channel and you’ll notice two buttons near the pan knob on the screen.  Select both of those, this will tell the channel that it needs to hit the Stereo fader and Mono fader.  Voila!! You should be hearing something now, but it probably doesn’t sound right. Here’s why, you need to set delay times for your mains and delays based on their distances from the front fills.  May need to do this for the subs also.

“Sounds great Joe, but how do I do that?!?!”

I’m glad you asked…

Under the SETUP menu you’ll see SYSTEM SETUP tab.  In that tab you’ll see OUTPUT PORT SETUP and a button for OMNI 1-8. Select OMNI 1-8.  That will display a menu like this…

Select OMNI 1-8 then you’ll see this window…

Set it to feet on the top of the menu and highlight DELAY, then select each knob and turn it to the desired distance.  Remember you want to start your measurements from the box closest to the back stage area.  For instance if your front fill are on, or in line with the lip of the stage that is your “zero” point (your line of speakers the does not require delay).  If your subs are 3 feet in front of them your delay time on the subs will be 3 feet (make sure to measure from the front of your “zero” speaker to the front of the next speaker in line).  Then your will measure from your “zero” line to the next set… perhaps your main stripe of line array… then from “zero” to your delays.  This will give you the same reference point for every zone of speakers you are using.  Remember also when you delay sound you are not just bringing the actual sound into alignment, you are bringing the speakers phase into alignment as well, this is a subject requiring its own set of articles, so suffice to say, if you think 3 feet will not make a difference in sound quality… think again.  That sounds way better now, right? Now that you’re hearing sound from all the boxes you can adjust their levels from the MASTER FADER LAYER.

 If you’re one of the really cool people, you’ve read my article on wireless control of the LS9 and you can walk around the venue and adjust your levels. You haven’t read my other article?!?!?! Ok fine, scroll down when you “get some downtime” and you can read how it's done wether you have a PC or a Mac.

Application time

Now that you’ve seen a practical application of some techniques let me tell you a little about why they work.

Using a Mix-Minus approach to Scenario #1.

It’s called Mix-Minus because it’s the mix minus an element or two.  Since you can’t normally send lavs or podium mics to a stage monitor (Because of dreaded FEEDBACK!!) this is a great technique to give the speaker on stage a way to hear and interact with the audience and make it look natural.  Plus the speaker doesn’t have to worry about straining to hear (or walking into your front fills), they already have a stage wash blinding them don’t give them anything else to stress out about.

Using the Matrix in Scenario #2

Using the Matrix is a fast and easy way to route and time align the Mains and Delays with the Front Fills.  In this instance it makes sense to setup the Matrix this way because it acts like a distribution amp. It’s taking a single source (The STEREO MASTER) and sending it to multiple outputs (MATRIX 1 – 3) while giving us independent level control of each output.  Plus it frees up all of the MIX outputs to be used for other sends that have different mix requirements.  (Fancy way of saying they want to hear different things and not be bothered with what’s happening in the PA.)

Well that’s about all I have to say about that, I hope you feel a little smarter and more confident about using the LS9. If you don’t then read it again!!!  Kidding, this can be some intense stuff if you’re not familiar with digital consoles.  If you have questions or just want to say hi, feel free to email me at joem@jsav.com.  I’d love to hear from you.

Until next time, may your faders run true and compressors refrain from pumping.

Article by Joe McLellan

Audio Technician/Engineer

J&S Audio Visual

Show Services

Originally from Florida Joe moved to Texas 2006 after graduating from Full Sail in Winter Park, FL. He formed an interest in a technical career at the age of 15 after taking up the bass guitar, and came on full time at J&S in 2007. Besides his duties designing, engineering, and running corporate events for J&S Audio Visual Show Services division, in his free time he records for local musicians and podcasters. He is also a music hoarder and frequents local record shops for the latest in obscure bands and music. 

What's In Your Tech Kit?

Your tech kit is your lifeline.  When you’re on show site, running to Guitar Center or Radio Shack is not always an option, so having a well-stocked tech kit can be the difference between failure and success. In this article I will present quite a few items to have in your tech kit, ranging from absolutely-necessary, to this-is-a-really-good-idea.  Glean what you can from this, and even perhaps chime in with your own comment.

When putting together a tech kit I start by looking at the equipment booked on the show, and how many days it will be out.  Using this as a starting point lets me get an accurate count on batteries, DI's, LSP's (Laptop Sound Ports) and cables.  Knowing the gear being used, (PA, console, com, wireless etc.) dictates the extras that go into the tech kit. For example, a show with a digital console needs fewer patch cables than one with an analog console.  A show with powered speakers needs to have power stingers and more iso's then a non-powered system.  Having the right number of adaptors/turnarounds is almost as important as the PA itself.  After all if you can’t get signal into your speakers they turn into really heavy, really expensive paperweights really fast. So lets talk about what should be in the ideal tech kit!

Some of these are obvious.  I am going to list them nonetheless.

Tools

                           

-    A flashlight is a lot better than trying to use your phone to light up the back of a rack.  It’s also a lot cheaper if it gets dropped.

-    Little Lights are a must to light up A2 world back stage, and to be able to hand off a couple for client usage

-    Multiple screwdrivers.  Little ones for tweaking those small hard to get to screws, and normal size ones for getting into racks and cabinets.

-    A multi-meter, Qbox and cable tester are quick ways to help you figure out if something is not working properly, so instead of labeling everything NFG you can give an educated assessment of what is actually wrong.

-    When you find a problematic cable, it can be mission critical to it gets fixed immediately.  A nice assortment of allen wrenches, pliers, crimpers, box knives, wire strippers and a soldering kit are all things that at times you can’t do without.

-    When flying speakers, ratchet straps and tie line can help get your hangs exactly where you want them to stay.

-    Even in this ever-growing digital age sharpies and board tape are still necessary tools to an audio engineer, so do not forget them.

-    (Speaking of tape), during the course of the show you will most likely want some gaff and/or electrical tape as well. They’re like multi tools: cleaning up cable runs, fixing speakers, taping connectors together, fixing set, stage markers (this list could be very long).

-    Clear medical tape is great to have when using lavs or head worn mics that are a little loose and do not want to stay in the desired position.

-    Until everything goes rechargeable, batteries will always have to be restocked. Wireless mic's, DI's and client equipment is what we are concerned with, so have a healthy stocking of AA, AAA and 9volts. Remember to take into account rehearsal and show days, and multiple shows in one day.  Keeping a stash of used batteries in your kit is a good idea (ones that are not drained, but you don’t feel comfortable using them for show either).  I find them useful for mic tweaking and rehearsals, as well as the many miscellaneous items used during the course of a show.

Cables, Adaptors and Connectors

-    Turnarounds are problem solvers and time savers! (Ever have a stagehand run a 100' cable up and thru some truss just to find out he did it the wrong way?)

-    XLR, 1/4", NL4 and RCA are an absolute necessity to all audio tech kits. Without getting into numbers XLR turnarounds are the most important, typically a couple handfuls of both genders will be adequate, followed by a handful of 1/4", NL4 and RCA turnarounds.

-    The most common adaptors are: XLR-1/4", XLR-RCA, XLR-mini, XLR-Y's, RCA-1/4", RCA-Y's, 1/4"-Y's, ISO's and pads.  (When it comes to adaptors, you will need to know how your gear is terminated both in and out.)

-    With digital consoles taking over the world, patch cables are used less often.  With that said, they can still be very important!  A variety of signal cables are necessary for patching in bands, DJs, playback sources, audio/video recorders, and any external effects/processors you might desire.  Here are the ones we most commonly use: XLR-1/4" (TRS & TS), RCA stereo, 1/4" (mono and TRS), RCA-3.5mm, RCA-1/4" and stereo XLR pairs.

-    When a cable fails it is usually at the connector, so it’s good have a variety of spare connectors in your tech kit.  The Hubble power connectors that we typically stock in our tech kits are for our 10/5, 12/3, 208 cables.  Depending on how well your signal cables are stocked you may want to have extra XLR and 1/4" connectors as well.

Transducers

-    Direct boxes are used for level matching, to minimize noise, distortion, and grounding issues. There are three types of direct boxes that we concern ourselves about, and you will find them all in our tech kits,  Active, Passive, and PC.  They all pretty much perform the same function, they just go about it in different  fashions.  

-    Active DI's require phantom power; they usually have better attenuation pads, and are great for noise filtering. 

-    Passive DI's simply convert from line to mic signal without all the fancy filtration.

-    PCDI's can be active or passive but have more input options (e.g. RCA, ¼”, 3.5mm). Keeping several on hand is always useful.

-    Laptop Sound Ports are a very nice tool to have, because they convert 3.5mm to XLR, have a ground lift and attenuation knob which allows for easy volume control of your PC/iPod or anything with mini jack. Their limitation is that you will only be able to get a mono signal from the source that you are plugging them into.

-    A nice analog to digital converter is a good addition to any tech kit for those impromptu recording sessions.  Whether you’re trying to capture voice-overs, or a stereo program record, something with a couple pre-amps can be clutch.

-    A push to mute stomp box is a handy device to give to a sick yet savvy presenter, who can use it to kill the mic with his foot while coughing and/or sneezing.

-    Sanitizer.  When that sick presenter is done and your mic is throughly infected, you can quickly and easily make it safe for other presenters to use.

-    Have an assortment of microphones as back ups just-in-case the client decides they want more audience mics, or throw a few more people onto the panel discussion etc.  Having extra dynamic and condenser mic's in your tech kit can be a lifesaver.

-    Microphone wind screens are a nice tool to have for those blustery outdoor shows, and may be helpful for the voice over talent who is an extreme P-Popper.

        -  Throw in one or two VOG (voice of god) mic's with a quiet on/off switch (we stock the Sennheiser e815s with a magnetic switch), or an inline push to talk box, and the transducer section in your tech kit will be nicely stocked.

Recap

Tech kit necessities:

-  Batteries

-  VOG mic's

-  Push to talk for VOG mic.

-  SM-58's

-  SM-81, or RTA mic of preference

-  Microphone wind screens

-  Sanitizer

-  Volt meter

-  Cable tester

-  Q-box

-  Ratchet straps

-  Tie line

-  Tools

        - Wire Cutters

        - Crimpers

        - Box Cutter

        - Screwdrivers

        - Strippers

        - Soldering kit

-  Tape (board, electrical, gaff and clear medical)

-  Sharpie’s

-  Turnarounds - XLR, 1/4", NL4)

-  Adaptors XLR - 1/4", XLR-RCA, RCA-1/4", XLR Y's (both genders), RCA Y's, ISO's, pads)

-  Patch cables - XLR-1/4", TRS, RCA stereo, 1/4" signal, stereo XLR pair.

-  Hubble connectors - 10/5 (HBL 2913 Female, 2511 Male), 12/3 (HBL 2313 Female, 2311 Male), 208 (HBL 2623 Female, 2621 Male), and standard Edison, both male and female.

-  DI's (active and passive)

-  2 Channel USB A to D Converter

-  Push to mute box

-  Laptop Sound Ports

-  Table top mic stands

Tech kits come in all shapes and sizes, it is up to you how extravagant you want to get.  I have covered my “must haves”, but everybody’s needs/wants are a little different.  So, what's in your tech kit?  Feel free to add a comment onto this posting with anything you find indispensable in your tech kit.

Article By:

Brett Speer

Audio Technician

J&S Audio Visual

Show Services

Started down the audio path going to work for Crossroads Audio, a Dallas based sound company in 1995. He spent 9 years with them learning system design, troubleshooting, engineering, installation, repairs and mixing. While at Crossroads, Brett moonlighted with most of the area nightclubs, offering solutions to their audio problems.  In 2006 Brett went to work for Dallas Stage Right gaining experience in the lighting, video, staging, installation and warehouse management fields.  Brett joined the J&S Audio Visual Show Services audio team in 2010 as the gear coordinator where he currently puts his broad skill-set to good use.