I mentioned this on Twitter a few weeks ago but figured it was worth a full blog post. A little while back I happened on a weekly video podcast led by Dave Pensado called Pensado’s Place that has quickly proven it should be required viewing for anyone serious about mixing. If you don’t recognize the name, I promise you that you know Dave’s work. Dave has mixed hits for bands like The Black Eyed Peas to Justin Timberlake, Elton John, Jamiroquai, and Christina Aguilera. In 2002, Dave received a Grammy for his work on the hit single “Lady Marmalade,” featuring Christina Aguilera, Lil’ Kim, Mýa, and Pink. Do I have your attention yet?

Every week Dave and his manager Herb Trawick host a roughly 45 minute program that is full of techniques, q & a, and interviews with well known current mix engineers. The content is really well done and it’s really important to me that we support Dave in this since it is highly unusual to find someone as successful as Dave creating this kind of resource.

One of my favorite episodes was #4 when he interviewed Tony Maserati. I’m linking to it here to get you hooked and then go check out Dave’s site to catch up on the rest of the back episodes already in the can.

If you follow me on Twitter, you probably noticed quite a bit of chatter a few weeks ago regarding a post Mike Sessler wrote back around Christmas about mixing FOH with in-ears that has generated quite a bit of buzz.  It was recently republished on ProSoundWeb and in Live Sound magazine.  You can find the original post here, along with a later response here.  Dave Stagl wrote a great response to the topic here.  I took it pretty squarely on the chin by some people for challenging such a popular post so I thought it might be a good idea to try to more clearly outline the why behind my objection with more than 140 characters per post.  I’m going to try not to re-tread the things already written by others so make sure you catch up with the original posts if you haven’t already.  This actually piggybacks well on an overall topic that’s been on my heart for a long time but I’ve not had the specific inspiration to outline fully until now.

I agree completely with what others have said that the most important thing is that the people signing Mike’s checks are happy with the results he is getting from a challenging PA deployment and mix position.  I also fully understand that you need to do what you need to do when you’re mixing in a challenging room, and I’ve certainly had my fair share of those kinds of situations.  What actually set off the whole epic Twitter conversation was a tweet I saw posted to Mike that said this:

SOUND GUY #1: I was a doubter but I just mixed all practice with IEMs and it sounds really good! I just got a new method.

ME:  Boo!  I don’t get disconnecting from the live room.  It’s called live mixing for a reason! #justbecauseitsonablogdoesntmakeitgood

In the conversations that followed, here are a few examples of other responses I heard:

SOUND GUY #2: I could find the specific balance between instruments faster in my ears

ME: I would argue spend the energy learning to find that balance in the room rather than IEMs

SOUND GUY #2: Why? What is the argument for taking the harder path?  I’m not opposed to hard work if it has significant benefits

ME: Because your job is to mix for what the audience is hearing.  They aren’t listening to IEMs with ambience mics.

Another snipet went like this:

ME: How do you compensate for what’s happening in the room?  Great mixes, especially in worship, are emotional & responsive.

SOUND GUY #3: Same way a worship leader using ears does. Pay attention, watch and feel what’s going on around you. Respond to the moment.

SOUND GUY #4: Well hopefully the worship leader has audience in his ears and not solely depending on his eyes to gauge reaction

SOUND GUY #3: Which is the same reason we put the audience mics in our IEM reference mix.

I believe strongly that this conversation reflects a path some are heading down that is the antithesis of sound technique for mixing great music in churches.  I’m going to probably go a bit more spiritual here then I normally do on this site, but after mixing more than several thousand services professionally, I’m really passionate about this from first-hand experience.  Mixing worship is a beautiful dance that happens with every note between the musician on stage, the engineer creating the mix, the audience who is participating, and the Holy Spirit who is hopefully moving throughout the room.  A weakness or mis-step in any one of these areas will hinder the overall experience.  Try having an engaging worship experience without the Holy Spirit being present and tell me how that turns out.  Expect amazing worship with a disconnected audience and while the musicians may enjoy themselves, the overall room will suffer.  Really great music played by great musicians with a really responsive, great mix takes the experience for the audience to an entirely different level.  I believe all music and the emotions stirred within us as we engage come directly from God, so really this dance applies to any great music experience.  Stand in a stadium as U2 plays “Where The Streets Have No Name” and tell me that God isn’t behind the feelings you’re experiencing.  No Way!  Experience Fleetwood Mac playing “Don’t Stop Thinkin About Tomorrow” with the USC Marching Band and tell me that isn’t a spiritual experience.  No way!

When you put IEMs in your ears or headphones on your head while you mix, you are altering that balance between those four pillars in a fundamental way.  It is impossible for you to react to what is happening in the room the same when you are disconnected artificially from what’s happening in the space.  Yes you can put ambience mics into your reference mix and technically you are hearing the “audience” while you mix, but in no way is this the same thing.  You’re removing the intangible, inexplainable spirit from mixing and trying to replace it with a “technical” solution that while on paper is giving you the same technical experience, IT IS NOT THE SAME!  You might as well be outside of the room and mixing from a video monitor of the room.  There is no end to this slippery slope.

The only compromising solution I’ve ever experienced that will improve the mix location while still keeping you somewhat connected to the room is nearfield monitors at FOH, delayed back to the PA so they are time-aligned.  This is still a huge compromise because it is very different having speakers within several feet of your ears vs. the rest of the audience who is 30-100 feet away from drivers, but at least in this scenario you are still connected to what is actually happening in the room.  If something goes left while you’re mixing, the likelihood of you catching it is much higher.  Personally I love having monitors at FOH vs headphones.  I have mixed that way for three years and I could never go back.  I don’t even keep a set of headphones at FOH when I’m mixing – if I need to solo something up, it goes in the monitors so it places in front of the PA soundstage for me.

Are there some people using IEM’s on tour with good results?  Sure.  There are guys doing close under-micing of every cymbal in a drum kit too but that doesn’t mean that is the best approach to a natural and dynamic drum mix.  There are lots of things being done in live audio all over the world that someone has good luck with but doesn’t deserve to be replicated widely.  I promise you that you will not find one of the greats – Robert Scovill, Joe O’Herlihy, Dave Skaff, John Cooper, Dave Kob, and countless others – mixing for long stretches of time with something covering their ears.  Those are the guys that I aspire to mix like, and if you’re serious about this art, you should too.  Just like great stage artists of our time, these engineers’ work speaks for itself.

“The difference between greatness and mediocrity is not measured by the quality of the tools but by the quality of the approach.” – Robert Scovill, 2008 I believe @ Willow Creek Arts Conference

Here’s my point. It’s way too easy nowadays with all of the tools at our disposal to make mixing an intellectual, artificial, explainable task. We snapshot and program everything, we virtual soundcheck from a static performance for hours and hours, we use racks of plug-ins and outboard processing that would have never been available even 5 years ago, all in search of elusive greatness.  It’s human nature to take something that is unexplainable and try to make it explainable and repeatable.  I read many blog posts, magazine articles, and podcasts from experts with the best intentions every month, but the result way too often tries to make the inexplainable explainable.  Playing and mixing music is not an intellectual and repeatable task.  It’s mysterious…what works one day doesn’t necessarily the next.  The band doesn’t play the same every time.  The mix doesn’t develop the same every time from your snapshots.  Mixing is every bit a skill and an art that no matter how long you work to perfect, you cannot wrangle fully.  There’s always something new to learn, something new to adapt to. That’s what I love about what I do.

I challenge you with everything I have to embrace the tension of this beautiful dance.  It is not comfortable and it is not easy, but that’s part of the fun.  If your mix position is awful, I would argue there is NOTHING more important to address in your technical budget.  The reality is that people will tolerate poor lighting and poor video because neither are invasive to their person, but bad audio will keep a visitor from coming back to your church or a fan from buying another ticket to your show.  Can we fix every PA problem?  Certainly not – budgets often don’t allow it.  I mixed on an aged EAW KF650 rig for three years, 5 services a week, so I know well the frustration of knowing what the mix could be if it were only coming through a better transmission system.  If you have to come up with a creative solution to get you by until you can fix the real problem as Mike has done, by all means do whatever it takes to get the best audio you can get.

But I will disagree with you you every time if we’re considering mixing IEMs a superior technique outside of these compromised scenarios.  Master the art of mixing.  Only snapshot the bare minimum of what you can’t physically mix by hand.  If the band isn’t going to play it the same every time, why should you mix it the same each time through programming, plug-ins, and snapshots?  Plug-ins are an incredible tool, but run away from anything that takes you too far into the analytical brain and away from listening, mixing, and experiencing like a fan.

“Mix like a pro but listen like a fan. Always try to gain more of the listener’s perspective and turn off the analytical brain” – Scovill from the same conference.  This quote has always stuck with me.

Thank you for considering my perspectives and sticking with me through such a long post.  This actually sets up well the conversation we’re going to have next week at Gurus of Tech in Chicago on Mixing – It’s an Art, not a Plug-In.  I’d welcome dialog about this topic either here on the blog, via Twitter, or offline.

Yesterday we covered the thought process in tuning the main speakers in our PA.  Today we’re going to discuss delay fills.

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.

When I first visited newlifechurch.tv last winter, I knew immediately that one of my first tasks when we got here would be to start from scratch with PA tunings and system optimization across the board.  The tell-tale sign this would be necessary came from looking at channel EQ on the consoles.  Channel after channel showed LOTS of EQ and I heard complaints that the board mix that fed the internet campus never sounded as good as it did in the room.

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…

I’ve written over a year ago about the audio for video processing chain I developed at Kensington with a Venue console but thought it might be useful to revisit the process in a new place with a new console (this time, Yamaha M7CL’s).  The cool thing is that I’m using the exact same techniques as last time and the results we’re getting are stunning.  If you haven’t adopted a process like this for your outside world feeds, why not?  Seriously.  Ok, here we go.

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.

Today’s trick for the M7CL is not quite as involved as last week’s, but I think it is very powerful for efficient workflow.

Everyone has their own ideas for how best to lay out the 12 user defined keys on the console.  Up until a few months ago, my norm was for keys 1-8 to be direct sends on faders buttons for mix busses 1-8, with keys 9-12 reserved for a couple mute groups, talkback, and a scene advance button.  However, I discovered an option for user keys that I used to utilize on the PM1D and never realized existed on the M7CL:  page bookmarks.

The touchscreen on the console is a blessing and a curse to me.  There are some parts of operating the software that lend themselves quite well to a touchscreen but sometimes I miss having more direct access buttons to things such as the EQ & dynamics sections, as well as FX processors that reside on the Rack page.  Assigning a user defined key as a page bookmark is REALLY handy because it allows me to bring back some of that fast access I want to menus or screens I’m getting to all the time so I can remove a touchscreen key press and instead get somewhere quicker with muscle memory and button feel.

I’ve started setting up my user defined keys so the first row of 4 are set up as page bookmarks that get me in 1 button press to the EQ detail, Dynamics detail, and my VOX Verb & VOX Delay.  It’s a simple thing but it feels like it speeds up my workflow around the desk because now I can select a channel, press my EQ shortcut button, and then start dialing away at that channel’s sound.  Same for dynamics.  I’m constantly going to my verbs to dial them in exactly how I want them to sound.  Without this shortcut, getting to that edit screen is at least a 2 step process and might be 3 depending on what screen I’m coming from.  Now 1 press of the shortcut brings it up and another brings me back where I was previously.

Check it out.  You might find page bookmark shortcuts does the same for your workflow.

I’ve been blessed to spend the past 4 years or more mixing on some really great, really big desks.  When I first arrived at Kensington, we owned a Yamaha PM1D that, while I’d had previous experience on years before, I enjoyed learning inside and out.  Then 2 years later we changed directions and embraced the Digidesign (now Avid) platform.  Along the way I also was able to get my hands wet a bit with Yamaha M7CLs.

There are some tricks I’ve learned along the way from absorbing content from accomplished engineers in the field that made life mixing on the Venue and PM1D a bit easier and gave me better mixes.  A new challenge since coming to NLC.tv has been trying to find creative ways to get the same bang for the buck out of the M7CL.  I had already started exploring some of this process in my last few months at Kensington but I’m working now to flesh them out a bit more.  I’m going to share some of my favorite M7CL tips and tricks over the next few entries (there’s too much here for a single post).

Today we’re going to tackle parallel compression.  I’ve written on this before, as have others, but in its simplest form, parallel compression means double bussing a set of inputs to two different signal paths on the console.  In the first path, everything remains clean and unprocessed.  In the second path, a nice compressor is placed over the signals and they are compressed as a group, usually pretty hard with variable attack & release times depending on the song.  Then the clean and squashed signals are recombined before going to the stereo bus on the mixer, for me usually at a 2:1 ratio of clean to squashed.  This is especially magic for vocals and snare/toms for me.  Mixing the styles of music that I do, vocal intelligibility is normally one of the most important goals I’m fighting for and getting the vocal to sit nicely in its place with the rest of the band can be challenging.  Since I started implementing this parallel compression trick last fall, it has done wonders to the ease with which I can accomplish vocal consistency I really like.  It becomes an even more powerful tool the larger the vocal group becomes.  At Kensington it was normal to only have a single lead vocal and perhaps a BGV or two.  At NLC.tv, 5 to 6 vocals is the norm with sometimes as many as 7 or 8 on a given weekend.

Doing the parallel compression thing on an M7CL is really easy.  First, I like to set up 2 busses as fixed busses instead of variable so I don’t have to worry about making sure the sends to them are all at unity.  This can be accomplished under Bus Setup in the console setup menu.  Next, as long as I have enough mix busses available, I like to set up one buss for the clean group and just call this one VOX.  I unassign the VOX channels themselves from going straight to the L/R buss and instead route them to this VOX “subgroup”.  While I plan to keep the processing here as clean as I can, especially the larger the vocal group you have, it can be really handy to have a single place you can grab an EQ and deal with a problem area that effects all of the vocals during the heat of mixing.

Now I also route the VOX channels to the 2nd group that I call VOX Smash.  This group is setup just like the first one except on this one I engage the compressor on the buss, set to a 6:1 ratio with a medium attack and release.  The M7CL has an excellent feature called automatic delay compensation so even though the same channel is going through two signal paths with different processing times, they stay perfectly in sync so that when they are combined into the master L/R buss, they are still in phase with each other.

If you’ve never tried this concept before, I can’t suggest strongly enough that you do.  I always had the preconceived notion that a trick like this was only available to execute on larger desks but have been very pleased with the results I can achieve on our M7CLs and have started sharing the love with all of our engineers on this easy and effective mix technique.

This one goes in the category of why didn’t I think of this sooner?

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.

Microphone positioning and technique is largely a matter or personal tastes – usually whatever “sounds right” probably is right.  Nevertheless it’s a good idea to remind ourselves of some of the basics for getting there.

Following are a few tips that you might consider following when micing musical instruments for sound reinforcement.

• Try first to get the instrument to sound good acoustically before miking it.
• Use a mic with a frequency response that is limited to the frequency range of the instrument.
• To determine a good starting mic position, try closing one ear with your finger. Listen to the sound source with the other ear and move around until you find a spot that sounds good – put the mic there.  Remember, this may not be practical (or healthy) for extremely close placement near loud sources.
• Remember that the closer a mic is to a sound source, the louder the source is compared to reverberation or ambient noise.
• Place the mic only as close as necessary, keeping in mind proximity effect.
• When possible, use as few microphones as possible due to the Potential Acoustic Gain rule which tells us (among other things) that the volume level of a system must be turned down for every mic added in order to prevent feedback.
• If the sound from your loudspeakers is distorted even though you did not exceed a normal mixer level, the mic signal may be overloading your mixer’s input.  To correct this situation, use an in-line attenuator or pad to reduce the signal level from the microphone, or just back it away from the source some.
• More than anything, experiment and listen!

(Thanks to Shure for the tips.)

I’ve had quite a few questions regarding ambience mic technique, what to use, etc.  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 Audio Technica 835 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.  They’re inexpensive and sound good for what I’m using them for.  You could certainly upgrade the make/model here to something more boutique but this serves our purposes well.

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.

The three sets of ambience mics serve dual purpose for us – ambience for IEM and ambience for the broadcast/record feed.  In the IEM world, we lean most heavily on the shotguns at the front of the room at the corners of the stage and supplement that with the PCCs on the front of the stage towards the center.  These mics are used mainly because they provide localization of what the artist is hearing in their ears.  For example, when someone to the artist’s right calls to the stage, we want everyone on stage to turn their head in that direction.  Likewise, if someone on the front row is singing their heart out, it’s cool for the artists to be able to sense that from those PCCs on the front of the stage.  That localization goes leaps and bounds towards breaking down the isolation.  Every artist is different with how much ambience is just right.

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.

The shotguns are the most important piece of the puzzle for me because their pickup pattern is most effective at picking up the audience and rejecting PA and stage bleed.  The PCCs are next important to fill in the center stage imaging – think of the shotguns panned hard left and right and the PCCs panned at 9 o’clock and 3 o’clock or 10 and 2.  The hanging mics are almost halfway back in the room and that time delay adds depth to the ambience space. For IEMs, it can be distracting.  But for broadcast, it enhances the size of the room and makes it feel more real when you’re listening back.

I ran across some cool conversations happening on the Sound on Sound forum a few weeks ago in reference to unity mixing.  If you’ve missed the various things I’ve written about the topic, there are links on the homepage of this blog to the articles.

I’ve also seen another forum where one poster blasted me and called me a moron for approaching this topic.  It has been fun to see how literally some people take this whole thing and think I’m somehow saying you should literally mix your show from the gain knobs or advocate poor gain structure in your console or the rest of the system.  QUITE the opposite.  I run my console really hot because we all think the Digidesign mic pre sounds better when its hit pretty hard and the desk as a whole sounds great when you get up and go with it.  I think some are missing the point of my suggested unity mixing completely, so let’s dive into this just a bit to hopefully clear it up.

The whole point of this discussion is to make the inputs well mixed while setting gain, rather than just blindly going channel by channel and setting each to maximum individual gain.  It doesn’t have to be perfect but just get things in the neighborhood so you don’t have to run some faders -30 or -40 dB below others.  This makes mixing monitors from the same desk (which we do at every Kensington campus) MUCH easier.  In this configuration, the inputs are well mixed, so that the sends to the monitor mixes and FOH faders are (more or less) at the same place across the board, making adjustments for the band a lot easier.  If I know the band well, I can often dial in their monitor mixes from the FOH board ahead of soundcheck and nail it with few, if any, changes needed.  We’re still running the inputs as a whole as hot as they should be through the desk so let’s not get into the whole topic of maximizing digital bits in the desk and the like.

Another example… imagine trying to adjust a monitor send, pre-fader of course:

A) with a very hot input, therefore with the FOH fader pulled down a lot, and
B) with a low input setting, with the fader pushed nearly all the way up . . .

‘A’ will be very sensitive to tiny adjustments at 9 o’clock and
‘B’ will need large movements at 4 o’clock.

With a mess like this all over a big mix, the experience is not nice trying to keep the band happy with their wedges and that is what gets so many of our engineers in trouble!

Second, if you end up with a difference, for whatever reason, where one fader is sitting at -30 or -40 dB most of the time, you have a MUCH harder time mixing due to the same amount of travel for a 3dB difference at or near unity, now gives you a 10-20 db difference with every move (Or whatever it works out to on the board, but a LARGE difference).  This makes fine tuning a mix really difficult.

That’s it.  Does this help make the concept clearer at all?

We finally made it to the last and likely most important mix in the IEM series…lead vocal/worship leader.  In this particular example our lead vocal is also playing acoustic, so not surprisingly you’ll hear those two inputs most focused in the mix.

As with some of the other mixes we’ve discussed, our vocalists seem to largely prefer their voices dry…if they want to wetten things up, adding more of the extras channel does the trick.  This works well in our particular setup since the vocals are fed to the IEM system direct off of board channels – adding verb would increase the complexity of routing.

I strongly encourage everyone who wears ears to keep both sides in all the time.  My primary reasoning is that your brain does a cool thing when it receives similar input from both ears in summing that source 6 dB internally.  So, if you only wear one ear, you’re going to have to turn up your IEM mix approximately 6 dB in order for it to feel as loud as both ears in.  Obviously you have to be really careful with this because an untrained artist could easily cause themselves hearing damage night after night of crazy levels in their IEM.

I say all that to say, the mix an artist wants changes pretty dramatically depending on whether they are wearing one or two ears.  These mixes you’re going to listen to today are one ear in mixes, so you’ll find that the acoustic, vocal, and click are WAY more out front then in the other mixes we’ve analyzed since the artist is getting a lot of their sound ambiently through their other ear not wearing the IEM.

Our philosophy at Kensington regarding ears has been to do everything possible to gain band acceptance.  By doing that, the stage volume lowers significantly and vocals have a much easier time just hearing themselves through a wedge.  I’ve mentioned that we have 6 channels of wired ears and 1 wireless transmitter/receiver system.  Because of this wireless limitation, we haven’t pushed very many vocalists to jump to ears.  This isn’t necessarily a bad thing since vocalists often have a much harder time adjusting to performing through IEM systems then instrumentalists because it can effect their pitch center, how loudly they perceive their head voice, etc.  So we’ve had good success getting musicians on ears and, by doing that, getting enough sonic space on stage for the vocalists to be able to hear themselves comfortably through wedges and not have to deal with their transition yet.

I’ve heard from another large church that they have the policy that a new vocalist must sing BGV on ears for at least 6 months in order to get used to them before they would ever ask them to sing lead VOX.  I can certainly see the wisdom in that approach.

We’ll wrap this whole thing up next time with the live mixes of the 2 songs we’ve been using for test material so you can hear overall context.  See you then!

Continuing the IEM series, today we’re talking about the keys mix. We’re departing from the two tunes we’ve used for the rest of this series since there weren’t any keys in that band.

On a given week, we submix as many as three keys sources in mono to send to the IEMs. The first source is Ivory from our Slam Grand.  If the piano shell is not in the set, often times I will still run MIDI out of the primary keyboard into the Ivory system so that tone is available if we want a grand piano patch in the day.  The other two inputs are our house keys – a Yamaha Motif8 and a Korg Triton.  Every once in a while one of our artists might bring in their own boards but it doesn’t happen very often in a month.

I was concerned when we first installed our IEM system that the keys would need to be in stereo just like the drum kit in the ears mixes and that would present channel count issues with our desired allocation.  Thankfully, I don’t think anyone has ever commented on the keys being submixed to mono rather than stereo.  As a result, the only truly stereo submixes in the ears are the drum mix and the extras channels.  Everything else goes to the IEMs in mono and can be panned around in each individual mix as desired.

I’ve received a few questions via email for more information about the ambience mics specifically so I am preparing more info about that to tag onto the end of this series.

With that said, check this out…

Next time we’ll listen to the final IEM mix – lead vocal/worship leader.  Thanks for reading!

Continuing our series on IEM mixes, today we’re going to focus on the drummer’s mix.  If you’ve missed the other parts of this series, go back through the last few days posts to catch up.  Sorry for the data dumps in some of these posts but I’m just trying to make sure I cover things as thoroughly as possible.

When we first adapted the PQ system for our IEM mixes, I was concerned that a stereo drum mix that everyone shared might cause problems.  The reality couldn’t be further from that.  A stereo aux creates the submix for the drums and it is slaved to the channel pan so whatever panning is done for the house translates to the ears.  I actually leave the sends to this aux set to unity in my default console setup.  That way, as I soundcheck the drums, I can build my house mix with the faders near unity and it translates almost perfectly to the ears.  It is RARE to need to tweak the mix for the ears – I actually can’t tell you the last time I listened to it other than preparing for these posts.  It just works every time.

By now you should be sensing some trends.  The ears mixes are pretty dry and really simple.  If we had a separate monitor desk, I’m sure there are more things we could do to make the environment even better for the artists (things like manually riding audience levels, riding solos, adding effects if desired, etc) but this system works for our guys, as evidenced by the wide acceptance factor.

Song #1 is still “My Savior, My God”.  Check it out…

Song #2 adds the click and track.  Again the click takes almost singular focus in the mix since everyone is locking to it.  I can’t speak very much into what it takes to be successful with the whole click track thing from an artist perspective.  Kensington has been using click’s for so long that it was just something we do by the time I got here.

Again, if there are questions, let’s use the comments section to discuss.

Next time:  keys mixes.  We’ll have to use different songs then the ones we’ve been using so far since there wasn’t a keys player in this band.  It will be good to change it up a bit, as well!

Today’s IEM mix we’re going to evaluate is from our bass guitar player. Again, there’s very little processing on the mix. This one would probably benefit from listening on better speakers then those on a laptop. The experience is pretty different for me listening to these clips on my speakers vs. on some good IEMs so you’ll probably find the same thing.

Right now is a good time to stress the importance of good ear pieces in order to achieve wide acceptance.  When I started at Kensington, all that we owned were some Shure E1s and NO ONE was excited to put those things on for anything.  The first time we did a band with ears for a holiday, everyone but the bass player wore E1s and it was painful.  Now we use Shure products exclusively (I’ve written about this before – check the archives if you’re interested in why) and have a mix of SCL3, SCL4, and SCL5′s.  Since I consider us still in the early days of a transition campus-wide to IEM, only a small handful of our artists own their own ear pieces.  Most use our house sets.  We stock the replacement foam and change it for each set whenever a new artist is using them.

Focusing now on the mix, our artists get a stereo drum mix to their PQ mixer for their IEMs. Everyone gets the same drum mix and we find that works fine for us. The only exception is that the bass player also gets the kick channel individually in addition to the stereo mix since bass players always want more kick then the rest of the band. When you listen to the first clip from “My Savior, My God”, the first thing that jumps out is the placement of kick and bass in relation to everything else in the mix. There’s also quite a bit more of the extras channels in this mix then in the electric mix we listened to last time. One caveat though – due to the channel limitations of our system on this particular night, the bass player chose to have mono extras rather than the normal stereo. Personally the stereo part of ambience is really important to me but this particular artist isn’t reliant on ambience so its a simple trade off. I’m actually surprised in listening to his mix that he has this much extras present. Regardless…

The second clip, like last time, adds a click and track into the mix. What is interesting is that with this mix, the click is pretty buried. In fact, once the drums kick in I don’t think it is very present at all. Again you can get a better idea of the importance of the extras channel in the mix when you hear the crowd singing along.

Again, if there are questions, let’s use the comments section for this. I’m really not sure what else is helpful to share.

Next time: drums IEM mixes

Today we’re continuing the series on IEM mixes.  I’m in my fourth year mixing at Kensington and when I started here, every artist on stage used a wedge and those wedge mixes were created for better or for worse at FOH.  The job was daunting, to say the least.  Fast forward to today and all but three artists (40+) now use IEM solely when they play at our main Troy campus (the portable campuses have not begun transitioning to IEM yet but hopefully soon!).  All of these things I’m sharing in this series of posts are things we’ve learned along the way in order to gain such wide acceptance from artists.  This is by no means the only way to do it but it is OUR way.

The first IEM mix we’re going to evaluate is lead guitar. Most of our worship arrangements are electric driven so this is a pretty important place to start. Our mixes are generally very dry and we use the extras channels to add space when desired. I’ve worked with artists who like more processing in their ears but this setup seems to work for our artists since we don’t have a dedicated monitor engineer.  There isn’t any significant processing on the mixes you’re going to listen to – they are created with a stereo PQ channel on the Venue, the on-board limiter is engaged to protect the output on the top end, and that mix then feeds the wired IEM system.

I have two samples from tunes to listen to for the electric guitar artist. This first one is from “My Savior, My God”. The instrumentation is electric, acoustic, bass, drums, lead vocal, and BGV. It sounds a little cliche, but you’ll quickly see as we go through this series that each person’s mix probably starts with a “more me” philosophy. This makes sense when you remember that I strongly encourage all of our artists to wear both ears all the time and this results in lots of isolation from the outside world in what they’re hearing. The result is a big need to hear what the artist is doing individually, and then surround that with the other instruments to provide timing, pitch, and ensemble. The band is not playing to a click in this one so you’ll see where our artist placed the drums in relation to everything else. I think it is interesting how out front his guitar really is in his mix, yet you can clearly maintain the placement of the kit, vocal, and the worship leader’s acoustic.

DISCLAIMER – this obviously isn’t the only way to build an IEM mix but hopefully it is helpful to hear what a really good musician is listening to.

The second sample is a little different in the band is playing to a click & track. At the start of the tune, you’ll hear the count off and a really broken down intro. The electric is just doing atmosphere stuff at the beginning and really kicks in at 1:12. The interesting thing here is the difference in placement between the click and the drum kit in this one. As I listen to the mixes that most of our guys dial in, it seems when playing to a click, everyone really dials into that click and keeps it out front with the kit tucking in behind it. This might just be a preference thing but it works for our guys since everyone is so used to playing to a click. Obviously your mileage may vary.

If you have any questions based on what you’re hearing, feel free to use the comments on this post for discussion.

Next time: IEM mixes for bass

One of the biggest questions and challenges I hear about from other audio guys, especially in churches, has to do with IEM mixes – what should be in them, what do good ones sound like, how do you help your artists have successful mixes, etc. etc.

Today we’re going to start a little journey over the next 2 weeks to cover each major instrument – bass, electric, keys, drums, and vocals – and talk about what makes good mixes for each category.  I also have actual samples that I have recorded so you can hear what real guys dialed in and we’ll analyze each one.  I’ll be posting Mon-Wed-Fri this week and next to continue this little series.

To start things off, I want to refer you to a video blog I did a few months ago about our PQ system that might be helpful so you can see more about how our particular system functions for our artists, just for context.

Next, a tool that is REALLY critical to good IEM mixes is what we call our “extras channel”.  A common complaint you’ll hear from artists who are first transitioning to IEM is that it can feel really disconnecting and isolating.  All of a sudden, guys hear themselves in glaring detail – for good and for bad.  If the tone is aweful, it’s much easier to ignore it when its just coming from an amp at your feet or a wedge.  When it’s in your ear, there’s no escaping it.  The biggest trick to making the ears feel real and less isolating is effective use of ambience mics.  I’ve written about this before, but we have three sets of ambience mics that I use as the primary foundation of our “extras channel” – a set of shotguns on the side of the stage, a set of PCC mics on the 1/3 and 2/3 lines of our stage lip, and a set of small diaphragm condensors hanging over the crowd about half way back.  These mics are mixed on a stereo aux that is fed to the IEM.

The “extras channel” allows the artist to decide how isolated they want to feel when they’re wearing their IEMs.  It is really important that this mix be stereo so it will feel right.  A couple of our guys choose to have mono extras just because they want to use one of their PQ channels for something else, but for the majority of artists I don’t recommend letting them go mono.  Stereo is really important.

Along with the ambience mics, I also mix post fader sources for anything else the artist might need when they AREN’T playing so they can remain connected to what’s going on – speaker and greeter mics, video playback sources that might role while they’re waiting to play, and talkback from the stage manager so they can hear rehearsal requests and communicate back and forth.

Here’s a sample of what the “extras channel” sounds like when we isolate it all by itself during a worship tune.  It really doesn’t sound like much on its own, but the presence that this adds to the mix has done wonders for IEM acceptance at Kensington.  If you don’t have something like this in your ear mixes, you don’t know what you’re missing!

Next time:  electric guitar IEM mix

Today’s post is inspired by content I recently watched on Ownthemix.com (are you a member there yet??).  I’ve realized this is one of those areas where there’s lots of things I just do instinctively without really thinking about them in order to ensure smooth transitions between the elements I’m mixing of a service, band set, event, etc.  I’m going to try to outline some thoughts that I’m planning to share with my team the next time we’re together talking audio.

A primary responsibility of mixing audio well in a church is to do everything possible to eliminate distractions.  One of the best ways to do this is by having great transitions.  Transitions make a good audio guy into a great one.

• Don’t just mute and unmute inputs.  Always try to fade into and out of elements.  Make them smooth and seamless so they flow together.  If the mic starts feeding back, you can catch it before it gets out of control.  You don’t ever want to just turn on an input full volume – who knows if it will be louder then expected, crackle, or have feedback.
• Always turn on a communicator’s mic right on time.  Too early and you might catch a sideways conversation with someone in the front row or clearing the throat.  Too late and you miss the first few words.  It’s a simple thing, but earn the trust of your communicator by having his back.
• If the communicator is going to have the audience talk with each other for a moment and then they’re going to continue speaking, you should bring their mic back a bit (maybe 5-10 dB) while the speaker isn’t “on” so they can take a drink or speak to someone without disrupting the moment.  But keep your eyes closely on the communicator so you don’t miss when they are ready to proceed.  Again, have the speaker’s back so the mic is transparent to them.
• The worst thing you can do is to forget to turn the speaker’s mic off when they are off stage.  The music starts and you hear the speaker singing – this can be very embarrassing for said speaker.
• Slow, intentional fades are MUCH better then quick mutes and unmutes.

I recently ran across the work of another audio engineering legend that is worth checking out.  The names of the people he has worked with are too many to list, but when one mentions musicians like Count Basie, Duke Ellington, Oscar Peterson, Sarah Vaughan, Eddie Harris, Quincy Jones, Jennifer Lopez, and even Michael Jackson, a great deal is immediately understood.

Mr Swedien wrote a book in 2004 called Make Mine Music that gives away detailed information from his lifetime in the studio-from a musical, technical, and very personal perspective. This book has something for everyone who is interested in music, especially those curious about the stories behind the scenes of some of the best music to ever come out of the recording studio.  I came across the book in 2005 but had forgotten I even owned it.  Upon rediscovery, there are too many pearls of audio wisdom in this book to list.

Being such a fan of Bruce, it was with great interest that I found this snippet a few weeks ago on ProSoundWeb from the book that is solid content for anyone who practices this artform of mixing modern music.  Here Bruce writes about developing your own “audio personality” for how you evaluate what you hear and translate it into an actual mix “product”.

Do yourself a favor and check this out. If you’re like me and the article resounds with you, pick up the book.  I think you’ll be glad you did!

Continuing the discussion from Friday, here is part 2 of a cool thread I ran across recently on one of the discussion forums about the unwritten rules for mixing monitors that they don’t teach you at Full Sail.  This is all the kinds of things I think many of us take for granted but will take a growing engineer up several levels by mastering.  Credit goes to everyone who contributed to the thread!

1. This is mostly personal preference, but I like having my cue wedge on the ground just like the performers have it. Some people put them up high on cases, but I think I should be hearing it exactly like the performer does, so mine goes on the ground.
2. Keep your eyes open: sometimes you can ‘see’ where feedback comes from and fix it easier than trying to guess where it is coming from…
3. For festivals I sharpie and board tape “You’re sound guy’s name is __________________” on the monitors.
4. For tours I sharpie and board tape “You’re playing in ________________” on the monitors.
5. Get the band to work on the best stage balance they can before you add wedges, this can save a lot of grief and makes the job a whole lot easier. I know of at least one band (and have heard of others) who taken to moving back-line offstage (usually to the wings) to keep levels sensible.
6. Listen to what the artist asks for in their mix, then give them what they need. If you get a good working balance, with good musicians you shouldn’t have to push faders around much as they will be able to make adjustments themselves in their own dynamics. This has the added benefit of the artists being able to communicate better, musically, and should lead to a much better vibe on stage, hopefully leading to a great gig. They feel good, they are happy with the monitors because they were able to play well, you get the credit, win, win , win!
7. Amateur vocalist that have problems staying on the mic and pulling away, lowering their level in the monitors can help get them the on top of mic better in an effort to hear themselves. From time to time I’ll get vocalists that will back off the mic when the monitors are hot, then complain they can’t hear themselves. Then the vicious circle of upping the gain, hacking the EQ, them backing off the mic more may ensue. Dropping their level all the time may not help, but it often can when they are afraid hot monitors whilst on on top of the mic.
8. I also put “Mix # _____” on the tape, for the artist to see. It helps them in sound checks feel more involved and communication is good.
9. I think a lot of the discussion in this thread is saying “get the band on your side early into the game by serving them with a pro attitude.”  If there’s ever a situation where they’re just not happy in the check, I always go out on the deck and listen next to them, rather than just sit on my butt and use the listen wedge. This is usually a good thing to do for both tech shooting and gaining trust with the band. “Can do” is always the best approach.
10. When troubleshooting a problem such as no sound from a channel or mix, don’t turn it up before isolating the problem. When you do find the problem you could end up with horrible feedback until you can get to the trim or mix level.

I came across a cool thread recently on one of the discussion forums about the unwritten rules for mixing monitors that they don’t teach you at Full Sail (I added that last part!).  This is all the kinds of things I think many of us take for granted but will take a growing engineer up several levels by mastering.  Part 1 today, part 2 on Monday.  Credit goes to everyone who contributed to the thread!

1. Never put anything in someone’s wedge mix they didnt ask for.
2. Never change a mix after the third song, unless its a ballad or something rehearsed in soundcheck.
3. The best soundcheck adjustment can be moving a monitor slightly or changing the angle using a short 2×4.
4. Know what your wedges and IEM’s sound like. Know what wedges sound like solo, in pairs, and with and without the foh rig on.
5. Every stage is different but low end and especially low mids build up fast with 6 or 8 wedges all blasting away. Once you start adding a vocal to 2 or 3 different mixes you may have to start dialing back more low end than you did when you tuned one wedge with one vocal in it.
6. Be responsive. Sometimes on crowded stages this is hard to do when every musician seems to have wishes at once.
7. Don’t rely on your cue wedge all the time. Listen on stage whenever you can during soundcheck time so you can hear the actual mix, wegde and stagenoise combined on the musicians spot.
8. When the stage is loud and the singer wants more of his/her vocal in the wedge, and you’ve reached the point where this can’t be easily done, you can often subtract competing instruments/vocal from the mix and solve the problem.
9. Unless it is specified on a rider, number your mixes from downstage to upstage, and from stage right to stage left. This is consistent with how you would see it from FOH. It should also match the way vocals are typically laid out too. It doesn’t matter which side of the stage you are on, keep the mixes the same way. Usually the drum mix is the last mix in line.  A typical 4 piece band would be like this. Downstage right is mix 1. Center vocal is mix 2. Downstage left is mix 3. Drums are mix 4. You will find that this numbering scheme has become a de facto standard in the SR world and engineers and bands alike will appreciate the consistency.
10. Take care of your ears. The stage is a very, very loud place to be. Add in the fact that you have to listen to a cue wedge at pretty extreme volumes to get over the stage wash only makes it worse. I usually mix 2-3 songs off the wedge, then pop in my ear buds and use the headphone out on the console. This really helps you prevent ear fatigue, and it actually lets you here those tiny squeeks a little easier.

What are yours?  Add them in the comments section.

In the past few days I’ve come across a resource for church audio that I think everyone associated with house of worship markets should know about called OwnTheMix.com. I’ve never seen anything like this so please consider this a must investigate for your team!

From their website…OwnTheMix.com is a comprehensive audio training solution for the modern church. Created by Buckhead Church’s Audio Director, Dustin Whitt, and Music Director, Chris Green, you will learn secrets to creating irresistible worship environments. Whether novice or professional, there is plenty for everyone inside. Subscribers have an all-access pass to every video, and are connected to all of OwnTheMix.com’s users through a state-of-the-art social network that includes forums, blogs, and groups based on like interests or topics.  What makes this content unique is that there is an important balance between tech & music – they get the necessity of this unique partnership that is so critical to audio success in churches.

Training topics include basic EQ, compression, and gating for bass, electrics, keys, drums, & vocals, miking instruments, combat mixing, console gain structure, basic drum tuning for engineers, mixing musically, EQ and mic placement for speech, transitions, working with musicians, working with drummers, and much more.  There’s also training content that currently focuses on the Digidesign Venue platform, although videos are coming soon that will also train on Aviom & the Yamaha M7CL.  Finally there are several multi-part interviews with other members of the Northpoint audio team such as Dave Stagl and Chris Briley, and notable FOH engineers such as Jeff Sandstrom (Chris Tomlin).

A 1-year subscription starts at $200 for a single seat and the price per seat goes down as you add more users from your organization to the site.  The really cool thing is that if they were to put an accurate price tag on what is taught in this site, it wouldn’t be affordable for many churches.  The value is extremely high!  If you paid a contract soundguy $50 an hour to run your Sunday services just for one morning, you’ve paid for an entire year of training for a committed volunteer. The site really is one-of-a-kind and content is still being added weekly.

I’m proud to say Kensington Church has secured a license for our entire audio team and I will be leading our team through this content over the coming year.  It has been on my heart for months to find a way to focus on basic audio concepts and develop more of a culture of organized continuing education for the team but had yet to find a means that would be the right fit.

I showcased a new book I picked up this past winter called “Mixing Audio” on this blog that is the best resource I’ve ever read for the art of mixing music.  The problem is that a large percentage of our audio team are also campus tech directors or work full time jobs in other industries entirely, so I expected a tough sell to get everyone to read such a large, focused book and get together for discussions.

OwnTheMix is perfect for us because we can “assign” a handful of videos for guys to watch on their own time that are reasonably short for those with ADD attention spans (it seems most average 3-6 minutes) and pack lots of solid concepts in each one.  Then we’ll meet together every other week for an hour to unpack what we’re learning and discuss questions or applications.  I’m excited to see what God will do in our team over the coming months!

I can’t recommend this resource enough.  If you are involved in church audio in any way, RUN to this site and sign up for you and your team.  I’m certain you won’t be disappointed!

A few days ago I was introduced to the personal website for Michael Brauer.  I’m a big fan of his mixes – maybe you’ve heard a few:  Coldplay “Viva La Vida” & “Parachutes”, John Mayer “Continuum”, The Fray, Gavin Degraw, Ben Folds, & Matt Kearney.  It’s always especially interesting to find a well developed personal website for an engineer who’s work I enjoy.  Michael’s site has a rich Q&A/FAQ section and reprints of various articles featuring his work.  Definitely some good reads here.

Below are a few highlights of things in the Q&A section that struck a chord with me.

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Could you speak to the issue of how you achieve movement in a mix (or your philosophy about movement in a mix)? I read in one of your other posts of how you rode a pad in one of the Coldplay mixes. Do you generally ride all instruments/sources…or are there some sources that remain rather static and others source that your regularly “move”? If you can’t give a general opinion, perhaps you can comment on your philosophy in mixing with movement in mind. With the large amout of dynamics processing that today’s mixes go through, how important is movement in a mix when the song is going to get pulverized to 5 – 6 dB of dynamic range anyway?

The song will dictate everything that needs to happen. Not gear, not technique, not the “go to” button. It’s about the song and nothing but the song.

Dynamics are going to play a major role in giving the chorus the payoff it needs. I’m not necessarily referring to riding the stereo track up. I mean internal rides going into the chorus like riding the drums up on the last bar or riding up the first chord of the guitars. These are just tiny basic examples to get you started.

I’m riding a whole lotta faders during the course of a mix. I’m riding the vocal to drive the song, riding the bass, toms, cymbals…well pretty much anything that helps make the song come alive. I’m making the mix as animated as possible to get the message of the song across to the listener. Imagine you’re watching an action cartoon and that will be a good starting point for how dynamics work.

Of course, not all songs are going ballistic. I probably do more rides on a ballad than I might on a rocker. It’s the accumulation of many subtle rides that add up to an emotional mix. You can probably set a good level of a compressed string section in a chorus and just let it rip. I prefer to do internal rides within those strings and overall small crescendos of the group to accentuate the passage. I may very well compress them, but that may not be enough to do the section justice. The point is to add movement to the song in order to make it seem to come alive. You’ve got to ask yourself one question, “Do I feel lucky?, Well do ya punk?”…oops sorry, I mean, What rides can help a great hook?…punk.

There are no set rules for what stays static or doesn’t because every song is different, the recording is different, the parts are different, basically, everything is different. The point is to use dynamics to bring out the best a song has to offer. What can be done to make the story and the hook of a song burn into your brain forever.

Rides are an essential part of mixing a song to its full potential regardless of the amount of compression the mix is going to get hit with by the time it goes to radio. An emotional mix will help the song survive the squash. It’s all about the song. Repeat after me, it’s all about the song.

In your opinion, what do you think gives a snare drum stroke the ultimate catchy sound in a mix?

Thats a tough one to answer because there are so many variables.

For me every song is different. I decide early on how the snare should be placed and how important it is to a song. If the recorded snare doesn’t sound good in the the track I add one or a combination of samples to get the sound I’m looking for. I rarely replace the source kick or snare.

I like the snare to give the song’s midrange a brightness without interfering with the vocal. How bombastic, fat, funky, snappy, etc is determined by the style of music I’m mixing. If it’s funk, I’m not gonna give it a rock feel…unless it works. It’s always a work in progress.

The snare has over the years defined in what time period the song was released. Some snare sounds had a very short shelf life and dates a song so terribly that it’s hard to listen to. I decided many years ago to come up with snare sounds that would withstand the test of time. Many times, it’s the snare that gives the song it’s unique hit quality. Within my discography, Coldplay’s “Yellow” is probably a good example.

In general, i’m changing my snare sound every six months. As soon as I find one that is catchy, I drop it because I don’t want two different artists to share the same sound. It’s an easy temptation to just repeat the same snare sound but I think that’s lazy and eventually people would get bored by it, not to mention it gives every artist the same generic sound. That’s not the reason why artists want me mixing their record. It’s harder to accomplish but so what, it’s not as if I’ve got anything else to do.

Do you like the artist being present at the mixing session?

Hello? I’m not the artist, I’m just the mixer. It’s their vision, not mine that I want to mix. If the vision is unclear on a particular song, I can help.

The band has spent months making their record. They’ve been part of every decision, change, fight, and they’ve finally gotten their vision recorded and documented. The rough mixes are feeling great. And now for the most important part of the recording process, they aren’t allowed at the mix? Wrong, I’m not interested in mixing an album for myself. I want to know about each song including the story, the vision, the likes and dislikes of the rough mix, tracks that should be left off or changed, etc. For a new band it’s even more crucial because I want to help set up a sonic template that is unique to them that also sets them apart from other bands.

I make it very clear to the record company that if the band’s visionary or someone they completely trust to represent them isn’t at the mix, I’m not interested in mixing the record. There are very few exceptions to this rule. Sometimes the reasoning behind a record company’s request of “no artist attendance” may be valid, but most of the time I don’t buy it. I get the “they don’t know what they want”, “They’re too young to have a clue about the mixing process”, “they are such a pain in the ass that they’ll screw up the mixes”, “we don’t have the budget” and “they’re on tour.” In my mind, these reasons don’t qualify for them being excluded from the mix.

If the band agrees the producer or A&R has a better idea what is best for the band, then they should attend the session. If it’s a money issue, I can set up live Ednet sessions. The artist can be at a studio in London or LA and listen back to the mix in real time using the same speakers I’m monitoring. They can make comments via a talkback and they have visual via ichat . It works like a charm. I would prefer them to be in the same room, but this is a great alternative. And finally, I’m from NYC, nobody is going to want to be a pain in the ass for very long on my session.

It comes down to this: it’s their record, not mine. I have years of experience at their disposal. I have a short time to make their songs come to life and they have a lifetime to live with it.

Ok here we go…

The first sample is the snare and overheads.  I’m flipping the delay in and out on the snare channel every 4 beats.  The biggest place you’ll hear the difference is in the body of the drum.  The non aligned signal sounds really thin compared to the time aligned one.

The second sample is the entire kit with delay and without delay.  In order to make it easier to compare as you’d like, I included these as separate files.  First, the delayed and time aligned kit.

Second, the non aligned kit.

As you can see, maybe its subtle – I’m not sure how its translating across the web, MP3 compression, etc.  In my room, the difference is pretty huge when you’re trying to get great individual tones and make everything play nice together.

If you mix on a digital console, this post is REALLY important.  Unless you’ve been mixing under a rock, I’m sure you’ve heard about time aligning your drum mics.  I’ve been reading about others’ results with this for a long time but have never been faithful to time align every time I mix because most of the samples I’ve heard of before and after have been less than convincing.  My prejudice, and maybe yours as well if you’ve heard similar samples, is that time alignment on the console using channel delays is a 2% mix improvement step, not a 15% or larger mix thing.  Because of our intense pace at Kensington, it has been hard to have the discipline to time align every time.

When I was mixing on a PM1D every week, I made an effort each time to time align by ear and just slide back the snare, hat, and tom mics back to the overheads by anywhere from 2-6 milliseconds.  The difference was noticeable but still what I would consider a minor “glue” thing – certainly not earth-shattering.  When I moved to the Venue, I kind of set the time alignment stuff aside for a few months in order to focus on more core tasks and efficiencies.  A few months ago, I began to revisit time alignment and this time, for some reason, the results really were a game changer.  I’m not sure exactly what to attribute that to, but I can share some tips on the overall approach I’ve settled on that seems to give the best results.  If you haven’t tried this, I would suggest you should and see if it makes as big a difference for you as it did for me.

My goal in time aligning is so that the tone of a drum (let’s use the snare as an example) will sound nearly the same when listening to the close mic as when the overheads are added in.  What I found a few months ago in the run up to Easter was that it was bothering me to get a snare sound I liked with the close mic and by the time the overheads were where I wanted them that tone was destroyed.  Pull the overheads out and the snare tone would go back to where I wanted it.  What I found was that I was hearing a combination of overheads with too low of a high-pass filter setting and comb filtering between the close mic and overheads.

I should preface that there are two different philosophies for overheads on a drum kit.  The first is to place them strictly as cymbal mics.  For some music styles or drummers, this is necessary in order to get the expected mix results.  The other option is to use them as an overall kit mic that just so happens to have cymbals as a focus point.  In this scenario, the high-pass filter will likely be set lower than it would for cymbal micing applications – probably 100-200 hz instead of 500 hz plus for cymbals.  Another important part of the kit mic approach is using mics in a X-Y pattern over the kit.  I go back and forth between a Rode NT4 and Shure KSM141s on the Shure stereo adaptor bar.

The first step to the results I want is to use the overheads as kit mics and still set the high-pass around 250 hz.  This allows the overheads to still be the glue for the kit but the close mics to provide the “in your face” body of the sound.

Next, each mic is delayed back to the overheads.  I’ve found the easiest way for me to do this is to have the drummer give me a quarter notes on each drum and record that into PT.  (If you don’t have PT, instead of this technique, just go drum by drum and have the player give quarter notes while you dial each mic in)  Once in PT, I zoom in closely on those hits and measure the difference in time between each of the close mics and the overhead.  The results are usually somewhere between 40-160 samples.  (You could also do this with milliseconds but because PT & Venue can go to samples resolution, that’s what I use)  When you have it right, its the first time in my life that there is a negligible difference between the tone of the close mic and once you put the overheads in.

Next time I’ll post some samples that will hopefully convince you.  This topic has been written about all over the web so if I’ve piqued your interest, start digging into it and feel free to ask questions here.