The Audio/Visual ministry exists to enhance the worship experience through the use of technology. The ministry makes sure that the sound and visual/video has a positive input in all worship services. There are three areas of opportunities in the A/V ministry at Trinity mar Thoma Church: Sound, Media and Camera Operator. To join the A/V ministry please send an email of interest to firstname.lastname@example.org or call the Audio/Visual coordinator.
MISSION We are commissioned to extend the gospel of Jesus Christ through innovation that positively impacts the kingdom of God on earth. As the guardians of sound, graphics and imagery in worship, we take our service unto God’s kingdom seriously, while having a great deal of fun.
OBJECTIVES The AV Ministry requires a high level of commitment, but gives great satisfaction to our team members. To serve in this ministry, some technical knowledge may be necessary and training is available.
#1 – Create an environment throughout the facility that will allow worshipers to experience an encounter with our Lord Jesus Christ
VALUES To accomplish our goal, the following values describe our vision:
- We are committed to the worship and glorification of Jesus Christ
- We seek a deep relationship with God for an increasing level of faith in Him
- We are a team of collaborators for the purpose of the one and only God
- We strive for excellence in every service and special event
- The Holy Bible is the inspiration of our innovation
Worship Service (Sunday): 8 am Divine Service: 11 am Special Events: As Needed
CONTACT Ministry Lead: Email: email@example.com
Why Choose a Line Array Speaker System?
Line array systems have been used for huge (arena- or stadium-size) tours for decades. The reason is that line array systems are the best way to get clear sound throughout the venue, from the front row to the very back. This is for a few different reasons.
One aspect of line array systems that makes them so efficient is the shape of the dispersed sound. Point-source (non-line-array) speakers project sound in a spherical shape, radiating out in all directions from the speaker. But your audience usually isn’t surrounding the speaker in all directions — the audience is usually more-or-less in front of the speaker.
A line array system projects sound in a shape that is closer to cylindrical. This greatly reduces the amount of sound energy that is projected vertically (to the ceiling and floor). It’s good to reduce vertical dispersion for two reasons. One is that the extra sound energy bouncing off the ceiling and floor interferes with the direct sound from the speaker and creates a more confused, reverberant sound. The other is that it’s inefficient — projecting sound vertically is a waste of watts. With a line array system, you can use your watts wisely, projecting sound more directly toward ears, rather than all over the room.
Projecting sound more efficiently becomes more important as your venue size increases, especially if the acoustics are less-than-great. In a big, reverberant room, you have to turn your speakers up more to get clear, intelligible audio to the back of the room. But by doing that you end up exciting the reverberant field in the room more, which causes muddy, indistinct audio that is hard to listen to.
With a cylindrical dispersion pattern, the amount of sound energy decreases with distance at a slower rate. That means that as you move away from the speaker, you lose less volume, meaning that you can project sound farther away using less wattage. This is a huge advantage, allowing more direct sound at lower, less-reverberant-field-exciting volumes, which translates to clearer audio all throughout the room, instead of just at the front, without resorting to side-fill (reinforcement) speakers. And you can do that without destroying the ear drums of everyone in the front row.
These are the reasons that line arrays have been used in large tours for years, and they’re the same reasons line arrays are becoming more popular in smaller venues too. Companies like Bose and JBL use line array technology for their compact, tower-shaped all-in-one PA systems such as the L1 (pictured below), which is what enables those systems to fill up a small-to-medium-sized room so nicely, with comparatively little wattage.
But you can also use full-size line array speakers in medium-to-large-sized venues and reap the same benefits. With more companies offering line array models that are getting more sophisticated, and easier to set up without professional installation, we think that in the future more clubs, venues, and houses of worship are going to be using line array technology to achieve clearer, more efficient sound.
HOW TO HOLD YOUR MICROPHONE, PROPERLY.
Where to grasp:
There are two things to remember when holding a microphone:
- Don’t hold it too high
- Don’t hold it too low
Seems pretty basic, right? You would think so, but how many times have you seen someone holding a mic with their thumb over the head of their microphone? Or worse, their entire hand?! While it may make you look “cool”, it will actually ruin the dynamics of the microphone and how it was designed to perform, possibly resulting in, dun dun dun…. feedback!
Holding a microphone too low can cause similar issues. Generally speaking, the antenna for the microphone transmitter is located at the bottom of a microphone, the same place you would find the cable connected in a wired mic.
So if you put these two tips together then the only remaining section left to hold on the microphone is in the middle.
Still confused? Check out these pictures for some visual assistance.
How to hold:
Now that you have a good understanding of where to grasp the microphone itself, we can go over how to hold it in relation to your body/mouth.
A good starting point is holding the microphone as close to your mouth as possible without touching it. You do have a couple inches of wiggle room, just always remember to keep it consistent. Holding it in front of your stomach won’t do any good - bring it up!
Second, the angle at which you hold your microphone plays a big role. To avoid any issues, hold the mic slightly below your mouth and angle the mic slightly toward your mouth.
In the end, hold your microphone tight and hold it close. The exact distance from your mouth, and angle at which it’s held will vary with each person and sound system, so always test before going live!
And of course, we recommend you do not “mic drop” at the end of your performance. I know I know. All the cool kids and even former US President Barack Obama have done it, but the truth is…mics can be a bit expensive and can break.
When is it best to use hanging microphones for meeting and presentation rooms?
Having addressed the use of boundary and gooseneck microphones in our previous two posts, today we continue our series on meeting and presentation room miking applications by looking at another popular choice: hanging microphones. Microphones can often become distractions for the people using them, and suspending a microphone from the ceiling may reduce this problem. Hanging microphones are most commonly used for audience participation and general pickup of a room while recording or conferencing. The microphone preamp gain level may need to be set higher for hanging microphones than it would be for other microphones in order to sufficiently pick up the sound. Depending on the setup, this gain increase may lead to feedback. But such feedback can be combated with auto mixing and DSP processing.
How Do I Prevent Microphone Feedback?
Feedback, also known as the Larsen effect, occurs when the amplified sound from any loudspeaker reenters the sound system through an open microphone and is amplified again and again, causing a loop. We often tell customers that feedback is not the fault of the microphone because any microphone will feed back given the right conditions (or maybe in this context, wrong conditions). However, there are some steps that you can take to avoid or lessen the likelihood of feedback. Try some of these:
- Keep the microphone behind the main loudspeakers to minimize the sound that can reenter the microphone. If the microphone is in front of the speakers, then feedback is nearly guaranteed. You may notice this when a performer or presenter steps out into the crowd and finds themselves in front of the speakers. More often than not the result is that loud, ugly, screeching sound.
- Use a microphone with a unidirectional (cardioid) polar pattern. A cardioid microphone has its maximum sound rejection at the rear of the mic. Keep monitors or loudspeakers aimed at this area of maximum rejection. Please note that an omnidirectional microphone picks up sound equally all around the microphone and has no area of sound rejection. It is much harder to keep sound from reentering an omnidirectional microphone.
- Place the microphone close to the sound source. When you reduce the distance between the sound source and the microphone by half, you double the sound pressure level at the microphone. This is an application of the inverse square law. It increases your gain before feedback (i.e., it allows your sound system to produce more SPL before reaching a level that would induce feedback). In other, simpler words, if you move the microphone closer to the sound source (your mouth, for example) the sound will be louder, so you can turn down the volume at your mixer. This will greatly reduce the likelihood of feedback.
- Feedback will occur at different frequencies at different volumes. Use an equalizer or the EQ section of your mixer to find the offending frequency and cut back that frequency. There are commercially available feedback eliminators that automatically dampen the frequencies where feedback is occurring. You have to be careful when using these because sometimes they can go too far and notch out frequencies too deeply and make you sound a bit hollow.
Following these steps should help you avoid feedback.
A Brief Guide to Microphones - What's The Pattern?
In addition to classifying microphones by their generating elements, they can also be identified by their directional properties, that is, how well they pick up sound from various directions. Most microphones can be placed in one of two main groups: omnidirectional and directional.
To help you visualize how a directional microphone works, you will find polar patterns in our literature and spec sheets. These round plots show the relative sensitivity of the microphone (in dB) as it rotates in front of a fixed sound source. You can also think of them as a horizontal “slice” through the pickup patterns illustrated in Figures 3 and 4.
Omnidirectional Polar Pattern
Omnidirectional microphones are the simplest to design, build and understand. They also serve as a reference against which each of the others may be compared. Omnidirectional microphones pick up sound from just about every direction equally. They’ll work about as well pointed away from the subject as pointed toward it, if the distances are equal. However, even the best omni models tend to become directional at higher frequencies, so sound arriving from the back may seem a bit “duller” than sound from the front, although apparently equally “loud.”
Figure 3: Omnidirectional Microphone
The physical size of the omnidirectional microphone has a direct bearing on how well the microphone maintains its omnidirectional characteristics at very high frequencies. The body of the microphone simply blocks the shorter high-frequency wavelengths that arrive from the rear. The smaller the microphone body diameter, therefore, the closer the microphone can come to being truly omnidirectional.
Directional Polar Patterns
Directional microphones are specially designed to respond best to sound from the front (and rear in the case of bidirectionals), while tending to reject sound that arrives from other directions. This effect also varies with frequency, and only the better microphones are able to provide uniform rejection over a wide range of frequencies. This directional ability is usually the result of external openings and internal passages in the microphone that allow sound to reach both sides of the diaphragm in a carefully controlled way. Sound arriving from the front of the microphone will aid diaphragm motion, while sound arriving from the side or rear will cancel diaphragm motion.
The basic directional types include cardioid, subcardioid, hypercardioid and bidirectional. Also included under the general heading of directional microphones is the line – or “shotgun” – microphone, a more complex design that can provide considerably higher directionality than the four basic directional types.
Polar Pattern Considerations
Polar patterns should not be taken literally as a “floor plan” of a microphone’s response. For instance, in the cardioid pattern illustrated, response is down about 6 dB at 90° off-axis. It may not look like much in the pattern, but if two persons were speaking equidistant from the microphone, one directly on-axis and the other at 90°, the person off-axis would sound as if he were twice as far from the microphone as the person at the front. To get equal volume, he would have to move to half the distance from the mic.
A word of caution: these polar patterns are run in an anechoic chamber, which simulates an ideal acoustic environment – one with no walls, ceiling or floor. In the real world, walls and other surfaces will reflect sound quite readily, so that off-axis sound can bounce off a nearby surface and right into the front of the microphone. As a result, you’ll rarely enjoy all of the directional capability built into the microphone. Even if cardioid microphones were completely “dead” at the back (which they never are), sounds from the rear, also reflected from nearby surfaces, would still arrive partially from the sides or front. So cardioid microphones can help reduce unwanted sound, but rarely can they eliminate it entirely. Even so, a cardioid microphone can reduce noise from off-axis directions by about 67%.
The directional microphone illustrated in Fig. 6 is about 20 dB less sensitive at 180° degrees off-axis, compared to on-axis. This means that by rotating the cardioid microphone 180°, so that it faces directly away from the sound source, the sound will “look” to the microphone as if it had moved TEN TIMES farther away!
The maximum angle within which the microphone may be expected to offer uniform sensitivity is called its acceptance angle. As can be seen in Fig. 10, each of the directional patterns offers a different acceptance angle. This will often vary with frequency. One of the characteristics of a high-quality microphone is a polar pattern which changes very little when plotted at different frequencies.
A directional microphone’s ability to reject much of the sound that arrives from off-axis provides a greater working distance or “distance factor” than an omni. As Fig. 10 shows, the distance factor (DF) for a cardioid is 1.7 while the omni is 1.0. This means that if an omni is used in a uniformly noisy environment to pick up a desired sound that is 10″ away, a cardioid used at 17″ from the sound source should provide the same results in terms of the ratio of desired signal to ambient noise. Among other microphone types, the subcardioid should do equally well at 12″, the hypercardioid at 20″ and the bidirectional at 17″.
If the unwanted noise is arriving from one direction only, however, and the microphone can be positioned to place the null of the pattern toward the noise, the directional microphones will offer much greater working distances.
From a distance of two feet or so, in an absolutely “dead” room, a good omni and a good cardioid may sound very similar. But put the pair side-by-side in a “live” room (a large church or auditorium, for instance) and you’ll hear an immediate difference. The omni will pick up all of the reverberation and echoes – the sound will be very “live.” The cardioid will also pick up some reverberation, but a great deal less, so its sound will not change as much compared to the “dead” room sound. (This is the “Distance Factor” in action.)
If you are in a very noisy environment, and can point the microphone away from the noise, a comparison will show a better ratio of wanted to unwanted sound with the cardioid than with the omni.
Now, let’s repeat the comparison, but this time with the microphones very close to the source (a singer, perhaps). As you get within about two inches, you’ll notice a rising bass response in most cardioid microphones. This is known as proximity effect (see page 12), a characteristic that is not shared with the omni microphone used for comparison.
Selecting a Polar Pattern
Whether you should select a directional or omnidirectional microphone can depend on the application (recording vs. sound-reinforcement), the acoustic conditions, the working distance required and the kind of sound you wish to achieve. Directional microphones can suppress unwanted noise, reduce the effects of reverberation and increase gain-before-feedback. But in good acoustic surroundings, omnidirectional microphones, properly placed, can preserve the “sound” of the recording location, and are often preferred for their flatness of response and freedom from proximity effect.
Omnidirectional microphones are normally better at resisting wind noise and mechanical or handling noise than directional microphones. Omnis are also less susceptible to “popping” caused by certain explosive consonants in speech, such as “p,” “b” and “t.” Serious recordists will undoubtedly want to have both types of microphones available to be ready for every recording problem.
When miking must be done from even greater distances, line or “shotgun” microphones are often the best choice. Line microphones are excellent for use in video and film, in order to pick up sound when the microphone must be located outside the frame, that is, out of the viewing angle of the camera.
The line microphone uses an interference tube in front of the element to ensure much greater cancellation of sound arriving from the sides. Audio-Technica line microphones combine a directional (“gradient”) element with the interference tube to increase cancellation at the rear as well.
Figure 11: Line + Gradient Microphone
As a general design rule, the interference tube of a line microphone must be lengthened to narrow the acceptance angle and increase the working distance. While shorter line microphones may not provide as great a working distance as their longer counterparts, their wider acceptance angle is preferred for some applications, because aiming does not need to be so precise. (Some A-T shotgun mics employ an exclusive design* that provides the same performance from an interference tube one-third shorter than conventional designs.)
*U.S. Patent No. 4,789,044
Proximity effect can either be a blessing or a curse, depending on how it is used. A singer can get a deep, earthy sound by singing very close, then change to a more penetrating sound by singing louder while moving the microphone away. This kind of creative use takes some practice, but is very effective. On the other hand, singing at the same volume (with no special effects desired) and moving the microphone in and out will create problems of tonal balance, apart from changes in overall mic level. Some performers also like to work very close at all times to “beef up” an ordinarily “light” voice.
Proximity effect can be used effectively to cut feedback in a sound reinforcement situation. If the performer works very close to the mic, and doesn’t need the extra bass, an equalizer can be used to turn down that channel’s bass response. This makes the microphone less sensitive to feedback at low frequencies, since it is now less sensitive to any low-frequency signal arriving from more than a foot away. (This equalization technique also will help reduce the effect of any handling noise.)
Feedback is simply a condition in a sound-reinforcement application when the sound picked up by the microphone is amplified, radiated by a speaker, then picked up again, only to be re-amplified. Eventually the system starts to ring, and keeps howling until the volume is reduced. Feedback occurs when the sound from the loudspeaker arrives at the microphone as loud or louder than the sound arriving directly from the original sound source (talker, singer, etc.).
The right microphone will reduce the problem. A microphone without peaks in its response is best, as feedback will occur most easily at the frequencies where peaks exist. While a good omni might work well in some situations, a cardioid is almost always preferred where a high potential for feedback exists. When the loudspeaker sound comes primarily from a single direction (rather than mainly reflected from all the walls, ceiling, etc.), the null of a cardioid (or other directional pattern) microphone can be aimed to minimize pickup of the speaker’s sound.
Distance is also a factor. Moving the microphone (or speaker) to lengthen the acoustic path to the loudspeaker can often reduce feedback. Bringing the microphone closer to the desired sound source will also help. And in general, the microphone should always be located behind the speakers.
Current Choir Microphones
- Designed for suspension over choirs, instrumental groups and theater stages
- Wall/ceiling plate power module permits permanent installation in standard metal U.S. single gang electrical box
- Superior off-axis rejection for maximum gain before feedback
- UniGuard® RFI-shielding technology offers outstanding rejection of radio frequency interference (RFI)
- UniSteep® filter provides a steep low-frequency attenuation to improve sound pickup without affecting voice quality
- Accepts interchangeable elements to permit angle of acceptance from 90° to 360°
- Low-profile design with low-reflectance finish for minimum visibility
What Exactly Is a Line Array?
Early line arrays, such as the 1967 Shure Vocal Master system that featured the VA300-S Speaker Column, a “highly directional, wide range, line-radiator” (according to Shure’s manual) didn’t initially catch on, but they’ve steadily gained popularity over the past few decades in large sound-reinforcement applications where achieving balanced coverage is a considerable challenge. What is a line array? It is a series of loudspeakers that covers the same frequency range and is stacked above one another.
The key to a line array is that the speakers face slightly different vertical angles, allowing them to consistently cover a greater depth of field than a single PA speaker can. The effect is that people experience similar sound whether they’re in the back rows, the middle, or the front of the venue, providing a better experience for everyone in the venue. If you’re working in larger spaces and need reasonably high SPLs (Sound Pressure Levels) and clarity, then this is one option you should definitely consider.
How Do Line Arrays Work?
How line arrays work can be a fairly deep discussion. Instead of going down the entire theoretical rabbit hole here, we’ll explain how line arrays work in plain English, starting by understanding how a typical speaker disperses sound over distances.
The Inverse Square Law
The inverse square law tells us that SPL drops by 6dB each time the distance doubles from a point source of sound in a free field of intensity (meaning no boundaries). This is the behavior we’re normally used to with speakers, though there are many nuances to it.
The inverse square law assumes the speaker is radiating omnidirectionally. Except at very low frequencies, this is rarely the case. However as distance increases, even a typically directional loudspeaker (e.g., 90° horizontal dispersion by 90° vertical dispersion) acts like a true point source (i.e. omnidirectional) with respect to how the inverse square law applies.
A line array functions as what’s known as a line source, and therefore will not fall off in level by 6dB each time the distance doubles. Theoretically, it would only drop by 3dB per doubling, but in practice the results aren’t quite that good for a myriad of reasons that are mostly beyond the scope of this writing. However, the overall effect of speakers with line-source dispersion is that you can put more sound in the back of a hall or outdoor space, without requiring the high output level at the front of house speaker position that you’d need with single PA speakers with wider dispersion patterns that are closer to true point-source speakers.
Achieving Line-source Dispersion
Phase cancellation is usually one of the things you try to avoid in a sound system, yet it plays a central role to the way line-array speakers work together to provide a system of speakers with narrow vertical dispersion characteristics. Even with advanced speaker cabinet designs to shape the vertical dispersion, there’s still plenty of natural overlap between speakers in a line array. However, each speaker is at a slightly different distance from the audience, which introduces a small degree of phase cancellation. By introducing a small amount of additional delay, you can fine-tune these phase differences to reduce each speaker’s vertical dispersion.
A Few Important Caveats
While applied phase cancellation can shape the vertical dispersion of the speakers in a line array, their horizontal dispersion is not affected. So in effect, an individual speaker in a line array may wind up with a 90° horizontal dispersion by only a 20° vertical (for example). Also, even though phase cancellation can achieve a line-source distribution and dramatically improve long-distance coverage, as distance increases, even line arrays begin to take on point-source characteristics and succumb to the -6dB per doubled distance of the inverse square law.
There are limitations and caveats to a line array’s ability to approximate a line-source function. First, the overall top to bottom length of the array determines the lowest frequency that will behave accordingly. This is simply because as the wavelengths get longer, the relevant time arrival distances at the listening position must be greater to achieve the effect. That necessitates a longer array. At the other end of the spectrum, the wavelengths become so short that the drivers are too big to be placed close enough together, so the relative phase differences become too great to achieve line-source function. In those cases, waveguides (horns) are used to achieve enough directionality to approximate something between a point-source and line-source function.
Line arrays are inherently helpful in acoustically challenging spaces because you can control their vertical dispersion and reduce reflected sound. Keeping sound off of ceilings and floors is a great start, and then choosing speakers with horizontal dispersions that will help you keep excess sound off of the sidewalls gives more benefit.
Because vertical dispersion per speaker is so tight, you can effectively think in terms of dividing the listening space into sections from front to back, with each front section covered by only one or two speakers, while more speakers may cover the rear sections. This idea gives rise to the popular J-shaped line array so commonly seen in concert halls and outdoor venues.
The exact shape will vary depending on the layout and dimensions of the area needing coverage. You can see how this configuration will inherently put more sound toward the rear of the hall relative to the front rows, which helps with the overall coverage consistency. You can manipulate this further by regulating the power (wattage) delivered to each speaker cabinet — a technique commonly referred to as shading.
Line-array Challenges and Limitations
Though line arrays can be helpful in solving the problems of certain spaces, they do have complex limitations. In addition to the challenge of creating an array length long enough to control lower midrange frequency ranges, line arrays can sometimes suffer from weird anomalies such as certain frequencies lobing into areas directly above and below the array. In a poor acoustic space, this can be extremely problematic, and if your vocal mic is directly beneath a line array, you may experience feedback issues.
At the very least, a proper array requires a rigorous mechanical and electronic setup procedure. Because of the complex interactions involved, everything matters — the exact angle of each cabinet, the exact crossover points, the individual driver delays, and sometimes even the filter settings have to all be exact to achieve the best performance.
Even at their very best, line arrays are unlikely to deliver the purity of sound that high-quality single driver or single two- or three-way cabinets offer, which is one reason why they’re much less likely to be found in smaller spaces where a small number of speakers are more appropriate. Cost and space are considerations too, of course.
Small Line Arrays
While line-array technology most often appears in large sound-reinforcement applications, there are a number of companies, such as Bose, Fishman, Turbosound, and others, that offer small, personal PA systems that use miniature arrays of small speakers (typically 2″–4″) to create the same line-source-dispersion effect. When used correctly, these systems can provide excellent coverage in small venues.
5 Most Common Sound Problems with Your Church Sound System
So Many Speakers, So Little Sound.
By far the most common problem we encounter in church sound systems is uneven coverage of the seating area. Some seats are way too loud or harsh, while it may be dull or impossible to hear clearly in others.
The reason this problem is so common is that good design takes a lot of engineering and experience. There are many factors that play into consistent sound coverage including speaker selection and placement, number of speakers/amps, and even the layout of the seats can be a factor. In fact, it’s almost impossible to get it right without sophisticated computer modeling. However, it can be done.
We often tell pastors that you never know where a guest is going to sit, so it’s absolutely critical that every seat is a good one.
If You’re Covering Your Ears, It’s Already Too Late.
The second most common complaint is the unpredictable, screaming feedback that can cause jarring disruptions in the course of any service.
Technically speaking, feedback is caused by a positive gain loop between a microphone and a speaker. Without digging too deeply into the nerdy details, the microphone picks up the sound from the speaker, and then the speaker amplifies the sound back into the microphone until the system overloads. The result is that too-familiar ear-piercing screech we’ve all experienced at one time or another. In most cases, feedback issues can be resolved with a combination of equalization, volume adjustment and speaker placement, but on a stage full of monitors, it can be a real challenge to figure out which one is the culprit. To make matters worse, it’s often not just the equipment. Poor room acoustics can also be a contributing factor.
With the tools available for modern design and production, there’s really no excuse for feedback from your sound system.
Balancing the Mix
But Where Did The Vocals Go??
You can see them singing on stage, so why can’t you hear the background vocalist? The third most common problem with church sound systems relates to your operator’s skill level with balancing the mix. Do they really understand how to dial-in and manage the right mix for your worship team in your sanctuary?
Perhaps the reason this problem is so rampant is that the solution is not naturally intuitive. To the untrained technician, balancing competing instruments and vocals may seem as simple as setting each channel at an equal level. However, when channel volumes are competing, it’s like all of the cars on the freeway trying to drive in the same lane at the same time.
The solution lies in finding the right place for each and every instrument in the mix. For most worship music this usually means panning instruments across the stereo field, controlling the dynamics, carving out sections of the frequency spectrum with EQ to make sure there is room for each element and then riding the faders to adjust for changes in level over time.
Many sound techs try to balance everything without this understanding, and that’s why you often can’t hear the vocals.
Hint: If Your Acoustic Guitar Sounds Like a 5-String Banjo, You’re Probably Doing It Wrong.
Good EQ is about tailoring the individual sounds to fit together as a cohesive whole. Unlike Balancing the Mix, the EQ puzzle doesn’t deal with volume so much as frequency.
The human ear is capable of perceiving frequencies between the 20Hz and 20,000Hz range. If that’s Greek to you, it means there’s an almost infinite number of choices with which you can adjust the character of the sound. Easy, right?
The most helpful way to think about EQ, in general, is to remember that every sound needs to be complementary to the others based on how they work together as a whole.
There’s an art to contouring the bass guitar to make room for the low-end of the kick drum. It takes a practiced skill to manage the high-frequency content of the guitars and cymbals to ensure that the airiness of the vocals can still cut through. Electronic keyboards can cover the entire frequency spectrum, often stepping on vocals and other instruments if they aren’t managed correctly.
For a quick illustration on how each instrument’s EQ fits into a well-designed mix, download the FREE Instrument Frequency Chart, which demonstrates the ranges of common worship instruments. At a glance, you can see that most instruments are in direct competition with the range of the human voice.
This, of course barely scratches the surface of EQ, but hopefully, these pointers will serve as a launchpad as you continue to explore this delicate art.
How to Prevent Feedback in Church Sound Systems
Feedback in a worship service or concert is one of the most overt, embarrassing moments for a sound engineer who is trying to mostly be invisible behind the scenes. Besides being a blatant interruption, there’s precious little you can do about it after the event has started — oh, and by the way, the whole room is now staring at you. It’s a hot seat nobody wants to be in. So what can you do about it? Be prepared. Here are some pointers on how to avoid feedback!
What Causes Feedback?
What is feedback? Feedback happens when the output of a sound system is “fed back” into the input of the sound system. It’s sometimes called “howl,” because in live sound that’s what the problem sounds like. What’s happening is that your microphones are hearing the output of a speaker. The mic then loops the sound back into the system and amplifies it again, and again, and again until certain frequencies are accentuated in an exponential fashion, and that’s the screech/squeal/howl that ultimately makes your life difficult. The objective, then, is to break (or ideally to AVOID) the loop!
Know Your Mics
The best solution for feedback is a physical one. If you have a mic that is too close to, or maybe even pointing at, a speaker, don’t EQ to avoid feedback. MOVE THE MIC! A sound engineer trying to fight physics with EQ is like David versus Goliath without a sling; give yourself a fighting chance!
The best way to do this is to know where the blind spots (rejection) on your mics are. Each mic has a pickup pattern. Cardioid mics excel at rejecting sound from the rear, but supercardioid and hypercardioid are better at rejecting sound from the sides. If you have a singing pianist, maybe a hypercardioid mic is a great choice, because you can put a stage monitor to his or her side. Conversely, if the stage monitor is facing the singer, meaning it’s right behind the singer’s mic, a cardioid microphone would perform much better.
If you really can’t maneuver your way to success (which happens sometimes), you can eliminate the feedback sources by eliminating the stage monitors, which can be done by using in-ear monitoring.
Notch Out Those Problem Frequencies with EQ
If you’re using stage wedges, this process is the show prep you need. It will improve your quality of life as a technician. It’s a method of applying specific EQ cuts to the most problematic frequencies to stabilize your system and prevent feedback. These steps should be done before rehearsal, or soundcheck even, when no one else is in the venue. The whole process is best done BEFORE you’ve sound checked anything, because your channel EQ will mislead you. If you’re having feedback problems, resetting the board and starting with this technique may be a good idea anyway.
Let’s start with the routing. For every mix that you’re sending out to a speaker, insert a graphic equalizer (usually 31-band) in between the console and the amplifier (including the mains and the monitors). On a digital mixing console, an EQ is very likely included in the console itself, but you might have to make a digital patch to include it in the signal chain. On an analog board, you’ll use a hardware EQ that you’ll probably keep in a rack next to the board. Make sure all your monitor mixes are PRE-fader, not post-fader. This will keep the adjustments you make to the main mix separate from what your performers are hearing.
You’ll need a Real-Time Analyzer (RTA) of some sort. These are available for smartphones and tablets and most digital consoles have them built in. An RTA displays how much energy there is at specific frequencies, so you can see what frequencies are popping up (louder). It’s easier to play “sonic whack-a-mole” when you know where the moles are, so unless your ears are so well trained that you know exactly what 325Hz sounds like, it’s a must-have.
Next, take each mix, one at a time, and mute every other output. For example, let’s use the worship leader’s monitor mix. Bring the master output of that mix all the way down. Take some notes on where the levels in the mix are before proceeding. Then bring every input channel up to unity (0 on the fader) in that mix. So if his or her mix is on Aux A, you should now have whatever is in his mix at unity for that mix, but there still is no output because the master is all the way down.
Turn up the master SLOWLY until feedback occurs. Turn it down quickly because feedback can damage your system. As the feedback happens, look at the RTA to see what frequency just popped up the highest. Go to your EQ, and cut that frequency by 3dB. Push the system up again until feedback. Rinse and repeat. You’ll do this until either A) your system is bulletproof; you can push it higher than you would ever need in a performance situation, or B) you’ve cut one frequency by 12dB. If that’s the case, you may need to re-evaluate. The leading suspect here is a physical problem somewhere, usually a mic pointing at a speaker, or a serious acoustic issue that should be addressed with treatment. Now that you’ve found and suppressed the problem frequencies, reset the levels in your mix as they were.
Repeat the whole process for each different mix. You should now have a VERY stable system.
Take Out the Low-frequency Trash
A great practice to establish is using your highpass filter (HPF) as much as you can. A highpass filter cuts low frequencies and allows the high frequencies to pass. On an analog board, it’s usually a button near the preamp gain (if it’s fixed frequency) that you can engage, and it ranges from 75Hz–100Hz, which means everything below that frequency is cut out. Some consoles (and most digital boards) will allow you to vary the highpass frequency. This is a HUGE tool for mixing defensively.
Feedback starts in the lows. You probably have some low-end sources in your mix like bass and kick drum, and you don’t need to cut lows out of those, but your lead vocal doesn’t really need to be in your subwoofer. Engaging the HPF on those channels removes unnecessary frequencies that are below the vocal range from your mix. If you’re on a digital mixing console, do the following in soundcheck: start with the HPF at 20Hz and increase the frequency until you start to hear it cutting into your source in a noticeable way. Next, roll the frequency back down a bit. Every frequency you dumped just cleaned up your mix and bought you more gain before feedback. You also just gained some power in your PA system that you can use for sub-bass that you want your audience to hear.
Repeat this for every channel on a digital board. Leave the frequency really low on a bass sound source, such as 40Hz for a bass guitar, for example. On an analog board, do this on every channel that isn’t a big low-end contributor (bass, kick, maybe a floor tom or a keyboard). This way, none of the frequencies you’ve cut can be used against you by feedback!
Defensive Panning and EQing
If you’ve done all your prep work and your vocalist still insists on holding a mic facing right into a speaker, there are a few last-ditch tricks you can try to get you through the service until you can teach how and where to hold a mic. Lapel mics are often the worst offenders.
Muting or pulling back volume can help, but it can also lead to more problems, not the least of which is that your audience is no longer hearing the vocalist. A better idea is to use panning, if the situation allows. NOTE: This trick will only work for sound systems that are running in stereo. This option is especially great for lapel microphones.
For instance, say the pastor is hammering home the point of the sermon. His volume is a bit higher than it was in soundcheck. (But you anticipated that and left him some room before clipping, right?) As the pastor walks to the edge of the stage, he gets closer to the speaker, and you hear it start to ring. Instead of bringing down his level or muting him, pan him to the other speaker. The volume difference to the audience isn’t huge, and most people won’t even notice.
One other tricky situation happens when you can tell there’s feedback in your performance, but you can’t tell where it’s coming from. Look on the console’s input meters, usually near the input faders. You’re looking for anything that is clipping or significantly louder than it should be. If you can nail down what channel or mix is the problem, but you still can’t hear the feedback clearly enough to know where it’s feeding back, just start cutting low-mids out of that channel or mix. I usually start at 325Hz. This is definitely a last-ditch attempt to save a bad situation, and after the end of the service, you’ll want to figure out what needs to be fixed for a more permanent solution. In the meantime though, you’ll get through it.