West Rehearsal Home Movie - August 2008 from David Stagl on Vimeo.

Within this video, there is footage that answers the question I’ve been asked more than any question.

Photo by Stephen Hunton

There were requests a while back to talk about bass guitar in the same fashion I’ve talked about vocals and drums. This isn’t really that post because I need to tap Briley for that one as he’s still my biggest mentor when it comes to mixing bass. But in the meantime, I tried something new last week that I kind of liked and is pretty easy to try if you’re looking for something new to play with.

These tips come courtesy of Jayce Fincher who is the Tech Director for the Attic at North Point. The Attic is where our high school and middle school kids get together and make a bunch of noise. Before Jayce ended up at North Point he used to play bass in a little band called Marvelous 3, and a couple weeks ago I walked into a conversation between Jayce and one of our FOH volunteers on mixing bass guitar. Jayce was talking about the challenge of getting the note of the instrument to cut which is something I struggle with. Jayce had a couple of pretty basic tips for helping to get the bass note to sit right without the whole instrument turning into a washy mess of low end.

Start by leaving the bass guitar out of the subs when you initially put it in the mix. This is most easily accomplished if your subs are aux-fed. If your subs aren’t on an aux, you can achieve almost the same effect by either turning off your subs or utilizing a high pass filter or a low shelf on the EQ. However you do it, the goal is to get all of that head-y low end stuff around 100 Hz and the rest of the sub-bass pulled out. Once you’ve got all of that out, it is a lot easier to find the note and get it defined. Start by cleaning up any remaining mud down low, and then go looking for the note if it’s still missing. Jayce recommended starting to boost around 500 Hz, but I think you could find what you’re looking for going up to 700 and even as high as 1k; I think it’s going to depend a lot on the instrument and player and what’s happening with the rest of your mix. Once you have the bass guitar sitting in your mix nice, start dialing it back into your subs until you get the right feel you like back without overwhelming the rest of the mix.

When I played with this last weekend, I definitely liked it, but I still found myself using a little of the “dirty” bass trick that Briley and a guy named Scovill like. Basically I take the bass guitar and double patch it on the Venue into two channels. One channel is the “clean” channel which I stick a compressor on and EQ in a traditional fashion. The second channel is the “dirty” channel which additionally gets a SansAmp plugin; the SansAmp gives it some grit and nice harmonics which help it cut a bit. I’ll then blend the two to taste with the dirty channel giving me more “note” and the clean channel giving me more “I’m not happy if I can’t feel it” low end.

I would love to hear any other tricks some of you guys have on getting the note of the bass to cut without overwhelming your mix with low end. The comments are open.

Here is the awaited demonstration of what I’ve been doing with time alignment on drums. The only caveat I have for you is to make sure you listen on a decent playback system. When I did this in the studio the results were clear, and when we listened back on our PA the results were clear, but when I listen on my MacBook Pro speakers it’s not so clear. So I guess if your sound system is a bunch of MacBook Pro speakers, you’re not going to get much out of this.

This is a quick demo so it’s only going to demonstrate the results of two mics, but it will give you an idea of what this does. Basically in the demo you’re going to hear the close mic on the snare along and the overhead with zero EQ on the inputs. You’ll hear each mic on its own and then a combination of the two. Finally I apply delay to the snare mic to time align it to the overhead by delaying it to the arrival of the snare in the overhead.

A couple of quick things I forgot to mention in the demo. When I play the 2 sources together with alignment, at the end of the sample the time alignment is turned on for the ENTIRE buildup and fill, but not the pre-chorus assuming you can tell where the pre-chorus is. I also mention that the delay used is about 110 samples. This is about 2.5ms. If you’re going to try this you probably don’t need to go all the way to the sample level. Since I do all my delay measurements in Pro Tools it’s very easy to work at the sample level, and I figure the closer I can get with initial measurements and settings, the better off I’ll be due to any shifting mics through the set from vibrating risers. However, you could probably get the same measurements by measuring the distance of your source to the two microphones and then using the difference to delay the closest to the furthest; remember 1 foot is approximately 1 millisecond of time.

timealignment-demo.mp3

UPDATE

Here are some drums in context of a band mix. NOTE: This isn’t a FOH board mix. This is more of a “broadcast” style mix that makes use of the same alignment techniques.

So here’s one for the rest of you system optimization junkies like me. I’m trying to solve a dilemma so I don’t have a lot of answers here, and I’d love some input via the comments section. This is just something I’ve been thinking about a lot lately. The basic issue is that I haven’t been satisfied with the way our audio for video has been translating in our big room when it has primarily been mixed in little rooms, and I’m wondering if there might be a better way to optimize our systems for this. This might ramble a bit so bear with me.

The whole thought process really started when I was talking with Tom Petty’s FOH guy, Stewart Bennett. As I’ve been building my optimization chops using an FFT such as SmaartLive, I’ve been optimizing systems to get a linear response. For those new to optimization and without going into the nitty-gritty on what FFT’s are, this basically means I’m optimizing my system so that that what we hear in the room is a linear representation of what the console is spitting out; this way everything we do on the console translates to the system giving the FOH engineer complete control at the console. This is the method that Robert Scovill is a big proponent of, and I would say I’ve had a lot of success with it. However, I’m still wondering if there could be a different approach worth trying even though everything we mix live in the room meets our expectations of our systems in their current state.

When I got to hang out with Stewart Bennett, we talked about a lot of different optimization curves that some other heavy-hitter FOH guys like to employ on their systems. Without going into details, some guys essentially make some sort of compensation for Equal Loudness contours which almost always entails some kind of cut per octave happening above 1k. Some of these approaches were also shared by some of my church audio geek friends. I am always up for experimenting, but there is definitely an element right now of “it ain’t broke” in the system so my plan was to hold off until we upgrade before I start experimenting again.

So fast forward a bit to this whole audio-for-video translation issue. Thinking about a different approach to system tuning got me thinking about something called the X Curve which is an optimization curve that movie theaters use. You can do some research of your own if you want to find out about it, but the X Curve basically looks eerily similar to some of the curves described in my discussions with other sound reinforcement guys.

Thinking about the whole crux of what we do, sound reinforcement in a modern church can really be a unique venture compared to a lot of other environments. I don’t know about your church but within one event or service ours can be required to meet the needs of a rock concert, movie theater, and college lecture in under 30 minutes. Just within these 3 elements are 3 different SPL level needs with almost 30 dB of dynamic range between the average programming SPL’s. In an effort to create a live music experience on par with a concert experience, a lot of emphasis these days gets put into optimizing systems for this, but I’m wondering if the rest of what we do might be suffering as a result and if there’s a way to compromise optimization to create the best experience for everything we do. Ultimately, my goal is to get things that are mixed in the big room for a service or event to translate to a smaller medium such as a video control room or hallway, and then vice versa. And while there is a lot of discussion on the net about room size and the psych-acoustics involved along with the difference between mixing on nearfield monitors vs. mid or far-field as it pertains to the post world, nobody seems to want to stamp a systematic approach to setting up for this.

So here are some questions for discussion from some of you other guys who like to play with this sort of thing and have more experience in this area than me. Is there a one-size-fits-all optimization curve that you like to use to get all programming types in the ballpark? Is it better to optimize the system to be linear(flat) and then to make adjustments to individual programming? What have you found works best in your rooms? Does the dynamic nature of what we do wihtin a church service make mix translation outside of our bigger rooms ultimately unrealistic?

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