Audio for Video revisited

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…

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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.

6 Comments

  1. Dave Stagl says:

    Good stuff, Tim. I’m balancing things out in a matrix, too, but an 8 dB bump on speech isn’t enough in my world to get it close to music, I need more like 15-20.

    What do you do about videos?

  2. timcorder says:

    I’ve found that in my applications, 15-20 dB would be necessary to make the wavelengths of music vs speech appear the same size in a sound editor, but I’ve always found 6-8 dB to be the right perceived amount to correct the difference when listening to the mix so you don’t reach for the volume control between elements.

    I remember dealing with video playback levels before once upon a time at Kensington but for whatever reason it hasn’t been an issue yet at newlifechurch.tv. What has been an adjustment for capture has been the addition of audience mics during video playback. The primary purpose of our capture is for people watching the service outside of the room so I’m ok with the feed being “live”. I’ve yet to find a reliable, invisible process that would duck the audience mics automatically inside the console during playback so it is a necessary evil for the huge wins during music and teaching (90% or more of our service).

  3. Ross Harris says:

    This is a really great post. Since I mix on an M7, I’m always trying to ‘fake’ the Venue workflow if possible. I have one question: How are you combining the ‘PA’ and ‘World’ sources back together? Are you just assigning both to the same omni output OR are you feeding them to an aux that outputs to the source? Thanks!

  4. Jeff says:

    Tim,

    Great ideas! I’m going to dig into my system routing to see if I can implement something like this. We rent, and share our system with the school we rent from, so making changes is a strange animal, but it may be doable transparently. Hmmmmmm… Where to find a pair of shotguns…

  5. Dave Stagl says:

    How loud are you running music in your room and how loud are you running speech? I think that’s going to play a lot into how you level things out. 15-20 dB is necessary for us due to the loudness difference between music and speech in our room which is typically greater than 20 dB.

  6. timcorder says:

    Music is in the low 90′s dB(A), speech in the low 70′s. Almost the same balance I had at Kensington so that 6-8 dB difference on the world feed seems to carry over the same. I’ve tried making them match each other level wise on a meter or DAW (which is more like 15-20 dB difference) but then I find the speech is way too loud in context on headphones or living room speakers. 6-8 dB has been about right for me.

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