Nuts and bolts of PA tuning
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…
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Introduction
My name is Tim Corder. I started this blog in February 2007 because there were so few of what I considered good church audio resources available at the time for my team. Fast forward over 5 years and I'm still at it, sharing learnings about the journey towards making audio great. I go through periods where I post a lot and other times when I don't. I'm thankful for the opportunity to share it all with you. Thanks for visiting! 
OK, Tim, can you please clarify a couple things for me because I’m confused. In your last post you mentioned that you want the loudspeakers to sound like the outputs of you console. Now you’re saying you’re not looking to create a “flat” PA. What do you consider to be a “flat” PA? In my world a “flat” PA is flat on an FFT and what I might also call a “linear” PA where what we put into the loudspeakers comes out sounding like it did going in.
While I think Fletcher-Munson curves are good to understand, they are a natural part of our hearing perception. Studio control room monitors aren’t voiced for Fletcher-Munson. Why would you do it on a loudspeaker system?
Thanks!!
Hahaha…
Just talked about this today with Bob & Jeremy from Moyers Group while we were hanging out today…
Here’s the thing: to me, I want a “good sounding” mix recorded to an offline recorder, broadcast on the internet, or sent to DVD to also sound good in the room. I’ve found that if I create a “flat”, 100% linear PA without any tonal shaping there to accommodate what is going to happen when I turn it up loud (especially in the 2k-6k range), instead relying on the console for all tonal controls, the result in the room still works great but the mix to recording is usually missing life in the 2k-6k range. I attribute this to laws of human hearing and Fletcher-Munson is helpful in understanding the why.
If, on the other hand, I shade the PA just a bit (depending on the room and speakers I’ve done as much as 4 or 5 dB at the deepest point), the room mix still sounds great and I don’t have to do as much channel EQ so the recording has more guts. You have to be really careful here because moderation is key, as with anything else. Over a few weeks of reviewing a mix in the room and the same mix captured to a recording, I’ve never had too difficult of a time dialing this in to the point where it sounds great in both places.
Again, the Fletcher Munson thing helps me understand what is going on here because that same mix coming out of the console is normally being listened to at two completely different levels inside the room live and at home on a DVD or the internet. At least that seems to be the best explanation. I’m ok if things are a bit more gutsy in another environment because the listener normally has control over the level with which they listen. In the big space, they are at my mercy and I want it to be comfortable and rich.
This is where the listening to the tunes thing comes into play as a test of room tuning. I can listen to a track that I know sounds great in headphones, monitors, or home theater world, compare that to how it sounds in the room through a flat channel strip, and once it also sounds/feels great in the live room through the PA, I know we’re going to have a linear transfer of the mix from one medium to the next. Maybe that is what I should qualify this as – I guess I’m not quite as concerned with perfect linear transfer from the console through the PA as I am a linear transfer of the mix from the console through the PA. What sounds great in the room needs to also sound great on a recording. I do not have the luxury of a broadcast mix position and will add monitor world before even considering it.
Now Bob’s push back was that whatever is lost in the console mix because the PA was “flat”/linear across 2k-6k could just be added back in to the outside world feed post-console further downstream before it hits his destination and he’s right. That is certainly another option. I’ve just found a repeatable process that has now worked in two completely different spaces with two very different signal chains and figured it was worth sharing. The end result once all of this hits the magic spot is a great mix in the room and a great mix on a recording.
I also have a difficult time putting things back into the world mix that I took out at the console because of the room/PA, needing another processor to look after the world feed, and now having to manage two different parts of a signal chain. Especially in Venue world, my dream was to be able to do everything broadcast related inside the FOH desk so it was stored as a part of the show file and easily repeatable week to week or even duplicating in a different venue by loading a file and adjusting the gains on some room mics to suit. One day we’ll probably have the same kind of system here at FOH and so I’m trying to set the processes in place with that end in mind, even if it means the bus compression I hear on my current world mix out of the M7CL leaves a bit to be desired. It will get better with time (or a different desk).
I’m really interested in your counterpoint to this
Well, I think you need to be careful with terminology because what you are doing isn’t a “linear” PA; sounds like you’re working towards having a mix that translates to different mediums and environments which I think is something that is only going to get more important as the number of different ways we are going to be distributing our service content continues to grow.
I think your way of doing this is certainly one approach, and something I’ve thought about doing where you’re essentially equalizing your PA to force you into making certain tonal decisions on inputs that will help those inputs translate. I would put this approach much closer to Dave Rat’s approach to optimizing his rigs. Although, in the case of Dave Rat, seems like he is doing his mixes in a different environment and then modifying the PA to have those mixes translate to the PA.
I guess the only thing I would wonder about this approach is what happens to programming material that happens at a lower level in your room. If you’re forcing yourself to bump your music inputs to get more in that 2-6k range, it seems when you’re doing lower level content you would have the opposite challenge. The impact is probably going to depend on the loudness difference between programming, but I think it’s very hard in a church sound reinforcement setting to have one fix for the mix translation issue with all the different content we do these days. Whether it’s your approach or the downstream approach Bob suggested, it’s still going to have varying degrees of impact on everything you do; that’s what makes the whole webcast/broadcast thing such a challenge.
For me, I think the linear PA is still the best approach to system optimization especially in my present situation because I want linear operation in my room for non-music content which really takes up the bulk of our services. The reality is the volumes we listen to the non-music portions of the service at are very very close to the volumes we listen to them at in halls and cars and TVs so content in those cases will have a much higher level of translation.
The music side is another animal, and I’m still experimenting with ways of getting it to translate, but I think I’m pretty close to something that probably doesn’t require a $50k+ console to do it.