Church PA / Sound – Top Tips
We have a mass of experience in doing live PA and sound mixing so here are some of our tips. Please also visit the Audio page for more content, some of which overlaps. Please note this is a guide, opinion only. We cannot be held responsible for any actions as a result of the advice below. Page updated February 2020.
PDF Downloads – done for charities
Live EQ, Compression, Gating, Reverb guide (especially using Mackie Master Fader – or similar)
Cheat Sheet of Live EQ settings (using Master Fader but applicable to any mixer!)
Basic normalising / ‘mastering’ in Audacity – done for a church for recording talks
1. Your PA
Is usually made up of speakers (sometimes called ‘cabs’), a mixer, a power amp (where you don’t have powered speakers) and the leads needed to connect things together. There are many ways of doing sound in a church, many ways of putting together a sound system (or ‘PA’). But the principles are pretty much the same how ever you do it.
2. The speakers – powered (active) or unpowered (passive)
Normally, speakers are put up on poles or stands, to the left and right of the stage, usually in front of where the band / person speaking will be standing (we’ll explain why later). There are very generally, 2 types of PA speakers. One is ‘passive’ and the other is ‘active’. Sound needs to be amplified in order for the signal to come out of the speakers. So where your PA speakers are ‘passive’ you will need a ‘power amp’ to generate the power to push the sound signal so it can be heard. Where the speakers are ‘active’ it means the power amp is built into the speakers, so there’s no need for a power amp – one less piece of kit. There are different sizes of speakers, commonly described as 12″ and 15″ ‘tweeters’ or ‘horns’. Where the speakers are ‘active’ (have their own power), there will be a volume switch on the back.
When setting up your speakers, have the ‘tweeters’ about level with the heads of the people (so adjust the speaker stands accordingly). Face them in towards the people at a slight angle so they ‘cover’ the room. The two speakers should be placed left and right, ideally equi-distant from the congregation.
The band should be positioned behind the rear of the speakers to stop ‘feedback’ (that high pitched squeaking sound that causes everyone to cover their ears!)
In terms of power of speaker, a 250 Watt speaker will cover most average size rooms effectively for general speaking. For more band and audio-intensive applications, go for more volume (and for sub bass speakers). We had 350 Watt speakers and they were excellent. Speakers can often perform better at higher volumes so judge well and take advice.
On the average speaker, you will see the front has 2 ‘tweeters’ (basically holes, often covered over but not always) where the sound comes from. This is done to split the different frequencies of sound, splitting the lower frequencies and the higher frequencies (very generally). The lower ‘hole’ will be larger and will push out more of the bass sound. The top ‘hole’ is where the ‘treble’ or higher frequency sounds come from. Why do the sounds need dividing? Well, because the speakers become more efficient and clear. If there is only one tweeter, then all the sound must come through that. Where there is 2, or 3 tweeters, only certain sounds come from them so the sound becomes more defined.
You can go one step further and get a ‘subwoofer’. This is where you want or need very deep sub-bass (40-120Hz), the kinds found in dance music, hip-hop, drum’n’bass – bass you can feel not just hear. You also need them to give a band that needed ‘kick’ and ‘punch’.
Generally speaking, these subwoofers look like black boxes. Bass at the lowest frequencies is unidirectional, so you could potentially put the subwoofer (often also called ‘bass bin’) anywhere in the room. Usually, however, they are placed underneath the main speakers, with a pole connecting the two (as opposed to speaker stands).
If you have one subwoofer, place it in the front centre of the room. Often, you will hear of something called a ‘crossover’. Very simply, the crossover can be built into the subwoofer, or the power amp. What it does is enable you to decide where about in the sound frequency range, you want to divide the sound coming from the subwoofer or the speakers. People often set this in and around 100-120Hz (where you have a choice).
Again, you get subwoofers that are powered or un-powered (active / passive), the better ‘active’ ones (powered) also come with a volume switch. In some PA setups, the main outputs of the mixer go to the subwoofer(s) and then via a ‘high-pass’ output, into the speakers. This means that the sound goes to the subwoofer, the sub takes the low bass frequencies, then sends out everything else in the sound spectrum to the main speakers. The better subwoofers are generally the ones made from wood.
So, do you go for an active or a passive set of speakers? Well, there is no right answer. An in-built amplifier in active speakers is advantageous if you want a simple set up, or where you don’t want to lug around an amp. However, passive speakers enable the sound engineer to decide which amp he or she wants to use, and give more flexibility. Obviously, the active speakers cost more than passive speakers – but with passive speakers you have to factor in the cost of the amp. At the end of the day, which speaker you choose is down to personal preference – listen to them in a shop. Highly recommended is anything from Mackie, HK Audio, db Technologies, JBL, Yorkville to name a few…
3. The Power Amp (when needed)
Only needed where there are passive speakers and an un-powered subwoofer. But which amp do I get and most importantly, how much power do I need? JBL recommend you use an amp that “delivers equal to or up to double the IEC power rating of the loudspeaker“. So, 2 speakers at 250 watts each = 500 watts. So you’d want an amp of up to 500 – 1000 watts (1K). If you add in a subwoofer, you need to have space for that too. So with 2 x 250 watt speaker, and a 500 watt subwoofer, you’d want a power amp of 1000 – 2000 watts. If you are going to add subwoofers (or may do), get a power amp that offers the option of (or has) a crossover built-in. Alternatively, you can buy a power amp specifically to power the subwoofer, depending on your budget. Good makes include those from Yamaha and Peavey. When the mixer is on, it is recommended that you have the mixer buttons set to maximum, or as near to as possible, then control the levels of sound (volume) from the mixer.
4. The Mixer – unpowered or powered
An extremely important part of your kit. Bear in mind that mixers often ‘maximise’ the amount of inputs they have, so be careful to take a look at the actual mixer and think carefully about how many ‘inputs’ you need. What is an input? It is simply a connection into the mixer. It is where you plug the leads from the band (or preacher) into the mixer. You can then control the signal (sound) coming into the mixer, before you send it out, via the power amp (if you have one) into the PA speakers. The reason for having a mixer, is that you can control a variety of sound sources coming out of the PA speakers. So, you can have more than one vocalist, guitars, CD player etc. – and control the volumes and the way the sound is heard.
Before purchasing a mixer, you need to decide what you want it for, how many inputs it has, whether you want effects in-built or not, and the level of control you need from the mixer. You also want a mixer that is as ‘quiet’ as possible, so it doesn’t introduce un-necessary sounds into the mix (hums, buzzes and electronic sounding noises etc.) Basically, however, all mixers do the same thing. Again, highly recommended is Mackie, Soundcraft and Allen & Heath. Though more expensive, you have top quality and noise free components. Behringer WiFi mixers are excellent and a lot better than their normal mixers.
There are a number of ‘inputs’ usually located at the top front of the mixer. Although there is only one type of input shown on the image above, we usually find there are 2 types of inputs – accepting leads known as ‘jack’ and ‘XLR’. A jack lead is a single lead with a connection at each end that slots into a round hole. The XLR is a different type of lead and you’ll see that there is a ‘male’ and a ‘female’ end with either 3 pins sticking out (male) or 3 slots for pins (female). We’ll look at leads later.
How it works… the input lead is put into the mixer – these are referred to as ‘channels’ so we see that from the top left to the top right, we’ll have channels 1-6 (using our pictorial example above). Then we usually have 2 controls to aid the volume of sound. Firstly, there is a button located near the input, controlling how much sound comes into the mixer from the input (this is called ‘gain’). Then we have the respective fader for that channel below, in line with the input lead etc. This controls how much volume comes out from the mixer, to the amp / speakers etc.
Gain… On the top right of the mixer, you will likely see a meter looking like this one above. The trick is to adjust the gain button so that the amount of sound coming into the mixer doesn’t ‘peak’. As sound comes into the mixer, you’ll be able to physically see it on the meter, with the buttons lighting up according to how loud the sound is. The more buttons that light up, the louder the sound. To help you, meters have green -> yellow/orange > red lights. To ‘peak’ means that the meter displays going into the red. You want to avoid sounds peaking / showing up as red. This is because the sound might distort and/or damage the equipment.
What you want is to maximise the amount of ‘gain’ coming into the mixer. This also reduces the amount of ‘hiss’ that you hear out of the speakers. To do this, you get each vocalist and instrument to play or sing as loud as they possibly can and then turn up the gain as loud as it can go while making sure the audio signal doesn’t go into the ‘red.’
Is the main way of changing the sound that people hear through the speakers. You will normally have a ‘treble’ setting (at the top), a mid-range setting, and a bass setting. On better mixers (and WiFi mixers run by software on computer, device or smartphone), you are able to set the area in the sound frequency that you want to change and by how much.
On most mixers, however, you can simply increase or decrease the treble, mid and bass.
Main faders and sub groups
On the bottom front right of the mixer, there are usually 2 faders, often in red or yellow colour. These are the main faders, and control the volume of the whole mix (all of the sounds coming into the mixer). Up means louder, down means quieter. On some mixers you also have ‘sub-channels’. These sub-channels allow you to control certain channels only. So, for example, if you had a band on channels 1-4, you may want to sub-mix those channels. So you’d tell the mixer that you wanted the ‘sub-channel-faders’ to control channels 1-4. So, when the band finishes playing, you only have to adjust those faders, instead of all of numbers 1-4. This is more important with large mixers, where bands commonly take up 12-24 channels.
On your mixer, you will also have other inputs and outputs to do different things. One of the important ways these are used is when using live recordings. Some mixers have independent (direct) outputs for each channel. These would feed into a multitrack recording setup where they would record, and could then adjust each channel separately after recording had taken place. Of course, if they didn’t have this kind of thing, you could only record what comes out of the main outputs of the mixer.
You will also have buttons such as ‘pfl’ which enables you to solo a channel in your headphones or on its own, out of the speakers. You may also have a ‘mute’ button – to turn the sound off a channel.
Balance The Mix
Finally, you need to balance the mix – only if the mix is stereo (left and right). So this means you’ll be using the outputs on the mixer / amp so that one lead goes to the left speaker, and another lead goes to the right speaker.
The mixer itself is usually placed towards the rear centre of the room. Why? Because the sound engineer needs to hear the speakers / overall sound that the audience / congregation will hear – in order to judge the right levels (volumes) of each sound. This means to decide which instruments need to be louder, or quieter etc. This is called balancing the mix. This also involves tricks such as panning certain instruments to the left or to the right. So, on our example above, 2 electric guitars are panned hard to the left and to the right. The lead vocals, bass guitar and drums all set to play across the whole of the mix left – centre – right. The backing vocals and the acoustic guitar are slightly offset to the centre right and centre left respectively. This gives a ‘stereo’ sounding effect and also means people can pick out instruments in the mix more clearly. There are other tricks to do this, using a graphic EQ which we’ll look at later on.
5. Basic Connecting Up
Once you’ve got the mix set up, you plug in leads (usually XLR) leads from the main output (left and right) of the mixer – and plug them into the ‘inputs’ of the power amp.
From the power amp, you take leads (usually XLR or a lead known as ‘speakon’) from the power amp and into the speakers. Sometimes, you will connect up the subwoofer direct from the amp, sometimes you will connect the subwoofer by another amp. Other times, you will send a lead from the back of one speaker into the subwoofer. Many speakers can take not just a lead in, but a lead out from them.
6. DI Boxes
You may have heard of DI boxes, again available in both passive and active formats. What are these things and why do I need one or more? Well, different types of instruments have different types of signals and impedance (electrical current). It’s important to make sure that these match. So basically the DI box balances these kinds of things. So, a DI box will convert a high impedance signal (for example a guitar) to a mic-level (balanced) signal. This also means you can reduce hum, and increase the length of leads used. A passive DI box uses a transformer inside it to change signals. A good DI box will also enable you to flick a switch to reduce ‘ground loop’ otherwise known as ‘hum’ !
There are 2 ways a DI box can be used. Firstly, they can be used directly between an instrument / amp (such as a bass guitar amp) and the mixer. So, with a bass amp, the bass player would plus his jack lead from his guitar into the amp (so he can hear himself). Then he’d use another jack lead to go between the ‘line-out’ on his amp, into the DI box. The DI box would then use an XLR lead to go out from the DI box and into the mixer.
Alternatively, if there is no ‘line-out’ or ’emulated line-out’ on the amp, the set up goes like this: bass player uses jack lead from bass guitar into DI box. Another jack lead connects a jack lead output from the DI box into the amp. Then an XLR lead goes out from the DI box as well, into the mixer.
DI boxes are highly recommended with bass players. Some amps have a DI output (the Marshall range of electro-acoustic amps for example). But mostly, guitar amps are mic’d up. This means a microphone is placed about 2 inches away from (and in the middle of) the amp speaker, direct into the mixer.
There are 3 main types of leads that you will need in your setup. These are: Jack – XLR – Speakon.
Obviously, the more you spend on leads, the better quality they are. The advantage of quality leads is that they don’t allow sounds into the lead that shouldn’t be there – and they stop sounds escaping from the lead. Without good leads, your PA sound may well have all kinds of hums and noises in, that you really don’t want there!
So how to avoid this? Spend money on good leads. Ideally, your leads should be shielded. Shielded leads mean that the centre of the lead (that carries the sound signal) and sits inside a plastic cover, is protected by a twisted metal cable all the way along. Then this itself is covered by the plastic (usually black) outer cover. So there should be at least 4 parts to your cable.
Secondly, Jack leads over 10 metres in length will potentially start to act as a radio receiver, and start picking up radio waves! The way to avoid this is to use the minimum length of every cable you possibly can. Secondly, with Jack leads, you van buy what are called ‘balanced’ leads (these can also act as ‘stereo’ leads, rather than a ‘mono’ (single) sound, picked up by unbalanced leads). This basically means that the lead protects itself from picking up interference (‘ground sense’). With ‘balanced’ jacks and XLR leads (the ones with 3 pins), one of the pins acts to stop interference (by acting as a ground). The interference stopping bit is the other 2 leads, which act as opposites to each other to ‘cancel out’ unwanted frequencies, hum and noise.
Jack leads are used for things like connecting up guitars to amps. XLR leads connect a microphone to a mixer etc.
A speakon lead is usually a connection between your power amp and your speaker. These are a quality way to connect up where you insert the speakon end, twist and it clicks into place (usually blue connections on each end of the cable). Again, there are varying levels of quality / shielding / connections. So go for the best quality and the right lengths.
8. Snakes / Looms / Multicore
This is basically a set of leads that stretch from the stage to the sound engineer / mixer, at the back of the room. This is usually connected at one end to a ‘stage box’ which you place on the stage.
The picture is a very basic representation of what it will look like. Basically, the instruments on stage all plug into the box (into XLR sockets, and sometimes 4 or so jack sockets). The box then has all the leads underneath, which are wound together down a single thick black cable (shown at the top of the image) back to the sound engineer at the back of the room. At the sound engineer’s end, all the leads then come out of the single black cable and can be plugged in to the mixer. On the box, the inputs are numbered from top left to right. On our image, we have 16 available.
So, for example, you plug 8 instruments on stage into the box. The sound engineer then winds the single multicore lead from the stage box, down the side of the room, to his mixer. At the sound engineer’s end, all the 16 available channels will have a lead each and these will poke out of the single lead to become 16 individual, numbered leads. So, the sound engineer would find leads 1-8 and plug them into channels 1-8.
You can get these in long lengths, sometimes attached to a cable reel so they are easier to wind back up!! Again, pay for quality and for good shielding / leads. You’ll pay more for one of these on a ‘drum’ (a reel). Although you may not be tempted by the extra cost, it’s wel worth it and easier to fold up and down! For purposes of foldback (not ‘feedback’) there are also leads which send a signal back from the mixer to the stage box. We’ll look at why in the next section.
When you are using a WiFi mixer, it usually lives ‘on stage’ so you won’t need multicore.
9. Monitoring / Foldback
You may have seen performers on stage with little black boxes that appear to be facing upwards towards them. These little black boxes usually sit at the very front of the stage? Or you may have seen a musician with an earplug type thing in his ear on stage. Ever wondered what these are?
These are what’s called ‘monitoring’ or ‘foldback’. Basically, it allows the musician to hear the sound that a band is playing, so the singer / musician will be able to play in time and to sing in tune (well, hopefully anyway!). This is incredibly important as it’s often very difficult to hear the sound up on stage, so a musician can lose their way in a song very easily. This is also the case because the main speakers (front of house speakers – the ones the audience listen through) are located in front of the musicians, so they often can’t hear the sound from them.
So, back to our stagebox or our WiFi mixer on stage, we need to look for the aux outputs. These are used to send the sound that goes into the main mixer – back to the musicians on stage. The aux leads send the sound we want to send to the ‘foldback’ or ‘wedge’ monitors, which sit at the front of the stage and allow the musicians to hear what they are playing / hear other musicians in the band.
So, for example, your vocalists are going into channels 1 and 2, and they’d like to be able to hear themselves on stage. Up on the stage, they have a foldback monitor each.. Vocalist 1 wants to hear themselves and their guitar (which is going into channel 3). Vocalist 2 wants to hear only Vocalist 1 and themselves. What to do?
On the mixer on Channel 1 (Vocalist 1), we turn up ‘Aux 1’. We also turn up ‘Aux 1’ on Channel 3 (Vocalist 1’s guitar). On the mixer you will also have a main control for Aux 1. Turn this up. For Vocalist 2, we need to turn up ‘Aux 2’ on Channel 2 (Vocalist 2) and on Channel 1 (Vocalist 1). Again, we turn up the main ‘Aux 2’ control – like the Aux 1 main control, often located near the top right of the mixer…
Up on stage, we take a jack lead from the channel marked ‘Aux 1’ or ‘Jack 1’ (usually along the lower part of the stage box, as in our diagram). This then connects to the foldback monitor of Vocalist 1. We then take a jack lead out of the stagebox ‘Aux 2’ or ‘Jack 2’ socket, and run it to the foldback 2, located by Vocalist 2.
If you want to use in-ear monitoring, it is similar. In-ear monitoring reduces sound levels on stage (foldback often has to be up loud for the performer to hear it on stage – sometimes too loud!) In-ears also offers the option to play to a click (a click you hear in your ears to a set tempo to enable you to stay in time), or a ‘click track’ / ‘backing track’ where you have a track playing with other instruments that aren’t in your band. An excellent tool to have – we have a separate headphone amplifier (Behringer HA8000) connected to our mixer so we can boost signal levels to our in-ears. I personally use a small Behringer MicroAmp when playing solo or in a duo.
Microphones are another area that can be hard for people to understand as again, there is so much information and things to think about.
The main type of mic you will use in a PA setting will be what’s called a ‘dynamic’ mic, the kinds you see ‘pop singers’ waving around when on stage singing ‘live’. You may have heard of a microphone called the Shure SM58. This is a mic commonly used the world over for singing. It has a good range and is incredibly sturdy, an important issue with vocal mics! Within the dynamic mics, there are different kinds of mics – these are called things like ‘cardioid’ and ‘super-cardioid’ mics. Basically, this tells you how the microphone picks up sound and where from.
The cardioid mics pick up from the front of the mic (in a kind of heart shaped pattern). The super-cardioid mics are much more directional, meaning they pick up only from the front of the mic. The cardioid and super-cardioid mics pick up a signal tight to the vocalist’s mouth, so that other signals don’t get picked up from that mic. This is very important in on-stage applications because of something called ‘feedback’ which is that painful ringing noise you sometimes hear from speakers. Feedback is when sound gets into the PA and loops around the sound system. This is something to be avoided!! The SM58 is a cardioid mic. Most of these types of mics have a graphic EQ ‘boost’ around 1-4kHz, to allow the vocalist to ‘cut through’ (be heard in) the front of house mix (the sound that the audience hears).
It is also important to tell the performer where to put the mic in relation to their mouth, so you get the sound you want / need. Other important kinds of mics are those used to mic up instruments such as guitar amps. A commonly used mic is the Shure SM58’s brother, the SM57. For drums, specialist mics are recommended (you can buy drum mic kits).
Mics don’t produce a lot of power themselves, so you need to put a mic through a mixer (which have in-built ‘pre-amps’) meaning that it gives the mic signal a boost before it goes out through the PA. Other times, a specialist ‘pre-amp’ is used. Commonly mics also have a compressor used on them (a device that processes the sound or ‘signal’ that comes from the mic, or from a PA). We’ll look in more detail later (below, point 13) but it helps to keep the singer’s volume levels sound smoother and more consistent.
A second type of mic is known as a ‘condenser’ mic. These mics are very sensitive to more distant sounds and high frequencies. So, you’ll see these as ‘gooseneck’ mics (long, thin necks) near preacher’s pulpits etc. But you’ll also see them hanging from the ceiling to pick up congregations in churches, and at the front of churches or hung above choirs. Compared to dynamic mics, they are often thinner. (You also find condenser mics in studios, but these are the ones that are bigger and often have ‘pop’ shields in front of them to stop the singer making popping noises when singing words beginning with ‘p’ or ‘b’). Condenser mics need to be powered, so on your mixer you will see a button labeled +48v Phantom Power. This needs to be one for the condenser mic to work. DON’T have this depressed if you’re not using condenser mics, though.
When a singer is working with a mic, they need to learn mic technique. This means, holding their mouth close to or up to 2 inches away from the front of the mic (have the mic facing the performer). When they hit high or powerful notes, they must step back from the mic, or pull it away from them if it’s handheld. A singer may also hit notes that cause a ‘pop’ across the speakers (especially with words starting with ‘b’ or ‘p’). In this case, you may have a ‘HPF’ (a high pass filter). This cuts out frequencies below 100Hz, commonly. However, pops on mics are often in the 60HZ-80HZ region so if possible, it’s good to cut the mic frequency there, rather than at 100HZ. This means having a separate graphic EQ which we’ll look at later. However, better quality mics will not do this. So whether it’s on handheld dynamic, or on radio mics, you get what you pay for.
Radio Mics come in varied formats. There is a ‘free frequency’ range of UHF mics which operate (as of March 2019) in the 863 – 865 MHz range. This is a free to use frequency range but it therefore means other people may be using that frequency range so you may get some weird noises or feedback! There is also a 2.4GHz range of mics which is also free but this is the range that bluetooth devices use, mobile data uses and some WiFi networks use.
You can then get wireless mics in frequency ranges for static venues and ranges where your mic isn’t just used in one venue. These are paid-for frequencies that you can choose (if using in varied venues) or can be given (if in a fixed venue). For both, you must legally register the mic and then pay a license fee to OFCOM yearly or bi-annually.
Then your options are handheld mics, lapel mics or headset mics – depending on your needs. If you do get a headset mic, go for an unobtrusive ‘skin coloured’ headset.
Quality makes include Shure, Sennheiser and Audio-Technica.
Firstly, a handheld mic. These are the ones that the artist will hold and that look like a microphone. Useful for rap, lead vocalists who like to move around a lot, for people leading a service/ giving the talk etc. The artist takes the mic (and mic unit), the sound engineer takes the ‘receiver’ plugs it into a power socket, then takes an XLR (preferably) out of the back into the mixer. Put the receiver aerials up and away you go.
Secondly, a tie clip (or ‘lavalier’) mic. These are the ones where you have a ‘bodypack’ which is usually clipped to someone’s belt, or put inside a suit pocket. There is a wire attached to the bodypack, which has an ickle mic on the end, with a little furry ‘hat’ or ‘pop shield / wind shield’. The little mic bit has a clip attached. The clip attaches to a shirt, tie, pocket etc. Put the mic near to the face as possible, preferably around the top of the tie kind of area. Again, back at sound engineer base, there’s a receiver needing to be attached to the mixer, with antennae up!
Thirdly, a radio mic that sits around the head called a headset mic. Usually they look similar to a pair of headphones around the top of the head, with a mic curved round from one side of the headband, towards the mouth. The latest types of these simply attach over the ear, with a small mic which is skin coloured and which stretches across the cheek towards the mouth on one side. Again, there is a bodypack attached to the user, and a receiver back at base.
Drum Mics – another specialist type of mic to pick up a drummer. Now, in very echoey halls (such as sports halls) and smaller venues, you won’t want to mic up the drums. Where you don’t have a very loud band, or don’t have good bassy speakers (or a subwoofer or two), you will also not need to mic up the drums. However, when you do need to mic up the drums is when you need to be able to control every sound from the band, so you can adjust the levels (and not the band themselves!) So, drum mics come in all shapes and prices. Again, go for the best you can get with the money available. Highly recommended are those from Shure, and those from Red Audio in the UK. Mics come in the form of: a kick drum mic (locate just off centre, in front of the kick drum); a snare mic (affix to side of snare facing snare drum head, using mic clamp which should be provided with drum mics); hi-hat mic (affix under hi-hat). Where you only have one mic for snare / hi-hat, put mic closer to hi-hat than to snare. Then you may have mics for the toms (affix as per snare drum); overhead mics (which will be battery powered or powered using phantom power from the mixer). These need to go some way above the cymbals / crashes, facing down onto them.
Mic-ing guitars / bass – Normally you would use a condenser mic (powered by phantom power or battery) but it’s not essential. I use some quality dynamic mics (like the Shure SM58). Electric guitars need to be mic’d so that the mic is in front of the speaker grille of the amp, around 6cm – 1 foot away, depending on the kind of sound you want (experiment). An acoustic guitar can be DI’d directly into the mixer / multicore – or can be mic’d up with mics near the ‘hole’ in the guitar (next to the strumming hand), and sometimes another mic nearby in front of the fretboard. A bass guitar should be DI’d.
11. Avoiding Feedback
Ever heard that horrific squealing noise in church? No, it’s not just cos the pastor has made 15 ‘last points and everyone’s bottom is numb! No, it’s because of feedback (not foldback). This is when a signal gets into the sound system and loops around it causing a high pitched frequency to suddenly appear. This often happens when sound is turned up too loud, without the right mics, right equipment – or an inexperienced sound engineer.
So how do we avoid it?
Well, first of all, use the right kinds of microphones. If trying to cover a wide stage, don’t use dynamic mics across the front (where possible). Instead, use condenser mics appropriate for covering a wide area and picking up a lot of sound. Using a dynamic mic will often cause feedback in this kind of situation. Conversely, don’t use these kinds of condenser mics for vocalists or where there are lots of people / instruments on stage. In this situation, use dynamic mics with a cardioid, preferably super-cardioid pattern.
Secondly, don’t have the sound up too loud. What some sound engineers do is to get everyone on stage to play and find the point at which feedback starts. What they will do then is: (a). Turn the sound down and make sure it doesn’t go up beyond the ‘feedback’ point. (b). They will try to define which channel(s) are causing the feedback and turn down those channels. Another trick is to use the HPF (‘high pass frequency’) switch on vocal channels. Vocal mics on stage don’t have any sound below 40-60Hz (even up to 80Hz) so these frequencies can be removed.
Thirdly, using a graphic EQ to destroy feedback. A graphic EQ will sit in between your mixer and your power amp. So instead of the mixer leads going out into the power amp directly, what happens is that there are 2 leads going out of the mixer – into a graphic EQ unit – then out of the EQ – into the power amp. This means that when you make changes to the EQ, the sound that you hear back through the speakers will change.
When feedback occurs, it usually happens in and around a specific kind of frequency. Below is a chart looking at the frequencies of graphic EQ..
On your graphic EQ, you will have either 15 or 31 ‘bands’ of EQ. This basically means the sound spectrum (as shown in the pic above) is divided up into 15 or 31 different bands from 20Hz up to 20kHz. 20Hz is a very low sound, 20kHz is a very high pitched sound. There is not always one place where feedback occurs (although it’s usually in and around the middle and often lower than you think).
The secret when you have a manual graphic EQ is to setup the PA sound so that you can hear the feedback starting to ‘ring’. Keep one hand on the mixer faders (to raise and lower volume to create / get rid of feedback) and then go through each of the faders and pull them down as feedback occurs (to ‘take out’ that frequency) until you find the place where the feedback is occurring. Now you can turn up the volume of the whole PA. As you turn up, you may find some more feedback occurs, so you will need to repeat the procedure to eliminate the naughty frequency..
HOWEVER – as you remove parts of the sound spectrum, you will start to change the sound coming through the speakers. Too much fiddling and you will ruin the sound. So adjust / take down 2 or 3 channels maximum on your graphic EQ.
You may well have a graphic EQ that has a ‘Q’. This is basically how tight to the frequency that you want to go. For example, the frequency causing feedback may be at 1kHz. Now, if you have a wide ‘Q’ then it will also reduce other frequencies near to 1K. If you have a narrow ‘Q’ then it will basically only reduce 1kHz, and the audio won’t appear so strange-sounding! Here are 2 diagrams (taken from Logic Express) to illustrate…
Firstly, a wide ‘Q’ graphic EQ at 485Hz – the green slope is very wide. Notice how the ‘Q’ level is 1.00.
Secondly, a narrow ‘Q’ graphic EQ. See how the green slop is much thinner with the Q set to 10.0.
12. Graphic EQ in the Mix
Graphic EQ can be used to ‘help’ the sound. For example, instead of turning up an instrument in a sound mix, or instead of turning them down, you can sue a clever bit of EQ. Different instruments and sound ‘sit’ on a different parts of the sound spectrum / frequency. By using a bit of extra EQ, or by removing a bit of EQ, you can help sounds stand out, or remove them a bit. So, for example a dynamic mic has a little boost around 1kHz. If the sound from the vocalist needs a little something extra, you can turn up the EQ on the mixer at 1kHz for the vocalist. You may then want to reduce other vocalists or instruments on the mixer by removing a bit of their sound at 1kHz. The result? The vocalist sound ‘sits’ more in the 1kHz range and can sound ‘turned up’ in volume, without touching the volume fader!
13. Compression / Limiting
We’ll look at the very basics of this for now. Compression is always used on vocals in studios, and regularly in live PA applications. What it does it to take the signal and compress it so that it reduces the difference between the loudest and quietest parts of a performance / service etc. This gives more of a constant level of sound – especially used on vocals and for bass guitarists. What the compressor does it to work on any signal that goes above a certain (defined point). A limiter does a similar but more extreme job, and is used on TV, radio and in live situations to stop sound distorting or ‘clipping’ (going into the red). What it does is to heavily compress a sound source (such as from your mixer) to keep it within very tightly defined parameters.
The ‘tight’ bass drum or bass sound you hear on all tunes is partly through excellent compression. It gives a punch to sound. Compression is also essential on vocals to maintain a consistency in volume of sound and to aid the overall vocal sound. Compression will aid in levels as evening up the overall levels of the song means you can boost the volume further, to maximise the sound, without distorting the song.
Basic idea is a ratio of say 2:1 will compress the signal to twice the input signal. The threshold setting is the level at which compression kicks in (so -19db will allow more sound through than -3db). The attack is how quickly the compression acts on the signal so if it’s 0ms or automatic it will be almost instant. The release (also called decay) is how quickly the settings are released from the initial action on the signal. So, a faster setting will mean the compression decays quickly, 10ms will allow a little bit of time before the compression is released. Hope that makes sense!
Easy compression settings
For brief help, compress drum tracks at ratios of 5:1 to 8:1, set attack to 5ms, release (decay) to 10ms and the threshold at around -15 to -19db. For vocals compress at say 4:1 (though you may go higher with the compression ratio), set the threshold at around -3db (though you can increase this) and attack and release to automatic or lowest setting. For the whole track try ratio 2:1, attack and release at fastest/auto and the threshold at -5db to -9db. If you want to only use compression over the whole track, not individual tracks. Just experiment and see.
14. Other ‘Signal Processors’
Commonly used is a ‘reverb’ effect, which makes someone sound like there are in a large space, such as a room. This is especially used live on vocals to add an extra dynamic to the sound. Without reverb, sound is very ‘dry’. So you hear a sound, and then you don’t hear it. What reverb does is to add ‘space’ to a sound so that it is heard for longer and has more of an impact. A reverb setting up high would sound like someone shouting in the middle of a cathedral. So it has to be used carefully and sensitively. One application for practical use is when 2 vocalists ‘clash’ in the sound mix. By adding a touch of reverb to one of the vocalists, it’s like they are pushed ‘back’ into the mix. Therefore one vocal would stand out above the one with more reverb.
Another effect is a delay, which is basically like an echo – it creates another sound after the original sound, but delayed. You can change settings to have different effects. For example, if you have a very short delay, it can have the impact of making audio sound as if it is layered (like there are 2 people singing the same thing).
An expander / exciter – is an effect that adds body to the overall sound.
Finally, we’ll think about a noise gate. A noise gate will shut down the audio when it falls below a certain volume. This is commonly used so that extra quiet noises (like hiss on a mic) can be quickly shut down and taken out of the PA sound. They are also used where mics are in close proximity to each other (such as on drum kits) so that one sound (for example a snare mic) doesn’t ‘spill’ into another mic (for example a hi-hat mic).
15. Power and Ground Loops / Electrical Protection
Sometimes you may have a low hum or a strange noise that seems to be coming out of your speakers when it shouldn’t. This may be due to what’s often called a ‘ground loop’. Without being too technical (as I can’t be!) this is where the power tries to ‘earth’ or ‘ground’ itself to two different points, thus creating a feedback / ground loop. So, for example, you have two guitarists on stage and they are plugged into 2 different sockets. Both those sockets are ‘earthed’ at very slightly different ‘earth points’. This means that a loop can be created between the 2 guitars. Without being a regular occurrence, or wishing to cause alarm, this can cause serious injury or death.
What is the solution? Well, there are 7 steps that may help..
First – check the electrics have been recently tested in the venue you are going to. There is a legal requirement that venues should be tested periodically to make sure they’re safe (I believe it’s once a year).
Second – make sure that the equipment you own is in good condition. If you’re being paid for hire of your equipment (without you being present), it must be PAT tested once a year I believe (tested to make sure it’s all working safely). If it’s a voluntary thing, make sure your plugs are all properly wired, that your equipment is well maintained and protected etc. Look after your gear and it will look after you, so to speak!!
Third – buy an RCD (Residual Current Device) as it’s called in the UK. This is like a plug that goes into your mains plug and helps to protect you in the event of a wire being cut etc. These afford some electrical protection and it may well be worth getting one for each of the plug points you use. In the event of a problem, the RCD will cut the electric current within milliseconds.
Fourth – buy surge protected power plugs. These are normal power strips but with a protection against surges and spikes in the electrical current. So, for example if there is a spike in the electricty (an unnatural surge of extra power, such as lightning or just an electrical fault), then a massive surge of electricity comes down through your plugs. This can go through them into your equipment and permanently destroy or damage it. A surge protected strip will protect from all but the most dramatic of these surges. The more you spend, the better the protection. Again, every main output from the mains should be protected with at least one surge protected lead. Why risk your equipment.
Fifth – on stage, try to use just one mains outlet for the band. Then daisy chain power strips (protected by an RCD and surge protected power strip at the mains end) to the whole band. This way, all the band is earthed to just one point, making it safer. This will also protect the band, as the whole band (and guitars etc) will be earthed to one central point.
Sixth – to address the hum and buzz issues specifically, try using a DI box on the leads going from your mixer / power amp, to your speakers. Or, if you’re using unbalanced jack leads in a long cable run, try using the Behringer Hum Destroyer. This converts unbalanced signals to balanced signals. For specifically feedback destroying, use an EQ unit, a feedback destroyer, or the DEQ 2496 from Behringer.
If all else fails, go through individual channels on your mixer – try to see if one speaker or other is damaged – whether you’ve got a faulty lead – faulty other equipment. Work logically through all the bits of equipment until you identify what the problem is.
Finally – when you test a mic, ALWAYS use the back of your hand. If you use the main part of your hand and there is an electrical problem causing an electrical shock, then your muscles in your hand will contort and grab hold of the mic and you will not be able to let go. If you use the back of your hand, this will be avoided. In the event of a problem, you will receive an electric shock, but live to tell the story.. It really can be that serious.
So, look after yourself and keep in prayer before God in all you do. Be ready to do what he says!
16. Wireless Connectivity
At my church, both the sound is controlled via an iPad (or other device) which connects to the mixer / projector wirelessly. Pictured is the original Mackie DL1608. We now use the Mackie DL32 which gives a lot more inputs and outputs (meaning more people can use in-ears or we can feed more than one set of speakers).
The number of inputs on the DL32 mean that we use fixed channels for instruments. So for example Channel 1 is the Radio Mic, Channel 2 is the headset mic, Channel 3 is the worship leader… And so on. The same is true for the Aux channels – so Aux1 is the in-ears for the worship leader, Aux 2 is the in-ears for the keyboard player etc. This makes plugging in more simple!
The use of iPad over a fixed mixer gives the operator the ability to be anywhere in the room and doesn’t limit them to sitting behind a PA desk! This also frees up space in the room with no space needed to fix a static mixing desk. However, some people prefer a static desk and the extra hard-wired options it offers. However, I have never had an issue using WiFi mixers or had the WiFi go down.
If you are considering buying a PA for live sound in a church context, this is increasingly an option to consider.
I personally use the Behringer XR18 which has excellent pre-amps (the sound quality is excellent). I personally prefer the options, range and quality of effects of the Behringer software app (XAir). If XAir doesn’t work for you (it’s not available on iPhone but only for iPad OS) then you can pay £4.99 to use the app ‘Mixing Station’ – this also works with Mackie devices.
The software for Behringer is available for Windows, Mac OS, Android and iOS. The Mackie ‘Master Fader’ range only allows for iOS use on its v4 software but this has changed with Mackie’s Master Fader 5 software (which they hope to make fully available even for older products).
Between the Mackie and the Behringer there isn’t a lot of difference. At church we will be moving to use the DL32 from Mackie to give more channels and more aux (in-ear monitoring) channels.