Church EQ Curve

[wschill]

I've been doing audio for quite some time and work as an engineer. I'm trying to find a "standard" equalization curve to apply when equalizing the house system, and thusfar, I've not had much luck. When I've done the house EQ thusfar, I've used the SMTP X Curve not because it is necessarily right, but it's one I'm familiar with. I've been doing online searches, but haven't really came up with a standard for church equalization.

Any suggestions as to where to look?

[bladeaudio]

are you simply applying this curve to your EQ settings, or are you measuring the actual speaker output curve with pink noise, etc?

The X curve was developed for movie theaters, basically to enhance the low end. Might not be the best for a church where speech intelligibility is important.

[cmchamp]

You won't. Best practice is to voice each system to the room in which they are installed. Utilizing a DSP with as many parametric filters on the input and each output is best. Presently, I'm using VTC (Yorkville) DMLS4080's.

[Brad Weber]

I'm alos not clear on whether you are referencing a target response or an equalization to apply. Even for response there are no 'standards' as it depends on the application, although there may be some general response characteristics that many people look at for certain applications.

[wschill]

I guess I should have been clearer in my post. When I've done the EQ, I've used pink noise set to a level of 85 db. I'm injecting it into my board as a line level signal with the parametric in bypass for the line level signal.

I've started by setting the microphone roughly in the center, eq'ing the front speakers, moving to roughly the 2/3 location in back and eq'ing the rear speakers, then putting the microphone roughly in between the two spots and checking with both front and rear on. I'd love to use a multiplexing microphone system, but alas, I don't have one.

My goal is mainly to get the "system" eq'ed appropriately with the 1/3 octave equalizers and then adjust the mixing board sound to fit the need.

With the setup I've done, generally video playback and oral speech sound good, but we have some real problems with the praise band. Choir also can sound decent when the vocalists project properly.

[cmchamp]

how are the front speakers arranged? Center cluster? L/R split?
When I voice, I target each cabinet at once and seldom use a 31 band anymore for mains.
I use a stereo FFT application from Faber Acoustics for the Mac called Signal Scope. I'll tap my pink noise source on the direct out of the channel, then feed my measurement microphone to the other channel to compare the input slope and result to keep as close to the slope as possible.
After getting "flat" as possible, I then move around the room, listen, listen, listen and adjust as necessary.
For a performance venue I work in regularly, I spent 3, 4 hour afternoons voicing the system, and come hot weather, I'll do it again.

[waynehoskins]

What do you mean rear speakers? Tell more about those.

In general, you want close to a flat response. How you get there varies widely between system engineers. Some pink the room, some SIM the room, others listen to pink noise, others listen to music. Often a combination is used. In the end, you do end up using your ears, to verify that it does sound "right", whatever that actually means at the time -- and in the end, it's usually not quite flat.

As a starting point, if I can find them, I'll put in either recommended tunings for the loudspeakers, or if those aren't published, I'll guess at the corrective tunings from the response curves. I'll also throw in the first-guess time alignment values there too, before I even run any sound through them. Rarely do I have to deal with room mode problems, so usually loudspeaker-corrective tunings are about all it takes.

[Brad Weber]

Originally Posted by wschill

I guess I should have been clearer in my post. When I've done the EQ, I've used pink noise set to a level of 85 db. I'm injecting it into my board as a line level signal with the parametric in bypass for the line level signal.

I've started by setting the microphone roughly in the center, eq'ing the front speakers, moving to roughly the 2/3 location in back and eq'ing the rear speakers, then putting the microphone roughly in between the two spots and checking with both front and rear on. I'd love to use a multiplexing microphone system, but alas, I don't have one.

My goal is mainly to get the "system" eq'ed appropriately with the 1/3 octave equalizers and then adjust the mixing board sound to fit the need.

X-Curve, 85dBA, rear speakers, EQ only... I'd guess your background is cinema or home theater. If so, you might start by forgetting much of that as the goals with system tuning in live sound are very different.

In the cinema/home theatre world it is all about having a specific, defined application and system performance so that what is mixed during production translates well to not only that space but to every space. In live sound it is all about that particular application and space. Aspects such as gain before feedback, intelligibility, subjective preferences and so on become relevant and may require different approaches or solutions for each unique situation.

Tuning a system typically has some subjective component in deciding what adjustments provide the best result for that application. One aspect of that is the content. The response you want for a speech focused application may be quite different from that desired for some music focused applications. Another aspect is that the system performance will likely vary throughout the listener area and what is the 'best fit' result is often a subjective evaluation of the system at multiple locations in the listener area. So you might want to consider looking at much more than just in the center as you could have very different responses at different locations and adjusting for just the center of the room could be negatively impacting other locations.

A few other thoughts include EQing at levels that represent typical use. Learning about and potentially measuring and addressing aspects such as intelligibility and relative phase. Room acoustics and the effects on audio (most cinemas and home theatres try to minimize the impact the room has acoustically, that is not the case for many churches). Speaker and microphone patterns and their impact. However the most important aspect is listening. RTAs, dual channel FFTs, Impulse Responses, Transfer Functions, etc. are wonderful tools to assist in getting a desired result and/or in determining why you hear something, but they cannot tell you what sounds good.

[wschill]

I'll try and explain our setup a bit better. Our church is a typical, 1950's Methodist church. It's dimensions are approximately 120' long by 40' wide with a ceiling of maybe 15 - 18 feet.

The front of the church has two main speakers, one centered in front of the left pews and one centered over the right pews on the first ceiling beam (not sure if that is the correct architectural term or not). The front beam roughly corresponds with the edge of the altar. These speakers are pointed towards the pews.

The choir loft is mainly behind the beam. However, on the left side, the choir loft has 1 row in front of the beam, and hanging in front of the speakers are choir microphones hanging from the ceiling. We have tremendous problems with feedback from them, as, you guessed it, the left speaker shoots directly into the microphone.

About 40 foot back or so, there is a second set of speakers which aim towards the back of the church, also centered L-R over the pews (one on the left; one on the right). These are smaller fill speakers which have terrible response relative to the front speakers.

There is a sub placed in the organ cabinet on the front right of the church.

The sound mixer is in the right rear corner.

The problem I constantly run into is feedback from the front speakers. I've found that if I want to avoid feedback, I need to drop their level about 5 db relative to the rear speakers. But, if I do that, then I get complaints about echo from the rear speakers off of the rear wall.

[cmchamp]

Your rear speakers are "delays" and should not have the same response as the front. The purpose of fills is to reinforce intelligibility at frequencies above @250Hz or so, not to provide full range. If they are not on a delay, they should be.
As long as you've got a speaker pointing into an open mic with that mic being fed back through that speaker, you'll always be fighting the feedback issue.

Question - is the speaker in the choir loft for monitor purposes? If so, then the choir mics should not be fed to it, only content required for singing to playback material or other instrument sources not easily heard from that position.
With the sub in the organ cabinet, if there is a speaker processor, does it provide delay? If not, one needs to be installed that will provide proper delay settings to align your cabinets and increase intelligibility.

[wschill]

Originally Posted by cmchamp

Your rear speakers are "delays" and should not have the same response as the front. The purpose of fills is to reinforce intelligibility at frequencies above @250Hz or so, not to provide full range. If they are not on a delay, they should be.
As long as you've got a speaker pointing into an open mic with that mic being fed back through that speaker, you'll always be fighting the feedback issue.

Question - is the speaker in the choir loft for monitor purposes? If so, then the choir mics should not be fed to it, only content required for singing to playback material or other instrument sources not easily heard from that position.
With the sub in the organ cabinet, if there is a speaker processor, does it provide delay? If not, one needs to be installed that will provide proper delay settings to align your cabinets and increase intelligibility.

Rear speakers are delayed via the dbx Driverack 260. We feed the output of the mixer into the driverack, separate out a front, rear "or delayed" speaker set, and a feed for the sub. The sub, if memory serves me right, is driven with a 125 Hz lpf. The outputs of the driverack (except for the sub) goes through a 2231 before going to the main power amps.

In terms of delays, I have the mains set for 0 delay, the sub set for 0 delay, and the rears set for around a 30' delay. I started with a distance measurement and then adjusted with a cd click track until I could walk around the sanctuary and not hear any audible "double clicks".

The front main speaker, on the left, is actually the main speaker for the left side of the church. What happened is the church had many remodels done, one of which moved the choir loft forward. When the new system was installed (prior to my arrival) the mics and speakers were just hung without any regards for each other. Given our setup, I'd almost rather have the mains back about 15 feet on the next beam and a pair of downward facing fills installed in the very front to cover the first couple of rows of pews. I think that would help tremendously (but's also a lot more work than I want to get into right now...)

As for the sub, it's currently not on a delay, as the organ cabinet is actually in the front right of the church (opposite side of the altar) right next to where the minister speaks.

[cmchamp]

The 2231 is really redundant, the DR260 should have enough EQ functionality to take care of sub, mains & delay EQ processing.