Archive for the ‘Sound System Basics’ Category

(Some) Church Sound System Essentials Part 3

Tuesday, April 15th, 2008

In this installment I’m going to talk about house (main) EQ. That is, having (ideally) a 31 band EQ somewhere between your console and your power amps. For instance, my signal chain goes like this; Mackie SR32.4 console, right main out goes to a Rane 31 band EQ, then a Rane AC23 active crossover, then to a QSC MX2000A power amp. The subwoofer signal path is similar.

Rane 31 Band EQ

I think most will agree that the purpose of the EQ is to match, or optimize your speaker response for your room to some sort of standard that yields a consistent and predictable behavior as far as frequency to amplitude relationships. What that means is, every speaker has some “personality”, it has it’s own frequency response biases (which can be a selling point for integrated powered speaker systems such as those from Mackie.) Every room has it’s own resonances; peaks “nodes” and nulls – wavelengths that are canceled resulting in a dip in response. Add those together and you do not have a predictable, consistent response in your system. If you don’t have a main EQ in your system, ask yourself these questions: Do you make similar EQ adjustments using the channel EQ on several mics to make them sound “right”, do you find that voices are consistently very dull, or very bright, do you find that most acoustic guitars sound boomy, or thin, or bass guitars have no meat to them. These are all possible symptoms of an improperly tuned system. Enter pink noise (and/or a whole host of other optimization tools/techniques) and the graphic EQ. Pink noise offers a consistent noise source that can be monitored mathematically by a calibrated mic (not just any mic will work – I’ve heard of people using SM-57’s for this!) and an analyzer. These tools help you hone in on problem frequencies – whether as peaks or dips. Another important factor I want to bring to the attention of those also trying to record their sermons with a simple workflow (not unlike mine which I’ve discussed here before), and that is that you will find you get a much more natural and balanced sound when you have your system optimized for your room because the adjustments you now find yourself making on a channel strip are those unique to the individual sources, not to compensate for (global) poor system/room tuning. This should translate into better performance on most other playback systems. Conversely, if you take a recording done on a system that is not optimized, and play it back on one that is, what you’ll hear is the correction you’re adding with your channel EQ settings to compensate for your system’s lack of optimization. Another option is to put a graphic EQ inline with your recording device specifically to compensate for this, but at that point you have to ask yourself why you don’t just put that money into a main EQ which will improve the performance of your entire system, not just the recording bus.

(Some) Church Sound System Essentials Part 2

Monday, March 10th, 2008

Topics for this post include Subgroups and AUX busses on mixing consoles. Subgroups can really simplify managing the mix, especially if your services are very dynamic. AUX busses are a necessity as well if you are running your monitor mixes from the same console as the FOH, and I will also elaborate on how I use AUX busses for my record mix.

Current Version of SR32.4

Subgroups are smaller (sub – below a full mix) mixes within the main mix generally. Here’s how I use mine (on a Mackie SR32.4) and why: I assign my vocal mics (for singing) to subgroup 1 by first depressing the button on each vocal mic channel labeled 1-2, then turning the pan control to the left which narrows the scope of the signal down to subgroup 1. I assign my two speech mics to subgroup 2 by again depressing the 1-2 button and turning the pan control all the way to the right. I assign my instruments to subgroup 3 by depressing the 3-4 button and panning all the way left, and lastly I assign my six drum mics to subgroup 4 by depressing the 3-4 button and panning to the right. What this allows me to do is run most of the service with 5 or 6 fingers. Once I have established the balance of the vocal mics (by pressing the solo button on the Subgroup 1 master channel strip) I can adjust vocals as a whole with a single fader. Same with my instruments on subgroup 3, the speech mics (for speaking, a wireless Shure LX series handheld with SM87 head, and an old AT W series with 831c lav), and the drums on subgroup 4. Because of the way I have my record mix setup (or should I say I setup my record mix this way because) I can tweak the speech mic level going to the record mix by adjusting the channel fader for that mic, and while watching the level meters on the PMD570 recorder simultaneously trim subgroup 2 to maintain the appropriate level at FOH. If the voice (person) leading a given song changes I can always trim up that channel’s mic to bring it out in the mix, and I may pull back subgroup 2 as well, depending on what kind of level we’re already at and if the other vocalists will be coming in at all. But you get the general idea: make your fine mix adjustments with the individual channels (by using the solo button on the associated subgroup master channel strip), then make the coarse adjustments with the subgroup. And again, note that this allows my record mix through the AUX busses to work out the way I want which I’ll explain. One thing that I know generates confusion about AUX busses is the whole “pre-fade” and “post-fade” thing. What this means is that the signal feed to a given channel’s AUX 1 control (being pre-fade, for instance) is pulled from the channel before the channel fader. Which means what? You can jack the channel fader around all you want and the level of that channel’s signal going into the AUX 1 mix (presumably a monitor mix) will not change. Which is how we want it 99% of the time; we don’t want to mess up the person’s monitor mix when we make a change to the FOH mix. Post-fade is the opposite, the feed to that channel’s AUX buss 3-6 (on the Mackie, and 3-4 is switchable) comes after the channel fader. Which now means that it doesn’t matter how high you hack the AUX 3-6 send on that channel, if the fader is all the way down, the most you’re going to get is maybe a smidge of crosstalk. Post-fade sends (tech speak for AUX buss) are used for effects such as reverb and delay, and also for record mixes (at least in my setup.) My theory on this is that if I need to trim a vocal channel or the speech mic for the house, then I would also want to trim it for the record. The trick is getting the AUX level set for each channel so that that relationship works for both mixes within tolerance. But that’s also the beauty of it – I can setup a different mix for recording than I use for live, which means I can use ambiance mics for the record and just not assign them to the FOH mix. And I feed the PMD570 into the tape in on the console so that I can monitor the record mix from the console as well – including the processing I’m doing between the console and the recorder, a Behringer Ultramizer (which I wouldn’t recommend to anyone.) It was a good idea in theory, but the hard compression that this unit applies means all it’s good for is a fancy limiter. Being 2 bands it does allow you to reduce some of the brutality by setting the crossover point just high enough that thumps and bumps and heavy low end don’t bring the entire mix to its knees. And why am I doing all of this? So that when I get home with the flash card I can have a good solid recording that only requires some simple edits in Fission, final MP3 encode and it’s ready to upload. See what you think of the results, our sermon downloads are here and here. Low bitrate versions are encoded with iTunes using this workflow. So, back on the main highway here, a very general guide is that you use pre-fade AUX busses for monitor mixes and post-fade busses for effects (but not processing devices such as compressors since in general they’re meant to be inline devices, not mixed in in a parallel fashion (as you typically would with reverb or delay), and to feed recording devices and/or even distributed sound speakers in the nursery or bathrooms. Our distributed feed is tapped off the output of the Ultramizer, which is feed by AUX 5 and 6. Which brings me to the last thing I want to cover, using a pair of AUX busses to generate a stereo mix for your recording device. I’d love to run a multitrack recording of our services, at least in theory, but that’s not a real need now and just another way to burn up 8 more hours of my precious time. So, for a time I was using AUX 5 and 6 for Left and Right channels, respectively, to generate a stereo mix for recording. What you do to “pan” a voice or an instrument is adjust the AUX 5 and 6 controls differently, for instance; to pan hard left you only turn up the AUX 5 control for this channel, and leave AUX 6 all the way down (and then feed AUX 5 to the left channel of your recorder, and AUX 6 to the right.) To pan in the middle just adjust 5 and 6 to be the same. And for somewhere in-between adjust one AUX slightly higher than the other. That’s the down and dirty way to pan. When we regularly had 2 guitarists playing it really added some nice width to the soundstage of the recording to pan each one fairly hard to one side and the other, and of course panning the cymbal mics and toms, and the vocals.

(Some) Church Sound System Essentials Part 0

Monday, February 18th, 2008

So I can’t help myself when I start talking on a topic and before I know it I’ve drilled down and lost sight of the 30,000 ft view altogether. Here’s the post I meant to do first, hence the “0″ designation. Here’s a list of what I consider to be basic necessary components for most churches of 70 – 300 people. My emphasis is going to be on getting a good recording of the sermon. I’ll drill down in following posts so please bear with my brevity.
Mixing console (‘ll be speaking about analog consoles since that’s where my experience is) with slightly more channels than you think you need, and, 1) at least 1 band of parametric EQ per mic channel, 2) at least 4 pre-fade AUX sends, 3) at least 4 subgroups, 4) high pass filter on every mic channel.
House graphic EQ, preferably 31 band.
Wireless, cardioid pattern: lav, headset or hand-held condenser mic.
Compressor for at the very least your main speech mic (regardless of whether you’re recording or not.)
Digital recorder of some sort if you plan to record sermons.
Good set of over the ear headphones for monitoring your record mix and individual sources.
Speakers. Yeah, you need them. I’ve built all the speakers I’ve used for the last 17 years so I’m a bit out of touch with commercial speakers, but I can give you some guidelines.
Subwoofer. You don’t necessarily need one if you’re under 100 people and/or depending on your style of worship, but what I would suggest is maybe go a touch smaller on the “satellites” and buy a sub rather than buying slightly larger mains and no sub. Remember, I’m keeping this particular post high-level.
Mics. Spend more which should allow you to spend less in the long run because of obsolescence.
Monitors. I built my own monitors too, with this in mind; the smaller the room, the smaller the speaker.
Amplifiers. Yup, gotta have them too.
Crossover; depending on whether you go with powered speakers or not. Hopefully I can clarify at least one aspect of crossovers – what and how the slope (expressed in dB per octave) effects performance and usability.

(Some) Church Sound System Essentials Part 1

Sunday, February 17th, 2008

My goal for this post is to talk about what I consider to be some of the essential components of a church sound system, for smaller-ish churches (50 – 300 people.) A mixer with parametric EQ (also commonly referenced as “sweepable mid” on many consoles), a high pass filter that’s at least 12 dB per octave around 75 Hz, at least 4 subgroups, 4 pre-fade (or pre-fade selectable) AUX busses and if you’re about to purchase a console you should buy more channels than you currently need because like digital files, they will expand into the space allotted – instruments and sources will expand into the channels available.

Mackie  EQ

Here’s a classic example and the replacement for (or you could say latest incarnation of) the console we have, the Mackie SR32.4. I’ll drill down on a couple specifics and explain why you should care. Parametric EQ (a.k.a. sweepable mid), because as much as we’d like, all the sonic problems we’re going to encounter do not happen at 1k or 2.5k (they actually happen at 3k :~P) Parametric EQ allows you to select not only cut or boost, but which frequencies to cut or boost. Just bass and treble isn’t going to get the job done, you need at least one band of parametric to address most tonal issues, especially on voices. For a (male) pastor with a good resonant voice you may boost the low EQ a touch then dial the parametric freq down very near it’s lowest point and then cut a little bit. This will give you a little FM action without it getting so muddy in the mid-bass (150 – 250 Hz.) Here’s an exercise that should paint a picture for you of what a parametric EQ does. Set the cut/boost control to center or 0. Now, play a CD through this channel and slowly sweep the freq control from one end to the other. What did you hear? The correct answer is “nothing!” Now, set the cut/boost control to about 3 o’clock and now slowly sweep the freq control from one end to the other. What you’re hearing is the frequency band you are boosting slowly being moved in the frequency spectrum from lower to higher (or higher to lower depending on which way to turned the control.) When you turn up a fixed mid control or the low/high bands you always hear the same range being boosted, those have a fixed frequency band that they effect. A parametric is like having a 1,000 band graphic EQ, with only one slider – you can have any one of 1,000 frequencies to work with, but you can only pick one. Try setting the cut/boost to 9 o’clock and sweep the freq control. What you’ll hear will sound similar at many ways, and therein lies the crux of deciding whether to cut when trying to fix a tonal problem, or to boost. Because the absence of certain frequencies will have a similar perceived effect of boosting certain other frequencies. You have to deduce if the problem you’re chasing is a peak – an excess of something, or shear lack of content many times requiring a boost.

High Pass filter

HPF


If you’re recording your sermons, simply put; you need a high pass filter somewhere in the signal chain of your speech mic, period. It’s just poor production to have thumps and bumps and pops in a speech signal. For 99% of people’s needs there is nothing significant below 75Hz for most voices – dump it, get rid of it, stop offending those of us having playback systems with horsepower in that range ;~) In fact, in my opinion and with very few exceptions, you should just clean house and turn on the high pass filter for every channel except electronic keyboards, the bass guitar, CD or other pre-recorded playback channels and the kick drum. Those are the primary sources that generate any meaningful content in the range below 75Hz. If you have a system where you can control signal routing to your subwoofer (if you have one) yes you can just not route the other sources to the sub, but if you’re recording at all I would still suggest engaging the HPF for the sake of the recording – absolutely on vocal mics.) We’ll get into subgroups and AUX busses next time.