With the understanding of the broadest groups of mixing, now we’ll move into individual instruments and how one might work with them. Know that these are only very general recommendations, and the best method of mixing for your particular situation is almost assuredly not mentioned here, but hopefully I can give you a good place to start from.
In all of this, an important thing to remember is: “If you can’t do it with an SM57, you shouldn’t be doing it.” Even when specific microphones are mentioned, or specialty microphones are mentioned in general, you can still get good results with whatever you have on hand. The aforementioned Shure SM57 is one of the most common microphones in existence, both because of its durability and versatility, and the quote above is a constant reminder to keep things simple. When it comes to mixing, the tools are just the tools — it’s what we do with them that’s important.
Vocals
The number one, absolute most important thing to think about with vocals is intelligibility. If the listener can’t understand what the vocalist is saying (a few specific genres aside), what’s the point? There are several standards to define speech intelligibility you can look up if you feel so motivated, but, in general, it comes down to consonants — specifically sibilants. These usually live in the 2kHz~8kHz range, and are the s, t, ts, and other related sounds.
Getting It
For handheld microphones, the best option is for the vocalist to have their microphone about a thumb’s length away from their mouth. This is to make the best use of something called “proximity effect” present in microphones with a cardioid or unidirectional pattern. This effect increases the low frequencies the microphone picks up and is an artifact of how unidirectional microphones are constructed. Keeping the mic a little bit away from the source will smooth out the low end response while still reducing the possibility of feedback or bleed from other microphones.
If you’re using a side-address mic, as you would in a studio or radio setting, the basic idea is the same, but it’s less important to be so close to the microphone. As long as the vocalist is speaking or singing into the mic within a foot or so, you’ll be fine.
In the Mix
Because the human voice is something we’re so used to hearing, we have a natural tendency to know what it’s supposed to sound like. This helps us on one hand because we instinctively know what our end goal should be. The downside is if the sound is just a little bit off, everyone knows it!
Generally you want a highpass filter between 60~120 Hz depending on the voice to cut down on handling noise (especially when using a handheld mic). Otherwise body and muddiness is ~250Hz and nasalness is ~1kHz. Those and intelligibility (2kHz ~ 8kHz) are the most important frequency ranges for vocals. Lead vocals usually benefit from a little bit of reverb and delay, where backing vocals are well served with chorus to fill out the sound and make them found fuller, and a little more reverb than the lead vocal to move them farther back in the mix.
Drums
There are two basic ways to think of drums, specifically the drum kit, when mixing them: is it a single instrument or many separate instruments? That question is the beginning of the difference between area-micing (traditionally used most often in jazz and related styles) and close-micing (traditionally used in just about everything else).
For area-micing, your primary microphones are your overheads and room mics (if in a studio setting), and the individual mics on any drums enhance the sound through those microphones. In close-micing, the primary microphones are the individual mics on the drums, with the room mic supplementing and the overheads focused on the cymbals.
Kick Drum
Most of the time, the kick drum is quite literally the first channel on a console. While there used to be a good reason for this, nowadays it is a combination of tradition and a reflection of the kick drum’s importance to a mix.
Getting It
The most common way to mic a kick drum is a speciality microphone (such as an AKG D112, Audix D6, Shure Beta 52, and many others) in a hole cut in the resonant or audience side head of the kick. Moving the mic closer to the batter or drummer side head will give you more attack, while moving the mic closer to the resonant head will give you more boom.
If the kick doesn’t have a hole in the resonant head, a microphone on the batter head just above where the beaters are will give you full sound with lots of attack, while one in front of the resonant head will give you a boomier sound more suitable for jazz or similar music.
For extra control, I like to use two kick mics, one on the batter head and one on the resonant head, and use each mic to give me a part of the sound I’m looking for. Make sure to check the phase relationships between the microphones, as one of them will almost certainly need to be “phase flipped”. Most mixers these days have a phase switch button, making this much easier today than when you’d need adapters on hand for these situations.
In the Mix
When mixing a kick, the low frequency (40~100Hz) is the “boom”, the low mid (~250Hz) is the “boxiness” and the high mid (~2kHz) is the “click” or beater noise. Boosting or cutting in these frequencies will create the sound you need for your mix. A short delay on the kick can also add a bit of “attack”, giving body to the high frequencies and reducing the low frequencies through phase cancellations.
Snare Drum
The snare is usually what gives the “backbeat” of a song, happening on the “two” and “four” of a song. It’s what people will clap to if you’re in a situation in which people would clap to the music. With the kick drum, the snare usually comprises the foundation element of a mix.
Getting It
The most common way to mic a snare drum is with one mic toward the edge of the drum pointed towards the center. For a snappier sound, add a second mic in the same position on the underside of the drum. As always when using two microphones on a drum, reverse the phase of one to avoid cancelling out low frequencies.
In the Mix
The body of a snare drum is around the low mid (~350Hz); the snap of a snare is in the highs (2kHz~6kHz). A short delay can add “attack”, much like with a kick drum, and reverb can give a snare some extra body. Another rather strange option, most viable in a studio, is to use a little bit of distortion on a snare to add some “fuzz” and fill out the snare sound.
Toms
Toms are most often used in drum fills, although they sometimes are used in a way similar to a hihat to move a beat along, either as a foundation or rhythm element. Toms generally come in two styles — mounted or floor — based on how they are supported. Floor toms have legs that sit on the floor, and mounted toms generally attach to stand or to a bass drum.
Getting It
Like a snare, toms are best mic’d from the edge of the drum pointing toward the center. If they’re not tuned correctly, toms are most likely to have strange overtones to them. If the drummer allows it, a “RemO” ring or a bit of something called “Moon Gel” (or, in a pinch, one of those sticky hands from old school quarter machines) can cut down on the overtones.
In the Mix
Like the other drums, body is in the low mid (~200Hz for higher toms and ~120Hz for lower toms), and attack is in the higher mid (~2kHz). Toms are the only drums that benefit from pan to give them separation from the kick and snare whose frequencies they somewhat overlap with. Higher toms should be panned to one side and lower toms to the other, based around the side of the drummer they are on.
HiHat
HiHats are will either be a foundation or rhythm element depending on their use. Hihats or ride cymbals often provide a regular pattern over whatever the kick and snare are doing.
Getting It
While the hihats will generally get picked up quite well by overhead mics, it can be helpful to use a dedicated microphone, especially if the drummer does a lot of subtle playing on the hihats. A condenser mic (one requiring phantom power) is usually best for this, pointed down at the hihat about halfway between the bell and edge of the the hihat. Make sure the mic is high enough that when the hihats are open, the top cymbal doesn’t hit the microphone.
In the Mix
Hihats can easily sound like a metal trash lid with the wrong mic placement or EQ. To get a good clean “sizzle” without too much bleed, a high pass filter is very helpful, but not so high that the hats lose “body”: ~250Hz is usually high enough. Also make sure to pan the hihat to the appropriate side.
Ride Cymbal
Rides are used similarly to hihats, giving a different kind of regular pattern depending on the style of music and section of a song.
Getting It
It isn’t usually necessary to mic a ride cymbal, but depending on the style of music and the particular drummer. If you do, mic it similarly to the hihat, around the middle of the cymbal. For more “ping”, put it closer to the bell; for a “fuller” sound, put it closer to the edge. Don’t put the mic too close to the ride, as you will get an unpleasantly muddy sound from such a large cymbal.
In the Mix
The ride will be picked up by the overheads even better than the hihat, so you usually want this mic to not carry the ride’s sound, but instead to supplement what’s already there. The high-mid frequencies carry much of the “attack” of a ride, while the low-mid frequencies will give you more “wash”
Overhead Mics
This is one of the most complicated parts of micing a drum set, and also one of the most varied. There are many ways to set up overheads, and I’ll highlight some of the more frequently used methods. The actual mixing of overheads depends on the overall theory of how you’re approaching your drum mix: the overheads are either the meat of your sound, acting as way to collect the sound of the whole kit or primarily used to pick up the cymbals, allowing close-mics to provide the sound of the individual drums.
Getting It
Here are a few of the more popular ways to do drum overheads; it’s best to experiment and find the one that works for you.
Single Mic
This is by far the simplest method. One mic, either omnidirectional or unidirectional, placed roughly over the drummer’s head. While you lose any stereo imaging, when in a situation of limited channels or equipment, or you just don’t need that much drum reinforcement, this is a great place to start.
Spaced Pair
The split pair is the next simplest. Two cardioid mics are used, usually of the same type, placed high above the drum set pointed straight down. The advantage is it’s easy to set up, but you can run into phase issues, especially with a kick and snare, as the microphones are different distances from those two sources. Each microphone can be panned to their respective directions to limit this effect, but it can’t be escaped entirely
“Glyn Johns”
This is one of the earlier methods of reducing the phase issues inherent in split pair micing. First, a microphone is placed directly above the snare (or above the kick beater pointed at the snare). Measure the distance between the center of the snare to the microphone, either with a tape measure, piece of string, or mic cable (in a pinch). Place a second microphone on the right side of the drumset, at around a 90-degree angle to the first microphone, exactly the same distance away from the center of the snare as the first microphone. This arrangement was first used to record Led Zeppelin by the legendy Glyn Johns, cutting down on phase problems while maintaining a good stereo image. (As an aside, on those Led Zeppelin albums recorded with this method, the only close mics used were a kick and snare mic.)
Recorderman
This method is about halfway between the Glyn Johns and Spaced Pair methods. Like in Glyn Johns, a microphone goes directly above the snare pointed down. Measure the distance between the snare drum, microphone, and the kick beater (a piece of string and a little bit of masking tape is great for this; tape the string to the snare and kick head and use the tape to mark on the string where the microphone is). Make an arc with this string so that it makes a 45-degree angle from the first microphone. Place the second microphone at your tape spot and point it at the snare drum. This method is a good balance between stereo image and sound quality, and is the one I tend to use when I use two overhead mics.
X-Y
This is another method of reducing phase issues with two mics. Instead of careful measuring, the diaphragms (or, in layman’s terms, the grill-covered part that actually picks up the sound) of two microphones are set up as close together as possible without actually touching at a 90-degree angle to each other. The pair of microphones are placed in around the middle of the kit (roughly above the kick beater) such that each microphone is pointed roughly 45-degrees from “down”. This arrangement nearly eliminates phase problems, though the trade-off is a reduced stereo image. Note that the microphone on the left actually picks up the right side of the drumset, and vice versa.
Weathervane
This isn’t a very common method of micing drums, but it is my preferred method. It works best when the ride cymbal is placed closer in, as often happens with 4-piece drum kits. This method requires three microphones, and is similar to the Glyn Johns method described above. Add a kick mic, that’s all you need to get a solid drum sound. It’s only real downside is how complicated it is to set up, but once you get used to it, it goes a lot faster.
First, the center microphone (this works best with a ribbon mic, but any mic will work) is placed 3~4 feet over the ride cymbal in line with the center of the snare drum and the kick beater, pointed at the snare. Then take a piece of string and measure the distance from that microphone to each tom, making sure the mic is as close to equidistant between them as possible, while still being on that center line. Next, tape one end of another string to the center of the snare drum, stretch it with your finger to touch the center mic, and tape the other end to where the beater hits the kick drum.
Next place the second microphone directly over the snare, pointed down. Find its height by moving the string (still taped to the snare and kick) directly over the snare by letting it slide over your finger. This (and the next step) makes sure you don’t accumulate phase problems between the three microphones. Set that string aside and, with another piece of string doubled over on itself, measure the distance between the snare mic and the center mic. Tape one end to each microphone and either put a small mark on or wrap a small piece of tape around the exactly middle of the string. When you stretch out the string, you’ll get an equilateral triangle between the middle of the string and the two microphones.
Finally, take the snare/kick string and stretch it out with your finger. Slide your finger under that string and move it so your finger meets up with the center of the center-mic/snare-mic string. It will be a little bit lower than the other two microphones. This is where you’ll place the third mic, also pointed at the snare. Pan the snare mic one way, the third mic the opposite way, and the center mic in the middle, and you’re done.
For a video explanation, check out Drum Overheads – The Weathervane Method on YouTube.
In the Mix
If you’re using overheads as your the main microphones for your drum sound, there isn’t much that needs to happen. Usually a little bit of reverb is all you need and your drums are good to go.
If your overheads are mostly picking up cymbals, you’ll want a highpass filter (250Hz~500Hz) to get most of the drums out of them, and then dropping the high frequencies (~10kHz) down a bit to remove some of the apparent harshness that’ll leave behind. Then, as before, a little bit of reverb to fill out the sound.
Room Mics
When mixing live, there is absolutely no point to room mics, but in the studio they’re quite handy. If you have the luxury of recording drums in a room that sounds good, they’re essential. Even if the room doesn’t sound good you can “fake it” using something called a convolution reverb (which is beyond the scope of this reference).
Getting It
Before setting up your room mics, make sure they follow the same center as your overheads. For example, if you’re using a spaced pair of overheads, the center is approximately the kick drum, but if you’re using the Weathervane method, it’s the line between the kick and the snare. Your room mics should be set up in stereo so that the sound of the drums is picked up along that ceterline, wherever in the room sounds best.
In the Mix
Room mics are meant to add some life to drums that are close mic’d so they sound more “live” on a recording. They’re more or less a natural reverb, so they’re best treated as such — they should be missed if they’re off but not noticed if they’re on. (Although, if you’re wanting to get a bit creative, room mics run through a guitar amp simulator and blended subtly back into the mix can make for an incredibly aggressive drum sound.)
Putting It All Together
Once each individual part of your drum sound is how you’d like it, it’s a matter of balancing it all out so that the sound of many individually mic’d pieces becomes one instrument. Because of how many parts are involved, drums can be very complicated at first, but like anything else, with practice they are a lot easier to deal with. Reverb, compressors, and gates can all be used to tie a drum kit together and get you to the sound you’re looking for.
Synthesizers
Synths are the stereotypical pad element in a mix. They’re what fill up space and create the atmosphere the musicians want out of a song.
Getting It
Almost always, a synth or keyboard will connect to a stereo DI box. In the extremely rare case in which you’ll need to mic a keyboard speaker (the most common would be a Leslie cabinet), use the same techniques as a stereo guitar amplifier.
In the Mix
As synthesizers are usually the pad element, they are usually found panned hard right and left in a mix and EQ’d to remove some of the much midrange. This keeps them out of the spaces the lead and foundation instruments take up, giving you a cleaner mix. If the synthesizer is operating as a different mix element, of course, it’ll need a bit different treatment, and you can use techniques similar to that used for whatever element it is emulating.
Synthesizers
More and more often in today’s environment, it is more common to experience a piano coming from an electronic keyboard, but especially in churches or performance halls, pianos can still be found.
Getting It
There are almost as many ways to mic a piano as there are for drum overheads. Here are a two of the easiest.
Single Mic
When using one mic, it’s best to use an omnidirectional mic in the middle of the piano keyboard (right around “middle C”). On an upright piano, put it about a foot and a half away from the wooden back (the “soundboard”). On a grand piano, place it around where the bass and treble strings cross. For more attack, move it closer to the hammers; for more fullness, move it farther from the hammers.
For a surprisingly good grand piano sound (for what it is), grab an SM58 and a small cloth, lay the SM58 on the cloth on the metal frame of the piano so the ball of the microphone is in one of the holes in the metal, with the mic pointed towards the hammers. Experiment to find the best location for your piano. A big perk of this technique is you can leave the lid closed and there is almost no bleed into the piano mic from outside.
Spaced Pair
If you can use two mics for a piano, a spaced pair is quite easy. Place two microphones, pointed down, 6 to 12 inches above the strings just behind the dampers and at least 3 times farther apart than the distance from the mics to the strings (this cuts down on phase problems).
In the Mix
Pianos can serve as any number of mix elements, and should be mixed depending on what role they serve. For EQ, generally, low is “fullness”, low midrange is “boxiness”, high midrange is “attack”, and high is “presence”. Pianos usually benefit from light reverb.
Guitars
Guitars serve a variety of roles in music. Depending on the song, or even what part of a song, the guitar can be every single element. Whatever element the guitar is serving as will mean a slightly different method of putting it in the mix, and so I’ll be covering general principles that can be adapted to any situation.
Getting It
While Acoustic and Electric guitars are very similar, from a mixing perspective actually getting the sound through the system to reinforce it are quite different from one to the other.
Acoustic
The simplest method of hooking up an acoustic guitar is if a pickup is installed, in which case a DI box is the best option for live sound. Even if the guitar player is using an acoustic amplifier on stage for a monitor, usually the direct signal from the guitar itself will give you the best result. The only exception is if the guitar player is using some manner of effects, be it through an amplifier or foot pedals. In this case, you want either a direct out from the amplifier or to take the output of the last foot pedal in the chain.
If the acoustic doesn’t have a pickup or you want a mic to get the tone coming out of the guitar, it’s easiest to use a unidirectional microphone about a foot in front of the guitar pointed at a 45-degree angle to where the neck and body meet. As usual, experiment to find the best placement for the desired sound.
Electric
Contrary to the acoustic guitar, it very rarely sounds good to run an electric guitar — even after effect pedals — into a DI box. Unless the guitarist’s effects setup includes a cabinet simulator, it’s best to mic the amplifier of an electric guitar. Unfortunately, each amplifier will need to be mic’d a little differently depending on the cabinet. Often it’s best to start by asking the guitar player which speaker sounds best!
In general, take a general instrument mic (like an SM57) and point it directly at the speaker, about 2 inches away from the grill and just inside halfway between the middle of the speaker and its edge. This can sometimes be hard to find depending on the cabinet. Adjust the distance, position, and angle of the microphone until you get the sound you want. If the guitar has a stereo cabinet, do the same thing with both the left and right speakers. (Be careful though, most of the time a cabinet with more than one speaker isn’t a stereo cabinet; you only need one mic for a mono cabinet no matter how many speakers it has.)
In the Mix
I usually like guitars to be panned between 20% and 50% left or right to clear out room for my lead instruments. (Unless, of course, the guitar is the lead instrument.) Guitars and vocals occupy very similar frequency ranges, and panning guitars off center makes sure there isn’t as much overlap. For recorded guitars, it is often handy to record the same guitar line twice and pan it in opposite directions to create a bigger sound.
For electric guitars, apply EQ to get the sound you want, but it’s usually good to avoid effects; most electric guitars will already have them by the time they’re in the system. Acoustic guitars do well with chorus and reverb, but make sure the guitar player isn’t already adding those in first. More and more acoustic guitar players are using their own effects and it usually doesn’t sound good to double up.
Bass Guitar
It’d be easy to include the bass guitar in the above section, but it usually fills a different role from other guitars. The bass is typically part of the foundation of a mix. Like the bass singer in an a capella group, the bass guitar is usually what every chord in the song builds from. It is the sustained note that balances out the kick drum’s attack.
Getting It
A bass guitar is very easy to get set up, as most of the time it’ll plug directly into a DI box and into the system. If you absolutely have to mic a bass cabinet, use the same techniques described above for micing a guitar amplifier, except use a kick drum mic instead of a general instrument mic.
In the Mix
The biggest temptation with a bass guitar is to turn up its bass; it’s in the name right? Resist this temptation. More often than not, turning up the bass of a bass guitar will make it sound sloppy. Instead, use the low midrange (~250Hz) to control how full you want the bass guitar to sound. Attack is generally in the high midrange (~2kHz).
Signal Processing & Effects
This section is all about the tools we can use to change how something sounds. The trick is never overdoing it. It’s important to always remember with any of these effects; be it EQ, reverb, or anything else; it should be missing when it isn’t there, but not noticed when it is.
Equalization and Filters
EQ is one of the easiest to understand ways to change the way something sounds. Everyone has played with the bass and treble controls on their car or home stereo right? Even the novice can figure out how it works. With EQ, like everything else, less is more. Most problems that can be solved with EQ would be better solved by changing what mic is being used or where that microphone is. Additionally, it is usually a better idea to cut with EQ and not to boost; it’s easier to remove something undesirable than to try and create something desirable.
When it comes to EQ filter settings, there are 3 basic settings: gain, frequency, and width. Gain is the amount of boost or cut being applied; frequency is the point at which the boost or cut happens; width is the range of frequencies around the center that gets boosted or cut. (Width is often referred to in “octaves” or “Q”. 1 octave is a doubling of frequency and corresponds to an octave in musical notes [e.g. A4 is 440 Hz, therefore A3, an octave below, is 220 Hz.] A 2 octave filter is, understandably, 2 octaves wide; a ⅓ octave filter is ⅓ of an octave wide. “Q” is sort of the inverse of octaves; a higher “Q’ is a narrower filter and a lower “Q” is a wider filter. [e.g. a 2 octave filter has a “Q” of about ⅔, while a ⅓ octave filter has a “Q” of about 4.3. It’s a lot easier to think in terms of octaves and not in terms of “Q”, but many EQs use “Q” for filter width.] Width is a lot more complicated than it seems, but a full discussion how exactly width effects an EQ filter is far beyond what this reference seeks to achieve.)
Shelf
A shelf filter applies gain to all frequencies above (for a high shelf) or below (for a low shelf) the set frequency. Often, shelving EQs don’t have a width setting, instead using a fixed width, but if they do, the width controls how “quickly” the gain rises before reaching the center frequency. Narrow width shelving EQs sometimes will “overcorrect” and give a boost at their center frequency, which can be used for effect.
Bell
Bell filters apply gain to the frequencies around their center according to their width. For example, a bell filter at 1kHz with a 2 octave width will roughly boost or cut from 500 Hz to 2kHz. Generally, to change the way something sounds (e.g. removing muddiness from a guitar), a wider filter is better, but to fix a problem (e.g. a microphone feeding back), a narrower filter is better.
High-, Low-, and Band-pass
These filters do exactly what they describe, although it seems a bit counter-intuitive at first. A high-pass (sometimes called a low-cut) filter will “pass” all frequencies higher than the center and cut all frequencies lower than the center. A low-pass (sometimes called a high-cut) filter will do the opposite, passing frequencies lower than the center and cutting the ones higher than the center. A bandpass filter combines low- and high-pass filters to only allow frequencies between the two center frequencies to “pass”. (A bell EQ filter is a variation on a bandpass filter that also has a variable gain.)
Several of these filters together is called a crossover, which are often used behind-the-scenes in speakers and system processors to make sure only the frequencies a speaker is designed to reproduce are sent to that speaker. In a two-way speaker, this ensures the smaller high-frequency driver doesn’t get overwhelmed by low-frequencies and the larger low-frequency drive only reproduces the low-frequencies it was built to handle.
The high-pass filter is mostly likely the one of all these filters to be used. If I have a high-pass filter with a variable frequency it is in use on every single channel to reduce the amount of low frequency noise in the system, leading to a much clearer mix.
Graphic
Graphic EQs are a collection of bell filters, usually ⅓ octave widths, arranged to cover approximately the entire range of auditory spectrum, from 20Hz to 20kHz. These offer a rough visual representation of the EQ curve produced by them, though, with the advent of affordable digital mixers that clearly show the curve produced by a parametric EQ, graphic EQs are somewhat losing their usefulness.
Echo
This category is based on “reflections” — adding extra copies of a signal to change how it sounds. These methods imitate the same natural processes that make cathedrals sound like cathedrals and certain canyons yell back to you when you say, “Hello!”
Reverb
Reverbs use a number of very short echoes — often too short to hear distinctly — to create the perception of a certain kind of physical space. There are a number of different kinds of reverbs, often called “Hall”, “Room”, or “Plate”. These are electronic simulations of what used to be real items. For instance, a “plate” reverb was originally a metal plate with a “speaker” on one end and “microphones” on the other. (To help you in your trivia knowledge, there are still actual rooms buried underneath Los Angeles studios with speakers and microphones in them.)
Most reverbs have an abundance of settings, from “length” to “pre-delay” and many more. These can be intimidating at first, but most FX processors come with presets to give you a good starting point. The more you use them, the more comfortable you’ll be with the various adjustable settings of each reverb.
Reverb is a great way to add depth to a mix. The more reverb used, the farther in the background something will sound. A good practice with stereo reverb is to pan opposite and a little wider to the “dry” signal. For example, if an acoustic guitar is panned 50% left, its reverb should be panned 75% right. This allows you to create space in your mix without sacrificing clarity.
Delay
Delay is very similar to reverb, except that instead of echoes indistinguishable from each other, delay uses discrete echoes to fill out a sound. I will often use delay instead of reverb with a very “full” voice, as delay will produce a similar result to reverb while not sacrificing clarity.
The most important settings in a delay are feedback and time. The time setting determines when the echoes will repeat. Delay usually sounds best when it fits musically, and most delays have a “tap” feature, which you can use to time your delay to the music. Used well, this means a delay can become a fill element of a mix, “repeating” the lead after a line and filling space in the mix. Most of the time, when I use delay I set the time to match the tempo of the music, unless I am using a stereo delay, in which case I use a “3-to-2” ratio between the left and right times, letting the delay “bounce” between the left and right speakers.
“Feedback” sets how much of the original signal is repeated each echo. When feedback is set to 100%, the delay is turned into a looper — continually repeating the original signal until it is lost in inevitable signal corruption. A feedback setting of 0% will repeat once. I usually like to set feedback between 10% and 50%; this gives you a repeating effect without overwhelming the original signal.
Modulation
Where echo effects change a signal in time, modulation effects change a signal in frequency. This is achieved by either slightly modifying the pitch of the incoming signal, moving the signal slightly in time, or both.
Chorus
A chorus effect simulates the sound of a choir. In a choir, many voices sing together, all slightly different, to produce a particular tone. One of the first artificial choruses was the céleste stop on a pipe organ, a series of pipes slightly out of tune from the others. Other examples are the piano, 12-string guitar, or mandolin; all of which use more than one string to produce a single note.
Chorus can be used to fill out a single instrument to make it sound “bigger” or to subtly cover up an instrument that is slightly out of tune by exaggerating its dissonance. It’s very effective on guitars and background vocals and, used sparingly, will sound quite nice on lead elements as well.
Phaser
Phasers sound similar to choruses, but do so in a very different way. Instead of shifting pitch, phasers pass a signal through something called “non-linear” filters (like a small delay but not consistent across the frequency spectrum), intentionally creating phase problems to produce the desired effect. This interference behaves like a series of very incredibly narrow EQ filters, giving a “sweeping” sound.
Flange
A flanger is like a phaser, except the entire signal is uniformly delayed slightly to produce the desired effect. This gives a much more pronounced version of the phaser’s sound, turning the “sweep” into more of a “swoosh”. While they sound very similar, the measured frequency response of flanging is much more uniform than that of a phaser.
Dynamics
The third general kind of effect alters the signal in amplitude or volume. These can be used to “tighten up” a mix, but, used incorrectly, can cause unexpected problems as well. The two general types, compressors and expanders, work similarly to shelving EQ filters, only instead of working in the frequency domain, they work in amplitude.
These tools have the same basic settings, though sometimes the application is different. “Threshold” is the volume in decibels (dB) at which the compressor or expander starts to work. Ratio is the amount of compression or expansion done to the signal. Attack, Hold, and Release times define how long before the compressor or expander starts working, how long it continues to work when the threshold is no long reached, and how long it takes for the compressor or expander to stop working. “Knee” defines the severity of the compression or expansion. For example, a compressor with a “hard” knee will start working as soon as a signal crosses the threshold, whereas a “soft” knee will smoothly add compression as the signal crosses the threshold.
Compressor and Limiter
Compressors work by changing the volume of an incoming signal after it surpasses the threshold, according to the ratio. A 2:1 ratio at a threshold of -20dB means a -30dB input gives a -30dB output, but a -10dB input gives a -15dB output. The signal over the threshold has been “compressed”. Compressors often have a “make-up gain” setting as well, applying a boost to the output volume to “make-up” for the amount lost in compression.
A limiter is a special kind of compressor with an ∞:1 ration. This means that a signal will never get louder than the threshold level. Limiters are often used in loudspeaker protection, or as hearing protection in in-ear-monitors for just this reason.
Some compressors also include “negative” ratios. A 1:-2 ratio (sometimes labelled as -2:1) means the -10db input from the above example will output -15dB. While something to be aware of, negative ratios have very limited uses.
Compressors are a very handy tool for smoothing out a musician who doesn’t play consistently loud, or even tightening up the sound of a group of singers. A warning with compression, however: in live sound, a poorly set compressor is one of the leading causes of feedback. The make-up gain is usually the culprit here, as the extra volume from the makeup gain can easily push a system into feedback, usually right after the compressor stops compressing.
Gate and Expander
Expanders, and their more common sibling gates, work very similarly to compressors, but instead of acting on a signal above a threshold, they work on the signal below the threshold. For example, a 2:1 ratio at a threshold of -20dB means the -10dB input stays -10db at the output, but the -30dB input will become a -40dB output. Expanders often go into fractional ratios as well; continuing with the example, a “1:2” ratio means a -30dB input will become a -25dB output.
Like a limiter is a compressor with an ∞:1 ratio, a gate is an expander with an ∞:1 ratio. Gates will often have a “gain reduction” setting, allowing you to control how much, if any, of the input signal remains when the gate is “closed”.
Gates are one of the most useful tools we have to mix with. I use them to tame kick drums that are too boomy, reduce wind or handling noise in a microphone, hide a buzz coming from an overdriven guitar, and much, much more. If a gate is set incorrectly, then problems will arise. A gate with too high a threshold and too long an attack will leave the gate closed for the opening portion of a speech — and after any time the speaker pauses for longer than the hold time. The wrong gate settings on a snare mic will not open for a drummer’s ghost notes. While helpful, the wrong gate settings will quickly ruin an otherwise good mix.
There are many variations on compressors and expanders, such a duckers, de-essers, sidechain gates and compressors, multi-band compressors, serial or parallel compression, and more, but all of these are more involved than would be relevant here.
Saturation and Distortion
This last kind effect is somewhat strange to talk about, but it bears mentioning. Most of the time we consider distortion “bad”. We want the sound coming in to a microphone to sound like itself when it comes back out of the speaker. Distortion usually gets in the way of that. But just like a guitar overdrive can sound fantastic when used correctly, the same is true for other kinds of distortion as well.
When it comes down to it, every single thing we do adds a little bit of distortion to the signal we get. Microphones have their own distortion, preamps and mixers do as well, as do amplifiers and speakers. The trick is finding the “right” kind of distortion for the application. We often make decisions like these without really thinking about it: if one microphone sounds “better” than another, it usually means we prefer one kind of distortion to another.
I’m talking about very minor amounts of distortion here of course, but sometimes a more aggressive kind of distortion is valuable. One of the late Stone Temple Pilots vocalist Scott Weiland’s signature moves was singing through a megaphone; it created the sound he was going for.