In Part 1 of this article, we talked about the basics of reverb and delay — two important effects that add spaciousness to a sound. Now let’s describe how exactly they can be added to your mix and give you some tips on how best to use them in live sound.

Return to Sender

Adding reverb or delay to a sound (or group of sounds) is almost always accomplished with something called a send, which takes some of the audio from a channel (or channels) and routes it either to a built-in effect or to an output on the mixer, where it can be patched to an external effect. These sends are usually called aux sends (short for “auxiliary”), although they are also sometimes called Effect [or FX] sends). The original unprocessed sound is called the “dry” signal, while the added reverb or delay effect is called the “wet” signal.

For example, the effect send on a Yamaha MG06X mixer consists of a simple on/off pushbutton that sends the signal to the effect. The aux sends on more advanced Yamaha models like the MG16XU use rotary knobs, as shown below; in that particular mixer, you turn up the AUX4/FX knob in order to send the signal to the effect. The use of rotary knobs allows you to apply the effect (in differing amounts) to more than one channel. (To learn more about aux sends, check out this blog posting.)

Once the signal is sent to the effect processor, digital technology does its thing, creating a reverb or delay effect. But now we have to bring that effect back into our mix, almost like it’s a new signal — and in fact it really is a new signal. That’s where an effect return comes in. An effect return is like a mini input dedicated to an effect.

Effects returns normally have level controls and various routing options, such as panpots. The effect return on the Yamaha MG06X has both a level control and a switch. Bring up the level control to return the effect to your mix. Use the switch to choose between the different effects.

What Program Should I Use?

Generally speaking, you should use short reverbs and delays on fast songs, and longer reverbs and delays on slower songs. That’s because, when a band is playing at a fast tempo, a long reverb or delay will make the mix “muddy” and reduce the clarity of an instrument by stepping on the original sound.

Here are some suggestions for using reverb and delay on specific instruments:

Drums: Reverb almost always works better than delay for drums. Start by adding a small hall to the snare drum and see if you like the way it puts the drum into an imaginary space — but don’t add too much or the snare will lose impact. You can do the same for toms. Avoid using reverb on cymbals or high-hat because it will make them too “splashy.” Reverb on a kick drum is generally a bad idea because it makes the bottom end of your mix muddy and you’ll lose clarity on both the kick drum and the bass guitar.

Percussion instruments like conga and bongos can benefit from a bit of short room reverb to dramatizes the “slap.” Long reverb on a tambourine works for ballads where the tambourine is playing on the two and four, but avoid it if the tambourine is playing eighth or sixteenth notes, or is being played on an up-tempo song.

Vocals: You can use reverb or delay on a lead vocal — or both! For ballads, a hall program on the lead vocalist generally works well and adds the sense of depth you get from being in a large room. If you want something more intimate, try a small room program. A plate reverb on vocal can emphasize the highs of the vocal sound but be careful about adding too much plate — especially on sibilant vocalists — because it can emphasize “s” and “t” sounds.

If can view the actual delay time, start at around 250 ms, with enough feedback to get two or three repeats. At faster song tempos, try shorter delay times; at slower tempos, try longer delay times. This prevents the delay from interfering with the original vocal and making the song’s lyrics difficult to understand. You can also briefly switch on a long, rolling delay to emphasize a word at the end of a lyric — but be sure to turn it off so that the delay doesn’t cover up the vocal.

Background vocals can be fattened up with a doubling effect (i.e., a very short delay, around 30 ms) but keep it subtle so that it doesn’t make the vocals sound sloppy. Ditto for doubling on a lead vocal.

Guitar: Delays work great for giving guitar solos a sense of continuity but ask if the guitar player is already adding their own effects because you don’t want to overdo it. On rhythm guitar, really long delays with a lot of repeats will result in a sound similar to The Edge from U2.

A bit of reverb on a rhythm guitar will help it “sit” in your mix and blend well with other instruments. Try a room type reverb with a short decay or an ambient program. This is especially useful when you are mixing outdoors because there are very few natural reflections in an open space and so guitars can sound very “dry.”

Keyboards: A small hall or medium-size room reverb works great on piano to add depth and size. A bit of short room reverb on an organ sound works nicely to add a sense of width and space. Delay works well for synth solos … especially if you’re in a prog-rock band!

Horns: A small room or an ambient program makes horn stabs more dramatic. Longer rooms or halls work well for sax and trumpet solos. Delays can be useful for horn solos but tend to work better when the tempo is slow and instrumentation is sparse. If you’re working with a ska band, you can get away with a short delay on the horns

Bass: For the most part, stay away from using reverb or delay on bass. It reduces definition and adds clutter to the low end. An exception might be if you had a fretless bass taking a solo in a ballad — in which case adding a little hall reverb can elongate the note endings in a very flattering way.

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Two of the more fun audio tools are reverb and delay — effects that are used to add a sense of space to your mix. Let’s look at each, and then see how you can best use them in live sound.

The Basics

Reverb can be described as the sound left over in a room after the source stops. What do we mean by that? Let’s say you walk into a quiet, empty church and clap your hands. The handclap itself only lasts a fraction of a second, but then there’s an explosion of sound created by it being reflected off the walls, ceiling and floor. Those reflections create a reverberant sound, which we call reverb for short.

The amount of reverb produced depends on how big the room is and how reflective the surfaces are. What’s inside the room changes the character of the reverb, so an empty room will have a longer reverb time than a room that’s filled with people — because people and their clothing absorb sound. Reverb is very “dense,” meaning that there are so many reflections happening closely together that you don’t hear them separately. They’re kind of mushed together.

This is different from delay, which is sometimes called echo. Delay is also sound being reflected from a surface, but with delay you are able to hear a distinct repeat of the sound. For example, maybe you are watching a baseball game. You hear the crack of the bat, and then a moment later you hear it a second time. That’s because the sound hit the other side of the stadium and bounced back to you, creating a sort of duplicate. The repeat — the delay — is usually lower in volume than the original, and probably not as bright sounding.

Delay and reverb are really useful in both live mixing and in recording. They add a sense of space or distance to a sound, or can be used creatively to emphasize a lyric or the end of a solo by repeating it or making it last longer.

In the old days, reverb was created using live chambers, vibrating metal plates or springs, each of which  produce distinctly different sounds. Delay was primarily created using analog tape machines, which sound great but are expensive and difficult to maintain. Nowadays we use digital processing to create reverb and delay effects such as those built into the Yamaha MG X/XU Series mixers.

Reverb Types

As an example, the entry-level Yamaha MG06X mixer provides several different reverb “programs” (types) such as Hall, Room, and Plate.

– Hall: simulates the environment of a concert hall. Hall reverbs tend to sound big, and last longer.

– Room: simulates the sound of a large room like an auditorium. Room reverbs are shorter than halls and sound smaller.

– Plate: simulates the sound of a vibrating metal plate. Plate reverb was very popular in the 1970s. It tends to sound thin and emphasizes the high frequencies of a sound.

The intermediate-level Yamaha MG20XU adds a few more reverb programs such as:

– Ambient: creates the sense of a small space around a sound. Ambient programs are much shorter and “tighter” than a hall, room or plate reverb.

– Early Reflection: This effect is not quite reverb since it includes only the shortest reflection paths. Imagine you are in a tiled bathroom and clap your hands. You wouldn’t hear a long washy reverb; instead, you’d hear a very short, tight series of reflections from the tile. This a good example of the early reflection effect.

– Gated reverb: This is an effect that was really popular in the 1980s, especially for snare drum. Gated reverb is non-linear, meaning it does not behave the way you’d expect. Normally, bigger rooms produce longer reverbs. With a gated reverb, the room sounds big but the end or “tail” of the reverb is cut off by a noise gate instead of fading smoothly (see the Tools of the Trade blog posting How To Use a Noise Gate.)

Delays

Delay effects are usually defined by two things: delay time and number of repeats.

Delay time is the amount of time before you hear the first repeat or echo. The number of repeats is exactly that. Longer delay times give the illusion of distance. If you shouted “hey!” at the wall of a handball court, the sound would reflect back at you a fraction of a second later. If you walked farther away from that wall and tried it again, it would take longer for your voice to reach the wall and bounce back to you. Delay time is usually expressed in milliseconds, abbreviated “ms.” One millisecond is 1/1000^th of a second.

The delay time can usually be dialed in as a specific number, and it interacts with another important control called feedback. More feedback produces more repeats, but beware that adding excessive feedback to a delay can produce some unpredictable results like out-of-control looping echoes that get uncontrollably louder and louder.

Many different effects can be created simply by tweaking these two controls. For example, short delays (around 20 to 50 ms) produce a doubling effect. “Slap” delay like you hear on recordings from the 1950s is created using a slightly longer delay time of around 50 to 125 ms. Long, rolling delays have a delay time of 300 ms and up, often with a good deal of feedback.

As an example, the Yamaha MG06X provides three delay programs:

– Short: this produces a doubling effect

– Long: longer delay times than the Short delay.

– VO Echo: intended for use on vocals.

Reverb and Delay Parameters

Reverb and delay programs usually provide a number of adjustable parameters. This is a simply a fancy word for some aspect of the effect that you can tweak. For example, when you choose REV HALL 1 on the Yamaha MG10XU, the Parameter knob allows you to change the reverb time to suit the music you are mixing. If you choose the DELAY program, then the Parameter knob will allow you to adjust the delay time.

More advanced mixers like the digital Yamaha TF1 show the delay time in ms and give you separate control over delay time for the left and right channels. This allows you to do things like create “ping-pong” delays (delays that bounce between the left and right channels). The TF1 also allows you to create a rhythmic delay by entering song tempo in Beats Per Minute (BPM), or through “Tap Tempo,” a feature that lets you tap a button in time to the music, with the onboard computer automatically calculating the delay time based on your tapping! If your mixer doesn’t have those features, don’t be discouraged — you can always set delay or reverb time by ear.

In Part 2, we’ll talk about the way reverb and delay is added to a mix, and describe some creative applications, including instrument-specific tips.

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One of the most important tools in live sound is called equalization. It may sound a bit tech-y but equalization (“EQ” for short) is actually just a fancy word for “tone control.” EQ allows you to change the tonal quality of a sound, making it, for example, “brighter” or “more bassy.” We’ve all had the experience of adjusting the tone controls on a car stereo. That’s a simple form of EQ. Mixing consoles typically offer more advanced versions.

In this article, we’ll take a closer look at the various kinds of EQ controls you’re likely to encounter when doing live sound — as well as the best ways to use them.

Basic EQ

Let’s start by taking a look at the basic channel equalization offered by the entry-level Yamaha MG06X analog mixer:

This is called a “two-band” EQ because it gives you control over two frequency areas, labeled High and Low. Note the triangle at the 12 o’clock, notched position. That’s the “0” point, where the control has no effect. To the left and right you can see “-15” and “+15.” This indicates how many decibels (dB) you are boosting or cutting the selected frequency. (A decibel is a unit used to measure sound level.) If you turn the knob clockwise you’ll be “boosting” (raising) a frequency; if you turn it counterclockwise you’ll be “cutting” (reducing) a frequency.

At this point you may be asking yourself the question: What frequency? (Or perhaps even “What is a frequency?”) Well, you may have noticed that the Low EQ is also labeled “100 Hz.” (Hz — short for “Hertz”— is a measure of the number of wavecycles per second, which defines the frequency, or tonal area.) The lower the number of Hz, the lower the frequency, and 100 Hz happens to be pretty low, which is why you can think of this control as a broad “bass” control, same as the one in your car stereo. When you turn the Low knob on this EQ you will raise or lower the sound level at 100 Hz (and some of the neighboring frequencies too), while leaving other portions of the signal untouched.

Similarly, turning the High knob raises or lowers the sound level at 10 kHz. The “k” means times a thousand, so 10 kHz is the same as 10,000 Hz, which is pretty high, making this control roughly the same as a treble control on a car stereo.

When you see simple equalization controls like these labeled High or Low (sometimes called “HF” and “LF” for High Frequency and Low Frequency, respectively), they are almost always a type called a shelf EQ. The name comes from the shape of the EQ curve. The illustration below shows what happens to a sound when you raise the level of a 10 kHz shelf EQ:

Notice how the curve rises up and then levels off, gently boosting all the frequencies above a certain point (in this case, starting at around 2 kHz, or 2,000 Hz below the specified 10 kHz frequency). Affecting a broad range of frequencies above or below the specified frequency is what makes a shelf EQ easy on the ears.

In contrast, this is what happens when you use a 10 kHz shelf EQ to cut instead of boost:

Shelf EQ is useful when you want a broad stroke of EQ — for example, if you want to make a sound brighter. Basic shelf EQs allow you to boost or cut, but you can’t change the frequency. More advanced mixers like the digital Yamaha TF Series allow you to set the frequency area you want to affect, allowing precise control over the sound.

Some mixers offer three-band equalization that adds a Mid (midrange) frequency area. Here are the EQ controls provided by the intermediate-level Yamaha MG16 and MG16XU:

Notice that there are two controls for the Mid: one for the amount of boost or cut, and another that lets you choose the frequency area to be affected (known as the “peak” or “center” frequency). This type of EQ is called a peak/dip or peaking EQ. Peak EQ concentrates on a smaller, more specific frequency area than shelf EQ. The ability to “sweep” the frequency control is why it is also sometimes called a sweep EQ.

Here’s the effect of boosting a peak/dip EQ with a center frequency of 500 Hz:

As you can see, this affects a range of frequencies from approximately 220 Hz up to about 1.1 kHz. The curve drops to 0 on both sides, so it has no effect on the very low or very high frequencies.

Peak/dip EQs are useful for making a slight correction to a sound, or to prevent one sound from stepping on another. For example, you might find that a piano is fighting for space in your mix with a guitar. A dip of a few dB on the piano EQ in the region around 300 Hz can help reduce this problem. Or if you find that a vocal is not cutting through a mix, adding a few dB around 2 or 3 kHz can help … but be careful because adding too much EQ in this region can make the vocal sound harsh.

It’s important to understand that when you boost with equalization, you are not adding frequencies to a sound. Yes, you’re amplifying the signal, but you’re amplifying only a certain range of that signal. For example, boosting at 100 Hz does not add 100 Hz to the signal. It amplifies 100 Hz and some of the neighboring frequencies, while leaving other portions of the signal untouched.

Advanced EQ

Most digital mixing consoles (and some advanced analog ones) offer a more precise form of equalization called parametric EQ. This is similar to peak/dip equalization but adds another control called bandwidth (sometimes known as “Q”) that determines how wide or narrow a range of frequencies are going to be affected.

The next illustration shows a peak boost similar to the one above, but here the bandwidth has been narrowed, allowing you to zero in on a smaller range of frequencies. In live sound, parametric EQ is especially useful for reducing feedback without changing the overall sound — just cut a small area where the monitor or PA system is feeding back.

Many consoles also provide filters. A filter is a simple type of equalizer that removes a particular range of frequencies. High-pass filters (HPF for short) remove all frequencies below a certain point, while low-pass filters (LPF for short) do the opposite, removing all frequencies above a certain point. (LPFs are much less common in live sound consoles than HPFs.) Sometimes a high-pass filter will be fixed, meaning that you can turn it on or off but you can’t change the frequency. More advanced mixers give you the ability to set the frequency where the filter starts working.

And here’s the effect of a high-pass filter set to 100 Hz:

As you can see, frequencies below 100 Hz are being gently rolled off (notice that this actually begins a little bit higher than 100 Hz). You’ll find high-pass filters on most Yamaha mixers because they are very useful for removing unwanted low frequencies from vocal microphones, and that helps increase intelligibility. They also help control excessive low frequencies on guitar amps, and can eliminate the kind of low-frequency rumble that travels through mic stands.

Suggested EQ Settings

Finally, here are some starting points for EQ’ing common instruments:

• Lead vocal: HPF on; Low: 0 dB; Mid +2 to +4 dB at 200 Hz to add warmth and body; High: 0 to +2 dB to add presence, or cut -2 to -4 dB to reduce feedback
• Piano: HPF on; Low: 0; Mid: -4 to -6 dB around 200 Hz to reduce boxiness. High +2 to +4 dB to add presence and sparkle in the mix
• Acoustic guitar: HPF on; Low: +3 to + 4 dB to add thickness or -3 to -4 dB to reduce boom; Mid: -2 to -4 dB at 1.8 kHz to reduce ‘nasal’ quality, or +2 to +5 dB at 5.5 kHz to emphasize pick on strings; High: +3 dB to add presence
• Electric guitar: HPF on; Low: 0 dB; Mid: -2 to -4 dB between 1 to 2 kHz to reduce harshness; High: 0 dB
• Bass: HPF off; Low +2 to +4 dB; Mid: +3 dB at 2.5 kHz for more attack, or -3 to -5 dB at 300 Hz to reduce interference with kick drum, High: 0 dB
• Bass drum (“kick” drum): HPF off; Low: +3 to +5 dB; Mid: -4 to -6 dB at 250 Hz; High: 0 to +2 dB
• Snare drum: HPF on; Low: 0 dB: Mid: -3 to -6 dB at 280 Hz if it sounds wooley or +2 to +4 at 200 if it sounds thin; High: +3 to +5 dB at 4 to 5 kHz if it gets lost in the mix
• Toms: HPF off: Low: +2 to +6 dB; Mid: -4 to -8 at 250 to 350 Hz (reduces ‘cardboard’ sound); High: +2 to +4 dB to add attack
• Overheads: HPF on; Low: -6 to -10 dB; Mid: 0 dB; High: +2 to +4 for a bit of sparkle
• High hat: HPF on; Low: -6 to -10 dB; Mid: +2 to +4 dB at 4 to 5 kHz to add sizzle; High: 0 dB

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A big challenge for anyone doing live sound is something called gain staging. This refers to the way signal levels are set between the various components in an audio system. Poor gain staging results in high levels of noise and distortion, as well as a decrease in headroom. But when gain staging is set properly, you can get every last decibel out of your PA system, with minimal noise and maximum clarity.

The first step in achieving proper gain staging is plugging your gear into the correct jacks! This isn’t nearly as silly as it sounds. Mixers provide a variety of inputs to accommodate microphone and line-level signals. (Sometimes there are direct inputs for instruments, too.) Line-level signals (which typically come from smartphones, tablets, computers, CD players and outboard effects, etc.) are much stronger than microphone signals, so mic inputs incorporate a preamp that raises that weak signal up to line level, enabling it to pass through a mixer to a power amp or powered speakers. (Instrument level falls between mic and line level.)

Mic inputs typically use XLR connectors, but line inputs are usually quarter-inch TS (or TRS) jacks. (“TS” is short for Tip-Sleeve, while “TRS” is short for “Tip-Ring-Sleeve.”) Since instrument cables also use quarter-inch TS connectors, it’s tempting to plug a guitar or bass into a line input but this results in poor gain staging — a line input is generally not sensitive enough for an instrument, so you’ll end up cranking the mixer’s gain knob to the point where noise becomes an issue. Also, a line input does not have the correct impedance for guitar or bass pickups, so there’s likely to be a loss of sound quality.

Using an XLR-to-quarter-inch adapter to plug a microphone into a line input is another big no-no. You’ll have to crank the gain really high just to be able to hear the mic, adding noise. On the other hand, if you plug a keyboard into a mic input, you’re probably going to overload the input, resulting in distortion even when the gain is turned all the way down.

Once you have the source matched to the input, it’s time to set the input gain or trim control. This is like the main water valve in your home. You want to open the valve enough for sufficient flow (ensuring a good signal-to-noise ratio), yet not create too much pressure (distortion). If you mess this up, the signal is never delivered properly to the rest of the audio chain. If the gain is down too far, you can raise the fader as high as you want, but you’ll get nothing but noise. Conversely, if you have the gain way up and the fader way down, chances for distortion are much higher and you effectively lose the ability to use the fader’s position to mix.

Many compact mixing consoles don’t provide a dedicated meter for each channel simply because it’s too expensive. A typical alternative is a two-color LED with green for “signal present” and red for “overload.” In that case, adjust the gain control until the LED blinks red briefly, and then back it off by about 15 percent. Other mixers offer “ladder” meters, with several segments indicating signal strength. In those cases, a red LED will still indicate overload (it’s often labeled “Peak”), and there may also be one or more yellow LEDs to show when you’re approaching that point but not quite there. In those cases, adjust the gain until the loudest signals cause the yellow LEDs to light, making sure that the red Peak LED does not light at all.

Many mixers provide a switch on each channel called PFL (short for “Pre Fader Listen” or “Pre Fader Level”). Pressing this button temporarily allows the main output meter to show the level of that one channel before its fader. In other words, it’s measuring the water pressure at the main valve, but before the kitchen faucet. A good way to set levels is to put each channel into PFL and adjust its gain knob while watching the meter. (A bonus on Yamaha mixers is that, when you push the PFL switch on a channel input, it automatically sends just that signal to the headphone output for audition.) You’ll want to leave a bit of headroom — the safety margin between average signal level and the ceiling or distortion — to get the signal to read roughly -6 most of the time. If you set the gain on a lead vocal mic to read 0 dB when a singer is at average volume, the signal may distort when they get loud.

What if you turn the gain control down all the way and the meter is still at “0”? Look for a Pad switch on the channel and use it. A pad lowers the sensitivity of the input so that a microphone does not overload it, reducing the possibility of distortion. Some microphones have built-in pads designed for use on a loud source like a kick drum or a guitar amp. If so, use the pad on the microphone first because the microphone’s internal circuit is the initial “gain stage.” If you’ve overloaded the microphone, you’ll get distortion even when the input level reads well below 0.

A variation on PFL is known as Solo, though this is more commonly found on mixers designed for recording, as opposed to those designed for live sound. When a channel is soloed, the meter usually shows post-fader (as opposed to pre-fader) level, so the fader must be set at unity gain while you adjust the gain or you won’t get an accurate reading. This is the spot — usually marked next to the fader with a “0” or a small arrow — where the fader is putting out exactly what it is receiving, neither boosting nor cutting the signal. If gains are set correctly, most if not all of the channel faders should be at or near unity gain while you are mixing. Conversely, if all of the faders are very low or very high, that’s a sure sign that something is wrong with your gain structure.

Keep in mind that other “valves” affect the audio signal, such as the main mix fader — which should normally also be set at or around 0. Also, adding EQ or compression to a signal will change the gain staging. In the case of a compressor or limiter, you may need to use the its output control (sometimes called “Makeup Gain”) to reduce or make up the difference.

As you’re doing gain staging, start with the power amp controls set around halfway. If the volume in the room is too loud, it’s better to reduce the level controls on the power amp than to change any of the mixer settings. If you need to bring the master fader (and/or any of the channel faders) all the way up to get adequate volume in the room, either the power amp levels are set too low or your system is underpowered.

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