After starting with the main speakers, I work outward from there until we get to the speakers in the back of the room.  Often the first step after mains is downfills.  I like to work my way out from the mains because as I add more speakers to what I’m listening to, they all start to work together and I have to follow that energy rather than fight it.  Every change you make to one set effects everything else so you have to think holistically and experiment.

The key to making fill speakers work is two things – delay time and EQ.  If the main speakers are our starting point and benchmark that everything else must align to, fill speakers such as downfill or delays will normally arrive at either slightly different or perhaps greatly different times to your ear depending on where you are standing in the room.  The trick is to use delay so that speakers that are separated by 5 feet or 75 feet sound like they are all arriving at your ear at the same time and thus working together.  I use my software program to help calculate the necessary delay times, often times with a bit of trial and error in real world to nudge things forward or backward just a bit so it feels right.

After the delay time is right, I like to first turn off the speaker I’m getting ready to work with and just measure how everything else we’ve already optimized sounds in the location of the fill speaker.  This tells us what is missing sonically in this location from what we’ve already done and that is what we will focus on with EQ’ing the fill speaker.  I will high-pass the speaker so the low frequencies that are already hitting the area from the main speakers are not competing with additional stuff trying to come from the fill.  Now balance the sonic EQ of the fill so the response curve sounds as close to standing in front of the main speaker as possible.  Continue this process with each set of fills and eventually all parts of the system are operating at the same time.

As I said, this is a very high level discussion because the reality is that the process of tuning a PA is as much art as it is science.  For me, depending on the complexity of the system and the acoustics of the room it is installed in, there can also be a fair amount of experimentation & trial/error.  One of our largest rooms at newlifechurch.tv has a distributed audio system design – 4 main arrays in the front of the room with downfills below them and then a ring of delay speakers 2/3′s of the way back.  It took me three attempts at tuning this room before I arrived at a product that I’m reasonably happy with as a long-term starting point.  Each time I did it, I learned more about how all of the speakers interacted with each other, the acoustics of the room, and the result was less and less needed EQ each time.

In my experience with trying to get a great board mix for broadcast from the same console mixing FOH, a well-tuned PA is absolutely required to have any chance of success.

I have a basic process that I follow when tuning a PA.  Everyone probably approaches something like this differently so your mileage may vary, but here’s a sampling of my thoughts.

First and foundational for me is to not take anything for granted pre-existing in the system from past engineers, the installation company, or “helpful” volunteers. Crossover points in bi or tri-amped speakers, the manufacturer recommended EQ points on a box, amplifier processing either bypassed or enabled, every speaker functioning, balanced levels between boxes, and on and on.  I like to begin by ensuring that every box is actually functioning, the array is balanced so that the level remains consistent as you walk the room, and that all DSP in the system is either flattened, bypassed, or disabled.

There is nothing worse then spending hours working on a set of speakers to then find that a circuit was engaged somewhere, thus coloring what you’re doing, and you didn’t know about it.  In one of the newlifechurch.tv rooms, just this step alone brought huge improvements to the system because I found that one of the three boxes in each of 4 arrays was operating at 50% of the volume of the other two boxes due to amplifier trim.  The result was unbalanced coverage front to back that had likely existed for a long long time.  Don’t take anything for granted!

Next, I’ll always begin a tuning process with the main speakers.  In some rigs, like one of our main rooms, this is all I need to deal with because there are no delays or fills to add into the mix.  The main speakers will always carry the biggest load of the work in a system and put the most energy into the room.  Because of this, I like to start here and then fit the other boxes around the mains.  I’ll talk about the software I use in another post.  For now we’re keeping it to a 10,000 ft level and just talking process.

The ear doesn’t hear things “flat”, especially as the volume level increases to concert levels.  As such, I’m not looking to create a flat PA.  Some guys named Fletcher & Munson did lots of research years ago on this hearing phenomenon, resulting in the Fletcher – Munson curves.  (Google it if this is completely new to you – fascinating stuff.)  In working with the main speakers, I’m looking to smooth them out sonically and ideally emulate a Fletcher-Munson curve (smooth cut in the PA that starts around 1k and has its deepest point at 4 or 5k before returning to normal by 10k).

I go back and forth between measuring a 2 second sine sweep in order to graph frequency response and listening to a playlist of room tuning songs from my iPod that I’ve been using for this purpose for years and KNOW how they should sound.  Often times you can over-tune a set of speakers by going crazy with every little dip and peak on a frequency response curve, but the real test is how it actually sounds with music.  The magic is in a healthy balance between the two – science and art.  In the end, the ears always win.

Tomorrow we’ll cover delay speakers…

In my mind there are three places in a system where equalizers normally reside.  CHANNEL EQ –> SYSTEM EQ –> SYSTEM PROCESSOR. We’re going to talk about them out of this order in order to get my point across so here we go…

First is the channel EQ that exists on the mixing console.  I view the responsibility of this part of the chain to provide correction or artistic shaping of the interaction between a microphone and an instrument or vocal.  If a mic is too woofy on a vocal or a guitar amp is too harsh (and you’ve already tried to improve the situation at the source by working with the artist), the channel EQ is the place to make the necessary adjustments.  Carrying this out, once you’ve made all of the corrections to the various inputs, you should then be able to listen to the results of this mix through headphones or record the left/right mix and play it back on another speaker system and have a good representation of what you intended the mix to sound like.  It shouldn’t be too bright or too dull because you’ve fixed all of those individual combinations on the channel strips.

The second equalizer/processing location in a large sound system is the system processor which addresses the PA speakers themselves – the crossover points of a bi or tri-amped system, signal delay necessary to get all of the speakers arriving in a common time domain, and equalization that addresses the natural interactions between all of the speakers in an overall system.  Normally this degree of processing occurs inside a system processor – a Dolby Lake, BSS London, proprietary amplifier, or in my case, Shure P4800′s.  It is located in the chain directly before the signal coming from the sound mixer hits the amplifiers that drive the various speakers.

The final equalizer/processing location is the overall system equalizer.  This point in the chain falls between the output of the mixer and the input to the system processor/amplifier/speakers.  Sometimes this processing point might be combined into the system processor, in the old days it used to be simply a trusty 31 band stereo graphic equalizer.  The purpose of this equalization step is to optimize and correct the speaker system as a whole to the acoustics of the room where it is located.  If the room is boomy or harsh, live or dead, etc. the system EQ allows the engineer to craft the overall sound of the speakers so that what comes out of the mixer sounds transparent (meaning what came out of the mixer is what comes out of the speakers) in the room.

I outlined the three parts in the order I did because I kind of look at them in that order of mental processing as well.  The channel EQ corrects for individual inputs and builds the mix itself, the system processor corrects for all of the individual speakers and amplifiers and builds them into a cohesive system, and then the system EQ is the glue that brings those first two pieces together into what is heard in the room.  If any of those three places in the chain is not properly optimized, the overall product will suffer.

The primary goal I’m trying to achieve as I think about these three parts of the sound system is transparency.  I want what sound came out of the console to come out of the speakers with the same sonic characteristics (transparency).  The simplest way to test this is to play a known recorded track (I have a playlist of songs that I use to test how a PA sounds) through the speakers and see if what comes out sounds like how I know that track should sound because I’ve listened to it thousands of times through hundreds of different PA’s – the balance of low end to top end, the harshness or lack thereoff, the warmth of the system, etc.  Why does this matter?  Because if the PA is not transparent, if what comes in is NOT what goes out on the other end, I’m going to have to make corrections somewhere else in the chain (normally at the same places on every channel’s EQ) to make the mix work and this will likely compromise that board mix.  It also makes it far more difficult for a more novice engineer to achieve a great mix unless they are more comfortable with channel EQs or for me to get something great happening quickly because I’m going to have to apply more corrective EQ which will take more time to dial in.

In my experience, the biggest weakness I find in installed PA systems is in the optimization of the system EQ.  More often than not (and exactly what I’ve dealt with at NLC.tv), the system EQ does not make the PA system transparent.  Playing an iPod track through the console without any channel EQ will sound far different coming out of the speakers than it would listening through good headphones on the console or a different speaker system.  The core sonic characteristics of the song do not translate through the room.  As a result, the engineer needs to apply channel EQ to correct for these room problems in order to start mixing with a blank canvas.  In our iPod example this can be easily done because we’re only talking about a single stereo input.  However, explode this out to a 6 or 7 piece band, 5-8 vocals, and 30 or more channels and the task can be far more daunting.  The byproduct is that the L/R board mix, which often feeds video recording or internet feeds, suffers greatly.  All of the board mix changes that are made to make inputs work in the poorly tuned room are more drastic than what was really needed so the L/R mix isolated by itself is nothing to be excited about.

Next time we’ll start getting specific about a tuning philosophy to correct these problems.

First, it is extremely important that you be able to monitor your mix through headphones on the console or record the L/R feed, play it back, and it sound good.  This is a biggie.  If you’re working with a big room (more than 750 seats for this example) and sonically the mix you’re listening to just ain’t happening, you most likely need to revisit how the PA is tuned, something in the speakers themselves, etc.  I have a series of posts coming in the next week or two about my philosophy when it comes to system equalization & PA/room tuning so we’ll dive into all of that later.  For now, I’m assuming the mix you’re hearing in headphones sounds good, but its just dead – sounds like it was recorded in a studio.

We’re going to add two pairs of mics to the room.  The first pair is a set of shotguns that will be placed somewhere along the front corners of the stage, out of the way, aimed out perpendicular to the front of the stage and in such a way they can throw out into the room without picking up too much of the first couple rows.  I have ours on tiny floor bases that make them just poke up over the front lip of our stage, pointed up at probably a 15 degree angle so they aim over the heads of those first couple rows.  These mics will be the primary pickup point for the audience themselves.  Pan them to 9 o’clock and 3 o’clock.  The exact make and model of these mics is not that important to me.  At Kensington and now at newlifechurch.tv I have used Audio Technica 8035B’s from Sweetwater, the least expensive name brand shotgun I can get.  I might feel different about the level of quality necessary if I ever tried better mics, but I’ve always had to do this project on a budget so bang for the buck rules here.  I’m most interested in the pickup pattern rather than necessarily the sonic character.

The second pair of mics needs to be condensers hanging about half way back in the room.  These can be high and also out of the way.  I have mine about a foot below the lighting grid so they are very high.  At Kensington I used some Audix small diaphragm units that were sitting on a shelf.  At newlifechurch.tv, I’m using some existing Shure MX choir microphones that were already in the air.  The purpose of these mics is to add space to the recording, so it will feel like its happening in a big room (which it is).  These get panned in the console to hard left and hard right.

Now in the console we do two things.  First, we split up all of our inputs to one of two main busses – left/right and mono.  You must have a console that has left/right/mono discrete busses for this to work.  M7CL’s do.  Everything music related (band, vocals, playback, effects) routes to the left/right and we call this one music.  Everything speech related (pastor mic, MC’s, and spare pastor mic) go to the mono and we call it speech.  Now we need a few matrixes – a pair for the PA and a pair for what I call WORLD.  In the PA matrix, route music and speech at the same level.  But to WORLD, add about 6-8 dB to the speech side.  This will effectively balance out the perceived difference between music and speech on a recording.

Step 2 in the console is really easy with an M7CL – add the two pairs of audience mics in to the WORLD matrix we just built.  The ratio between the shotguns and hanging mics I’ve found sounds best in our rooms is almost 2:1 shotgun to hanging.  The hanging mics will wash things out really quickly so the trick is to get just enough to add the depth and dimension without putting in so much that it totally collapses the mix.  EQ both pairs of mics by adding a high-pass centered at least at 250 hz, pull out some 400hz & 2k, and I like to add a bit of sizzle to the hanging mics.  The last thing to do is turn on the buss compressor over the WORLD matrix, 10:1, medium attack & release, and set the threshold so the band pulls 2-3 dB off the top of the mix when things hit hard.

The finished result sounds like this…

DISCLAIMER: This is the first audio sample I’ve posted from NLC. It’s getting there but still a long way to go! Figured I’d give you samples in process rather then just waiting until it is all perfect. Ok, resume…

On our M7CL’s, I’ve invoked user levels in order to lock out access to the room mic channels and the WORLD matrix so that everything happening behind the scenes to build this mix will be protected and can be relied upon.

I’d love to hear your modifications on this system and, if you try it, your experiences.

Last time I wrote about the improvements to our IEM system that have come from implementing a new transmitter/receiver combo and some additional ear piece options.  The problem with adding more wireless into an already congested environment is a higher probability of failure.  I’ve personally managed to avoid Murphy’s Law when it comes to IEM for a long time but I realize it was only a matter of time before someone’s system failed during a service and we’d be in big trouble.

Enter Sidefills.  A friend spent a couple services with me post-Christmas and suggested that what we were missing was a good holistic stereo mix that will fill in the missing pieces when someone pulls an ear out and also provide some added energy to the first couple rows that might be a tad light from the PA over their heads.  It seemed like a good idea.

In my past life as a monitor engineer, sidefills were an important part of getting a great onstage sound since I worked with lots of vocal groups who would often times only wear 1 ear anyway.  For some reason, I never even considered it at KCC since our old PA already put energy everywhere except where I really wanted it – the prospect of adding an additional full-range sound source was unappealing.  However, in the new reality of a controlled PA without a ton of stage spill, it seemed possible this might just work.

Work it does.  I have a set of EAW KF300s that have been unused for a while.  I put them on top of the subs, about 5 feet off the ground, pointed in towards center stage.  Stereo mix gives more clarity & separation than mono, add some EQ to smooth out the rough edges of the boxes themselves, and add a few milliseconds of delay to the PA so the clusters in the air are in relative time alignment with the sidefills and the result is really good.  I’m feeding these boxes from FOH subgroups so they essentially get a rebalanced FOH post fader mix.  The mix is split up to rhythm, band, and vocals.  Using this approach keeps them specifically music-focused – playback and speech mics stay out of them.

We still keep a set of wedges on the front row for lead vocal monitors.  The addition of the sidefills allows the vocal to literally be surrounded by themselves – just a touch of vocal in those front wedges pulls the singer’s image forward.  Needing less band in those front wedges reduces mud heard at FOH since the sidefills are actually working with and as a part of the PA rather than against it.  I find myself putting 100% vocal in them, 80 or 90% band, and 60% rhythm (since the acoustic kit on stage adds its own ambient sound).

If you mix on a primarily IEM stage as I am and haven’t revisited sidefills in a while, this experience enthusiastically suggests they’re worth a try.

First, a few weeks before Christmas we were finally able to upgrade from wired IEMs driven from a headphone amp to Sennheiser EW300G3‘s.  The difference has been awesome in a few ways.  First, the sonic quality of the 300s is nothing short of amazing.  The low end is tight, the top end natural.  When paired up with a good set of ear pieces, I’ve been blown away by the mixes some of our guys are dialing in.  Second, not being tied to the IEM wired cable has made a big difference in the comfort level of our artists.  Often times we had to be very careful in how the service was programmed to make sure artists had ample time to get on stage, plug in their ears, put on guitars, tune, etc.  If something went wrong plugging in their ears or they accidentally pulled the cable out, we were in trouble.  Along with using more wireless instruments (I’ll write about this soon), the result has been a much smoother & more polished presentation.

We purchased 4 transmitters & 6 receiver beltpacks.  For the majority of things we do, we will use the 4 transmitters/beltpacks, along with our existing PSM600 and 2 wired mixes for keys & drums.  With some extra beltpacks for the 300s and our PSM600, we can easily expand by adding a few more mono mixes the few times a year they are needed.  It is so hard to justify extra transmitters in a rack if they aren’t going to be used very often so this seems like a reasonable compromise.

A big surprise is how much some of our female artists enjoy the bundled Sennheiser ear bud – the IE4.  The sound quality of the buds is probably on par with a Shure SCL3 or good set of iPod buds.  What is appealing about them, though, is how well they seem to fit smaller ears.  Our vocal director, who’s been wearing our generic Shure pieces for a year and a half or so, felt like these were the most comfortable thing and for the first time she can actually forget she’s wearing them.  Definitely a good problem solver.

Another important piece of the puzzle is the addition of some Westone UM2′s to our inventory.  After someone recommended trying them out, I’ve finally found something to make me give up my Future Sonics that I’ve had forever.  These are the most comfortable universal fit I’ve ever worn and they are the first universal bud that I’ve heard with really natural midrange, which is really important for vocals & guitars.  The Shures, Future Sonics, and Ultimate Ears I’ve heard all can sound really nice in the low and high ranges, but often times feel a little scooped, honky, or just otherwise unnatural in the midrange.  The Westones seem to bring a lot of that presence back.  Best of all, the Shure foamie’s I can purchase in bulk fit on these as well so its a win on all fronts.  Adding Westone’s into the mix have brought 2 of our hold-out musicians into the IEM fold.  Finally they feel like the guitar sound they hear in the ears feels like what it should actually sound like.

I’ve been really disappointed with the build quality of the Shure SCL3′s and 4′s so time will tell if the Westone’s can withstand heavy use.

So from a music team standpoint, we now have 100% of our band members transitioned to IEM.  Next time I’m going to outline another addition to our monitoring system that feels like its filled in the missing link in the chain.

For some reason I didn’t turn on the PA when I first arrived to prep for Saturday afternoon rehearsal.  Normally I set up iTunes with some music while we work, but this day silence seemed like a better idea.  Once the stage was set, I turned on the PA but we still didn’t put music through.  Sitting at FOH programming the console for the day, I heard a strange pop come from the stage.  It seemed like it might be from the PA but everything was still muted and I’d never heard a sound like this before.  A few minutes later, it happened again but this time I was looking at the signal meters of the desk and didn’t see a blip.  This time I became concerned that there might be a problem downstream of the desk and we went into troubleshooting mode.

Some iTunes music through the system revealed that the mid and low channels of the right side our tri-amp’d KF650s were not functioning.  A visual inspection of the Macrotech 5002 amps showed that both the mid and low channel signal lights were remaining on, in IOC mode, but obviously not passing signal.  After power cycling a time or two, we realized each amp would pass signal for 5 seconds or so and then set into the IOC mode.

At this point, we were less than 45 minutes away from the band arriving and trying to figure out how to salvage the weekend.  Thankfully, God provided exactly what we needed.  Kensington has an old Turbosound rig that we call our “Yellow Truck” system because it used to be the primary portable rig for off-site ministry events.  Thankfully, the amp rack for that rig, which just so happens to include 2 Macrotech 3600 amps, was on site and 30 minutes later was patched in place of the 5002s having trouble.  Catastrophe averted.

I’m not sure what lessons there are to learn from this.  It is a total fluke that I didn’t run music earlier in the day as I normally would.  If I had, we would have had twice as much time to come up with a solution.  It seems the biggest reminder is to make sure backup plans are in place for as many components of your system as possible.  What would you do if drivers were to blow during rehearsal/service, amplifiers don’t function, wireless microphones fail, etc?  Anything can happen and the most important thing is to be prepared to adjust and keep the “show’ going.

I guess our good ol’ KF650s didn’t want to go out without a bang!

On our old KF650 rig, DSP was extremely important.  We used a BSS London BLU-80 to provide an insane amount of processing to the entire rig, essentially trying to polish it to a point of usability.

When we decided on going with d&b, a significant bonus to the system is the awesome amplifiers with built-in DSP for system processing.  While some manufacturers lock down almost all system settings making it too difficult to customize the rig to the room, d&b has done a great job of creating a system with what I believe to be just the right amount of control to tune the room.  Because of this, my goal was to have as little DSP as necessary to make the rig sound good.

Our signal chain now is a left/right feed from the console that hits the first system amplifier, then simply loops through to each of the other amps.  The system is run full range from the console rather than aux fed subs so that the phase and spectral response of the PA as a whole will remain consistent from service to service and engineer to engineer.

Each amplifier channel has a 4 band fully parametric EQ & signal delay.  The coolest part of the DSP, though, is a few other settings available that provide subtle ways to sculpt the rig artistically for different responses depending on what’s desired.

• The CPL (Coupling) circuit compensates for coupling effects between the cabinets. These effects increase as the length of the line array is extended. CPL begins gradually at 1 kHz, with the maximum attenuation below 400 Hz, providing a balanced frequency response when T10 cabinets are used in arrays of four or more.  The function of the CPL circuit in these amplifiers can be set in dB attenuation values between –9 and 0, or a positive CPL value which creates an adjustable low frequency boost around 65 Hz (0 to +5 dB).
• The HFC (High Frequency Compensation) circuit compensates for loss of high frequency energy due to absorption in air when loudspeakers are used to cover far field listening positions. It’s almost like an “air” control that gives a few options for how open the high end responds.

The D6 and D12 amplifiers have three configurations for T10 cabinets, Line, Arc and PS (point source). The Line or Arc configurations are selected when the T10 loudspeaker is used as a line array. The chosen configuration will depend on the curvature of the array. The Line configuration is selected when groups of four or more T10 cabinets are coupled in a straight long throw array section, where the splay angles to adjacent cabinets are 0° to 2°.  The Arc configuration is selected when T10 cabinets are used in curved array sections, where the splay angles to adjacent cabinets are 3° or more. Within a typical array both amplifier configurations are used. The PS configuration is selected when the T10 is used as a single spherical loudspeaker.

Wrapping up this section on DSP and amplifier control, I’m very impressed by the R1 software included with the amplifiers to provide remote control from FOH of the entire rig.  Tuning functions such as EQ, delay, HFC, and CPL are all included, but the coolest function to me is a page we set up that shows the input level into the amps, the output level the amps are actually sending so you can visually see your system headroom, actual ohm loads on the amps, actual RMS output wattage, power supply voltage, gain reduction, and temperature warnings.  The software is completely customizable so you can set up pages to show exactly what you want, how you would best like the information.  Great stuff!

I must say, these subs have always been some of my favorite.  They are very very tight and punchy, with what feels like headroom for days.  Some subwoofers start to break up when pushed and “feel” like they’re being pushed.  These B2s can run right up to the brink and actually sound better as they’re doing it.  The result is a rig that feels like it has no limits!  Pretty awesome to mix on, I must say!  :)

From the ArrayCalc software, here is the predicted pattern of our initial configuration – 2 subs equally spaced on the stage deck approximately 40 feet apart.

This looks a little nastier on paper then it actually feels in context, but there is definitely some slight banding of the low end in this configuration.  Another option that I think I want to try is 3 subs in a cardiod configuration flown over the center downstage position where our old center cluster used to live.  Here’s its predicted response…

As you can see, there’s a big difference in how these two configurations will perform in the room.  The jury is still out on which direction we will go.  I’m hoping to try this soon with the boxes on the stage deck before we commit to the air and see how it translates compared to the equally spaced approach.

As you can see, our main arrays are each comprised of 6 T10 main speakers and 2 TSub subwoofer cabinets.  The T10 is made up of two 6.5″ drivers and a 1.4″ hi-frequency compression driver.  The TSub is a single 15″ high excursion driver.

This first output from ArrayCalc simulates the aiming, coverage, and SPL throughout the main floor of our room.  In the Top View, the white line shows the center throw point of each array.  The dotted yellow line shows the coverage pattern at the top of the array while the orange line shows coverage at the bottom of the array given the rake of the boxes.  Listening to the actual space with the boxes installed, as expected, the only shaded listening positions is the 5 or 7 seats in the extreme front corners of the room and the very last row under a slight balcony overhang.

The next drawing is Main L/R Profile at 10 degree aiming.  This shows how the center point of each box’s coverage and how that will translate to the seating area.

Finally, the Direct sound level vs. distance shows how the rig should respond in the space and what SPL fall off to expect from front to back.  As you can see, this prediction was probably my only slight concern since there’s as much as a 12 dB difference from the front to back of the room.  We expected to need to do some gain shading of the boxes in the array from bottom to top in order to smooth out this difference a bit.  However, in actuality, we found the difference to be far more understated then expected.  As I noted with my general observations in Friday’s post, everyone was blown away by how consistent the spectral balance and perceived SPL remains throughout the entire seating space.  It really does translate smoothly from FOH to both the front and back of the room.

Next time we’ll talk about subwoofer arrangement and performance.

For the background on how we’ve gotten to this point in the upgrade, check out this post I wrote a few weeks ago if you missed it.  This week was the big week to install the new rig!

All of the system components arrived from d&b by early last week so on Monday, the crew from Advanced Lighting & Sound began the install.  The process went very smooth and by the end of the day, both arrays were in the air and cabled, plus all of the old amps had been removed and the new ones mounted.  Tuesday morning began with connecting the new arrays into the existing lines from the grid to the amp/dimmer room and cabling and dressing the new amp racks.  By lunch time, we were ready to start tuning the rig in preparation for the first rehearsal on the new system to begin at 6:30p.

Awesome surprise #1 was what Easera Systune software showed when we turned on pink noise through the T10 arrays for the first time.  With the measurement mic placed about 1/2 way back from the rig, the response curve was literally FLAT.  Our team was floored.  A first pass of some tracks from ProTools sounded incredible.  We’re using three EQ filters on the entire rig just to provide some artistic sculpting.

Pleasant surprise #2 was the consistent coverage throughout the entire room from a single pair of T10/T-Sub arrays.  It was jaw dropping how smooth the 105 degree pattern of these boxes covered exactly where the array calculation software said it would.  At the edges of the pattern, the sound smoothly begins to roll off.  This creates some great “shaded” seats where those who prefer things just a bit quieter can be happy.  The spectral response in these seats is still very pleasing and even, it’s just a tad softer.  For the first time ever in this room, the listening experience at FOH is nearly identical to that 15 rows in front and 5 rows behind.

Surprise #3 was how efficient the rig is to get up and go with SPL.  Our team decided the T-series rig would be a solid solution for our room based on our tests with some of our campus’ rigs (L’Acoustics Kiva/Kilo and ARC’s).  However, if I had one nervous spot, it was based on whether the new system could keep up with our 650 rig when it needs to fill the room and get in your face.  While we don’t do it very often (95 dbA and up), I knew we might miss that extra headroom we’ve had with the current system when the room is full and energy is high.  Believe it or not, this d&b system is the little rig that can.  Early tests show we have no trouble ramping up into the high 90′s without getting beyond the first signal light on the amps.  Things start to break apart a bit once you start hitting around 100-102, but this is fine for us because like I said, our cap is really 95-97 peak.

I’m going to write more in the next week about the system configuration we went with and learnings along the way.  For now, here’s a few crude photos shot before I left Wednesday night for a few days off after a very long week.

With this lens in mind, I’m still curious for feedback on other suggestions we might be missing or should check out.  Thanks in advance.

Coming soon to Kensington Troy:  a new PA!  For the background on what’s been going on the past few weeks, make sure you check out this post from 2 weeks ago if you missed it.

I was recently involved in a process to actively prepare for a new PA that lasted almost a year.  However, due to the complete tanking of the economy in Detroit after the church finished pledging for our Trans4mation campaign, the project was put on hold.  A few weeks ago, I brought in a few different, smaller PAs then what we have currently or were planning to purchase, just to see how they would translate in our room.  Along the way, I think our whole team realized that a bit smaller rig (the leading candidate previously was d&b Q-series) turned out to have plenty of horsepower for what we do artistically and where our congregation is at regarding desired SPL.  Both the Kiva/Kilo & ARC’s rigs sounded fabulous in our space and had plenty of get up and go to make everyone satisfied.  It just so happens that there are some really great system options that fit in this category that also save us some pretty significant money which can then be invested into lighting and video, giving us an overall better product as well.  It really is a win/win for everyone.

So, we’re approved to proceed…evaluate, discern, and install!

The goals from a new rig are:

1. Even coverage with as few of boxes as possible so as to avoid all of the phase and comb filtering issues we have currently

2. Crisp, pleasing high end while maintaining rich, buttery, pleasing mids and thick, tight, feel-in-your-chest type lows.  The Kiva did exceptional with the highs and lows but the mids were lacking that pleasing quality.  The ARCs had wonderful mids and lows but were a bit smeary on the high end…especially with imaging and directionality.

3. Through our experimenting with these other rigs, we learned that a system that can get us into the mid-high 90′s without giving up is plenty for our weekend and midweek services.  Since we EXTREMELY rarely, if ever, bring in outside artists where we need to be concerned with rider acceptance or true concert levels, that goal isn’t the most critical for this system.

4. It is also a really good thing for the Troy main campus to have a similar level rig to what our portable campuses own since we’re all working in pretty similar size rooms.  If money were no object, it would be awesome to have a concert level J series or VDOSC in the space, but the reality is those boxes are TOO big for the room, throw too far, and are not the right choice.

D&B released a new PA a few months back that is an interesting solution for our room called the T series.  The system is comprised of T10 main cabinets, the T-Sub which is a single 15″ box that extends the response of the flown array down to 47 hz, and B2 subs for the ultra low-end.  The T10′s biggest claim to fame is that it is extremely flexible, used both as a line array and as a high directivity point source loudspeaker.  The HF driver is fitted to a unique waveguide horn producing horizontal dispersion to 105°, but by rotating the horn by 90°, the T10 is  transformed into a vertically oriented standalone full range loudspeaker with 90° horizontal and 35° vertical dispersion.

In our application this is incredibly unique and helpful because our rig will be comprised of two 6 or 7 box arrays at the front of the room covering 90% of the main floor, but we also need some front fill and balcony delays.  Due to the flexibility of the T10, we can use the same box for every application in the room, ensuring consistent voicing and performance throughout the entire space.  The dispersion of the T’s is also a better fit than the Qs (105 degree vs 75 degree) which will mean less fills to make up for dead spots in the dispersion of the main arrays.

I’ll write more about this as we start down the path of evaluating this rig in a few weeks. I’ve always been a really big d&b fan and this system carries on the same legacy of the larger J & Q series.  Can’t wait!

(For those of you who are more familiar with Kensington’s rigs, this is the same system we are planning for the Lake Orion campus which will be going live this fall.)

If you’ve been following me for a while, you know the current PA rig at the main campus is tired at best.  We went through an exhaustive design process last year to plan for an expansion and upgrade to the campus but due to the current economic realities of our area, those dreams have been on hold indefinitely since late last year.  On top of this, we’ve been especially struggling lately with questions of volume and volume quality as it relates to everything we do in the room.  This test provided a unique opportunity to evaluate how a more ideal speaker solution would effect the audio experience.

The test PA in question was a L’Acoustics Kiva/Kilo/SB118 rig.  6 Kivas & 2 Kilos per side, 2 SB118s on the ground per side.

I hung the PA on Sunday afternoon as a part of the stage turn from the weekend to midweek.  On Tuesday morning we connected everything up and sent signal through for the first time.  One important part of this test was to keep our current rig completely intact so we could simply A/B between the two different systems.

The long and short of it, as to be expected, is that the Kiva/Kilo system was like listening to a set of nearfield monitors in the room, while the KF650s sounded like there were a couple mattresses stacked up in front of them keeping you from hearing any direct sound and instead deflecting all the energy everywhere but at your ears.  The clarity and musicality was something I’ve never heard in the room before.  The room became FAR less of a factor because it didn’t feel like you were listening to the room – you were truly hearing what was coming from the speakers.  Subtle EQ translated, TL Space verbs sounded amazing on vocals, and everything just sounded like it should BEFORE you touched channel EQ.  The low end was tight and defined.  Kick and bass each had a place in the mix and someone said you could actually hear the string noise on the bass rather than just hearing tones that you assume are coming from a guitar.

The tests on Tuesday afternoon confirmed everything I suspected about the importance of the correct PA solution for the room and we’d never see the kind of progress leadership is looking for with volume without it.

We decided to go with the system for the midweek service and the results were pretty solid.  The entire night averaged 5 dB less than normal (mid-80′s to low 90′s rather than topping out at 95-97 dBA) but it didn’t feel like we compromised anything artistically.  With our 650s, there is a point you have to run at in order to have some degree of clarity and get over all of the interaction with the room.  A line array in the room all but removes that variable.  At the end of the service, we had an extended response time with broken down acoustic/vocal and the teacher trading back and forth.  This moment was the most beautiful thing I’ve ever heard in this space.  I pulled everything WAY back so the music was in the high 60s-low 70s dBA and it felt perfect for the mood, but the vocal had every bit the same clarity, presence, and body that it had when we were 20-30 dB hotter earlier in the night.  Amazing!

Learnings from this test is that, in light of economic realities, we need to consider how a smaller and lower cost rig would fit into our room permanently.  The Kiva/Kilo combo wasn’t my favorite box, per se; I felt like it was a bit harsh and bright.  However, it was certainly enough box to get what we need from it on a week to week basis.  The 4 SB118s were a bit light on the low end, though, but that’s to be expected.  I also felt like I wished they extended down a bit lower then they did. We’ll have to experiment with the correct type of sub and quantity to get this right.

The next test is a set of L’Acoustics ARC’s that will be in the room for a few services.  Even though I know the pattern of these boxes is not a good fit long-term for our room, I’m interested to see what we think after living with something different for more than a single service.

The best part is that this test set some further ideas in motion.  A bit premature to discuss right now, but there will be more thoughts later…

I’m trying a different video format for this one to see if I can improve the quality.  Any thoughts or feedback compared to the previous uploads?  Uploading to Vimeo takes a bit more time than native hosting with flash video, but it should load faster and look better.

This service has an exposed ballad with just piano & vocal.  I struggled to find a vocal EQ that would work – there was a wierdness in the midrange (1k-3k) that some on our team described as distortion but I described as a frequency hot spot.  Regardless of the EQ filters I tried in this range, it was impossible to make the harshness and “pain” go away.  In fact, that “painful” feeling existed in almost every tune in the day.  The worst part was that it seemed to be a moving target based on the note being sung so I was feeling pretty helpless.  I spent literally hours trying to deal with this but by the second night of dress rehearsal, was no further ahead then the outset.

On that second night, I had a revelation…the signal sounded normal on a board mix CD and in the headphones or monitors soloing the channel from the console.  That meant the problem must be part of the PA – perhaps worn drivers (its been several years since they have been replaced) or another malfunction.  Then it hit me to check the crossover points for the KF650s and lo and behold, the problem spot was occurring directly over the crossover point between the mid and hi drivers.  Now we’re on to something…

I called our local vendor who installed and performed the initial configuration of the London for advice and an outsiders perspective to see if he could give me any suggestions on the best path to follow for a remedy.  After listening to my description of what was happening and what I suspected to be the cause, he pulled up the EAW factory box settings while on the phone and suggested we double check the crossover numbers.  EAW has made two different models of the KF650s – the KF650E and the KF650Z (we own E’s).  As luck would have it, somehow when he first configured the BSS London BLU-80 a year and a half ago, he entered the crossover data for the Z’s rather than the E’s.  The result was the mid crossover point set too high, causing an aweful overlap between the mid driver and where it transitions into the highs.  This created all kinds of comb filtering and distortion as you swept through the 1k-2k range.  Each frequency caused a different horrible interaction between the two drivers.

Of course once I entered the correct settings, the difference was staggering.  Its like having a new PA.  Here’s a picture of the correct crossover settings:

As you can see, the large dips between the three different drivers is extremely important for correct box frequency response.  Rather than the mid low pass point being set to 1.12, it was set to 1.58.  That seemingly minor difference created HUGE problems!

MORAL OF THE STORY: Check and double check all of your crossover and factory specified settings. It never even occurred to me to check this stuff.  I’ve massaged all of the various parametrics since the box was installed, but have never opened the crossover and verified the correct settings.  Since finding this inconsistency, I’ve gone back and checked all of the settings for the driver parametrics and found a few other things that needed adjustment as well.  The result is literally night and day to the old settings.

Stop what you’re doing right now and verify your PA processing settings!  Everyone who listens to your PA will thank you.  Don’t take anything for granted…

I’ve meant to write about our PA processing for the past few months but an experience last week during Easter rehearsals has moved this up the topics list. We have a BSS Soundweb London BLU-80 that functions as our primary system controller. The London is tremendously powerful, allowing the user to create a completely customized signal chain that is only limited by the available processing power. In our configuration, it takes separate stereo music and speech L/R mixes from the Digidesign D-Show Profile and handles all system equalization, delay, crossover, and limiting to 12 output channels:

1. KF650 Left Hi
2. KF650 Left Mid
3. KF650 Left Low
4. KF650 Right Hi
5. KF650 Right Mid
6. KF650 Right Low
7. KF300 Left Outside
8. KF300 Left Inside
9. KF300 Right Inside
10. KF300 Right Outside
11. KF260 Downfills
12. Balcony Delay

Here is the routing design in London Architect for how the BLU-80 is configured. Each box on the diagram below represents a processing unit. From left to right, yellow boxes represent physical inputs or outputs from the box, orange represents delay, purple is equalization, gray is a high pass filter, green is a matrix mixer, blue is gain control, and green is limiting.

The signal flow through the box is as follows:

• Signal enters and is separated onto two paths – one for music and one for speech. Each goes through a delay that aligns the master mix to the subwoofers (12 ms) and then into a master “house” parametric EQ. This is the artistic curve that adjusts the frequency response of the entire system. There are different curves optimized for music & speech.
• After passing through the artistic parametric EQ, each of the two mixes are split into a full range version and a high passed version that then hits a matrix mixer which routes to the various individual speaker chains. With music, for example, a full range mix is sent to the main KF650s, but a high passed version is sent to the downfill speakers and center cluster in order to reduce 250 hz buildup in the room. The matrix mixer really acts more as a router – its all or nothing for each speaker channel, either full range or high passed.
• From there, the KF650s hit another parametric that allows for some sculpting of the 650s as a whole. The output from this parametric hits the three-way crossover for the boxes. These crossover points are factory specified. From there, each driver’s output hits a dedicated parametric with factory specified settings, driver gain controls, and then driver limiting.
• For the KF300s, KF260s, or Balcony feeds, the output from the matrix hits a dedicated parametric for box sculpting, then a parametric for factory specified box settings, gain and limiting, then channel delay to align with the KF650s.

While you read the signal chain in the London from left to right, the process of equalizing and aligning the system basically works from right to left. The first step is to enter all of the factory box settings, gain balance between low/mid/hi drivers, and align the delay times between the various boxes. From there, we use EASERA to start with the KF650s, scope them, and make adjustments until the response is as desired. Then add the KF300s one at a time, scoping them individually, then adding each one into the 650s and making the necessary adjustments. Continue the process until all of the boxes are active and frequency response is fairly flat. At this point the artistic master parametric EQ comes into play to sculpt the overall response to fit the mix and the room

What is Timecode? 
Timecode is simply a way of recording a “time” along with either a visual or audio event. This allows you to say, “The beginning of the song is at 0 and the cymbal crash happened at 2 minutes and 53 seconds.” The information is then stored along with the audio or video medium so that, as it is moved from device to device, the time information stays the same.

The most commonly used timecode format was developed by the Society of Motion Picture and Television Engineers. Referred to as SMPTE (pronounced Simp-Tee), it was originally developed to synchronize sound for film but has since been adapted for video as well as audio-only use.

The other commonly used timecode format is Midi Time Code (MTC), which was developed by Chris Meyer and Evan Brooks for Digidesign.

How does Synchronization work? 
Imagine that you and a friend have the only two watches in the world, so that you have no time reference other than your own watches. You need to meet at an exact time later that day. No watch is exactly accurate – maybe yours is a bit slow and your friend’s is a bit fast. Over the day the result is that there is a 10-second difference between your two watches. While it may be okay with your friend that you’re 10 seconds late because he likes you, it would be very bad for a cymbal crash. In audio, even delays of a fraction of a second with doubled vocals can be noticeable.

The next day, in order to circumvent the 10-second time difference, you and your friend work out a system. You pick one person as the Master and one person as the Slave (it’s that kind of friendship). The Master calls the Slave on his cell phone and tells him what time it is every few seconds, and the Slave resets his watch to match the Master’s. While this system may be an annoying prospect for human beings, it works quite well for machines. Similarly, in every timecode set-up you must determine which machine is the master (there can be only one) and which are the slaves. Usually the constraints of the machine will define who is master and who is slave. 

How would I use Timecode? 
Sync-to-Video - The simplest video sync scenario works much like this: a VHS tape has LTC on one of its audio tracks (it is the Master). The device to be synced (the Slave) reads that timecode and chases it. The tricky part is determining which type of SMPTE frame rate to use.

When using SMPTE at 30 frames-per-second (fps), one second equals one second. Unfortunately, NTSC color video does not run at 30fps — it runs at 29.97fps. Why in blazes does it do that, you ask? Well, let’s just say that video, like life, was simpler when it was in black-and-white. But when color was introduced, they decided to cram that little bit of color information in the space left by .03 frames.

At 29.97fps, a second of material is no longer a second, but just a hair longer. While a few hairs may not matter to you and me, they matter a lot to people in television who make their living selling seconds, so these people came up with a solution called “drop-frame”. This means a few frames are dropped so that SMPTE time matches with real time.

So we have SMPTE at rates of 30fps, 29.97 “drop” and 29.97 “non-drop”. All that really matters on a basic level is that you need a tool that can handle all these different frame rates, and you must know exactly what frame rate the video you are syncing with is using.

In the next post on SMPTE, I’ll go over our setup and implementation, along with some learnings from an intense production process.

We now use three sets of ambience/audience mics, each with a different purpose for capturing the room.  Inspired by some great discussion on the Digidesign User Forum, my implementation is…

• Mics #1 are a set of shotgun mics (purchased for this project) on the outside corners of our stage, mounted just under our side screens, pointed out into the room.  These are the primary audience pickup source because so much of the audience’s energy is directed towards the stage that these do a great job of capturing without adding too much PA to the mix.  
• Mics #2 are a set of Crown PCCs placed on the lip of the stage at the 1/3 and 2/3 lines from left to right.  These combine with the shotguns to provide some presence and immediacy to the signal since another set of shotguns wouldn’t be acceptable aesthetically in these positions.  
• Mics #3 are a pair of Shure SM81s hung from the first catwalk, almost at the half way point of the room, also on the 1/3 and 2/3 lines to the stage.  These mics serve almost entirely the purpose of room ambience – making the mix sound like it is happening in the auditorium rather than happening in a dead studio.
• We may still look to add an additional set or two but I’m stuck because adding them will mean needing to deal with delay times between those mics and the rest currently in place.  Two of my current three sets are directly in line with the PA so there’s little issue there.  The delay induced by the third set certainly does some phase stuff to the mix but the added space it puts in the mix when the speaker is up is worth the compromise to me.  So we’ve dodged the bullet to this point and I’d like to avoid having to go there with more mics.  This is one of the only limitations of creating the broadcast mix inside the Venue rather than in an outboard mixer/processor.  There is no way to have the degree of time alignment control that would be needed in the broadcast feed without making other more significant sacrifices.

All channels on the console are split up into one of three main bus paths – master L/R is music, center is speech, and subgroup 7/8 for standalone playback sources.  These three paths are then combined in the matrixes to feed the PA and outside world.  Our PA processing is also set up for speech and music, so there is a stereo music matrix (with playback included in it) and a mono speech matrix that feed the PA.  On the video side, music, speech, and playback are combined with the three sets of ambience/audience mics in a stereo matrix.  A nice Massey mastering limiter, phoenix cranesong, and 7 band EQ for any trouble spots are put over that mix as a whole and the finished result is what you hear below.

According to Smaart, music averages at 88-95 dB in the auditorium, speech and videos run 68-75 dB. That equals an average 20 dB natural difference that feels right in the main auditorium. In the processing of audio for video, I am actually adding back 6 dB cumulatively to speech, making the speech/music difference 14 dB rather than 20 dB and this is really important.  Too little added back makes speech too low compared to music, so you have to turn it up.  But too much added does the reverse – turn it down when the speaker comes up.

That is why playback has to be split up from music in the broadcast matrix.  If playback is lumped in with the band, it comes out too low on video compared to speech because playback is in that lower dynamic range in the room.  But if playback were lumped in with speech directly instead, it would be in mono and receive all kinds of undesirable processing in the speech world that isn’t desired.  So…on the PA side, playback and music are each sent to the music PA matrix at equal volume.  But on the video side, playback is sent to the matrix at the same level as speech but through its own path.  So if speech and playback were sent to the matrix at 100% level, music is sent at 90% level.

Here’s a sample of the finished result of worship from our New Community service last night…

In my next post I’ll outline the final steps necessary to fine tune for building feeds vs. broadcast.

My next post will outline the steps we’ve taken to implement a new process.

One of the biggest decisions that I’m most excited about is the new PA that will be installed in Troy’s auditorium next summer. After nearly 6 months in process with Acoustic Dimensions, our acoustical and systems consultant hired to lead the selection and design, this past week we finalized our decision. Drumroll please…

The main left and right clusters will be comprised of seven Qi-1 line array cabinets with two Qi-Subs hung in an adjacent column on the insides of each cluster, serving as mid-bass boxes. Three total Ci-7 cabinets will be used for outfill and center fill. 4 B2-Subs will be placed under the stage in ultra mode to provide the bottom end to the system. EAW UB82 will be used for front fill on the stage lip, JF80s in two rows for over balcony fill and more JF80s in a single row for under balcony fill.

Amplification will be a mix of d&b D12s for the main arrays and Lab Gruppen 28:4 & 48:4 for fills. Drive processing will be provided by BSS London Soundweb Processing (BLU-800 and BLU-120). We are using a BLU-80 in our current PA and I’ve been very impressed with its capabilities and sound, so I’m sure it will be a good choice for the new rig.

If you’re interested, I’ve attached the drawings of our room with box placement.

Boxes in the air

Boxes on the floor or under balcony

Elevation

1. Create two subsnakes from the multipair I’ve found – one for drums and one for keys. Drum snake should have 12 XLR channels and 4 XLR returns, keys snake should have 6 channels and 4 returns.

2. Implement an organization plan for audio gear in the auditorium. There are so many small pieces and parts and they are currently spread between three storage locations. I would like to centralize everything, label it with the label maker, and make things overall much easier to find and stay organized. At the same time, all of the wireless need updated channel labels as many have peeled off or are close to it.

3. Implement the use of a keys mixer for headphone monitoring rather than wedges as normal practice. The second Mackie 1402 will be a perfect fit for this. Need to also devise a stand and cabling that will look clean for this purpose.

4. Hang all of the lav mics on hooks by type so that they will not have so many kinks and bends in the cables.

5. Make six or eight more 10-15ft mic cables to use with the subsnakes.

6. Clean up the console default scene to better implement video submixes, monitor mixes, effects patching, etc.

Input List