and make sound.

Transients carry some of the essential information the human ear needs to determine
what the sound is, so playing with them can create a wide variety of musical effects.

For example, lower (faster) attack times can be used to attenuate the initial transient (like the crack of
the snare or the beater snap of the kick drum), making the tail of the drum hit more prominent.

More commonly, drum compression uses increased (slower) attack times. A slow attack lets the initial
transient slip through untouched while the compressor is still getting started compressing. This will
make the transient pop out even more, making drums punchier.

This screenshot from iZotope Nectar illustrates this technique nicely. Part of the fun of using plug-ins is
to try them on sources they’re not designed for; Nectar is an all-in-one solution for massaging and
optimizing vocal tracks, but in this case we’ve isolated the Compressor module to illustrate this trick
very clearly.

Nectar Compressor module

Here, the attack time is quite long (80 ms, the most the Solid State mode of Nectar Pro’s Compressor
can provide). The orange line shows how much compression is being applied at any given moment. You
can see that the long attack is allowing the sharp transients of drum hits to slip through and be heard at
full dynamics.

Check out the “Compressor Attack” video to hear this in action:

Similarly, stretching out or tightening up the release time of a compressor affects how notes trail away.
Sometimes release times are set to match the tempo of a track, causing a rhythmic “pumping” or
“breathing” effect; other times, release times are set very long (even over one second) to smooth out
the dynamic control of busier parts. In the screenshot above, the release time of roughly 100 ms allows
the compression to ease back to zero in a very musical way before the next hit.

Parallel and Sidechain Compression

There are two other techniques worth mentioning where compression is concerned. The first is parallel
compression, which has become more common in recent years. Parallel Compression is a technique that
involves mixing a lightly compressed signal with a heavily compressed (and sometimes high-pass
filtered) version of the same signal. This allows for a smoother result, with crisp and level sound in the
high end without any loud peaks or squashed transients. If you go back and look at all the iZotope plug-
ins that include compressors, you’ll see a Mix slider in Neutron and Nectar, and a Parallel slider in
Ozone—that’s what they’re for!

Finally, there’s sidechain compression. This form of compression uses one instrument’s level over a
threshold to activate the gain reduction (compression) on another instrument. For example, inserting a
compressor on a bass track that reacts to the kick drum will compress the bass every time the kick drum
is hit. This method will ‘tame’ the bass track and ‘duck’ it out of the way each time the kick is hit, while
still maintaining its overall level in the mix.

Another great example of sidechaining is an effect called a de-esser. In a de-esser, a vocal is run through
a compressor, and the sidechain input controlling it is the exact same vocal — after being run through
an EQ to isolate the hissy, essy parts of vocal sounds that we call sibilance. This way, sibilance causes the
vocal to compress a bit and makes the sibilance itself less audible. The sibilance teaches the compressor
how to remove itself.

Nectar has a dedicated De-esser module in its signal chain, that looks like this:

Nectar De-esser module

The vertical line sets the lowest frequency that’s fed into the sidechain, and the Listen meter shows how
much audio in this range is being gain-reduced in real time. Note how this module doesn’t have many of
the controls we’ve discussed, as its sidechain routing is handled internally behind the scenes in order to
make the process easier — but at its heart, it’s still a compressor.
Limiters

What do limiters do?

Just as a compressor “compresses” the dynamic range, a limiter limits it. The limiter serves as a ceiling
which signal cannot pass. If the signal hits this ceiling, it will be harshly compressed so that it does not
pass above.

You may be wondering if a limiter attenuates the loudest parts of a signal, how is it any different from a
compressor? Essentially, a limiter is just a compressor with a very high ratio.

As a compressor’s ratio increases, so will the amount of compression. Eventually, that compression
amounts to an impermeable ceiling.

For example, let’s say that we have a compressor with a ratio of 2:1 (not very high). We send three
signals through it, at levels of 2 dB, 4 dB, and 8 dB over the threshold.

With this ratio, the compressor would output signals at levels of 1 dB, 2 dB, and 4 dB over the threshold.
Closer to each other in level, but still not so consistent.

However, if we turned the ratio up to 8:1 (quite high), the compressor would output signals at levels of
0.25 dB, 0.5 dB, and 1 dB over the threshold. These signals are now much closer to each other and much
closer to the threshold level itself.

Eventually, as the ratio increases, the signal will not be allowed to cross the threshold, which becomes a
sort of “ceiling.”

The exact number you’ll hear changes from source to source, but any compression with a ratio of
around 12:1 or higher could be considered limiting.

iZotope Ozone Vintage Limiter

Limiter parameters
Every limiter will have at least one parameter: gain. This is used to boost signal until it hits the ceiling
and is compressed.

Some limiters will have an adjustable threshold level, which is also often referred to as the ceiling.

If your limiter does not have this capability, you can always compensate for the added gain with a
dedicated gain plug-in or at the channel fader. However, as limiters are mostly used in mastering as a
means to bring the signal to unity gain, you’ll rarely need this.

Most limiters will have a release time parameter as well. This functions like a compressor’s release time,
determining how long the limiter will take to return to zero compression.

Not all limiters will have an adjustable attack time, however. Some, like the Vintage Limiter in Ozone,
have connected attack and release parameters (set with the “Character” parameter).

Now that we know a limiter is essentially a compressor with a high ratio, take a look at our compression
output level equation again:

Output level equation

As the ratio increases, that fraction will approach 0. Therefore, the equation will eventually become this:

As expected, as the ratio increases, the output level for a signal that crosses the threshold will become
closer and closer to the threshold itself. The signal cannot pass it.

Why use a limiter?

The main use, and really only use, of a limiter, is in mastering. The compression that they offer is so
extreme that they’re rarely used on the channel level. Instead, limiters are often used on the master to
bring the track up to a commercial level, and through compression commercial “loudness.” This final
stage of compression can glue the elements of the track together and make the track louder at the same
level.

Remembering that our ears naturally prefer louder music, limiters provide mastering engineers a big
advantage in making a track sound professional. Just be sure not to overdo limiter settings, as the added
compression and eventual distortion can suck the life out of a dynamic mix.
Another use for limiters is in a live sound setting, as a fail-safe precaution. If a loud sound occurs (one
that would blow everybody’s ears out), this limiter will make sure to control it. Again, these limiters are
usually placed on the master channel.

Be sure to check out our “Introduction to Limiters” article for a bunch more information on uses and
parameters for different types of limiters.

Expanders

What do expanders do?

Again, like a compressor “compresses” and a limiter “limits” the dynamic range, an expander expands it.
Louder and quieter parts become relatively louder and quieter respectively. As such, it’s essentially the
opposite of a compressor.

“Upward expanders” amplify the level of signal that passes the threshold, rather than attenuate it like a
“downward compressor.” A “downward expander” attenuates signal that drops below the threshold,
rather than amplify it like an “upward compressor.”

Be sure to check this article out for more information on upward and downward expanders in mixing.

Most expanders are upward expanders (like the expansion featured in Ozone when setting the
compressor/limiter ratio to a negative number), but you’ll find plenty of downward expanders too.
Downward expanders act similarly to gates, which we’ll get to in a second.

Nectar Gate module used as an expander

Note: The expander above is actually the Gate module found in Nectar Pro, which has an adjustable ratio
parameter. We’ll see in the Gate section that a gate is essentially a downward expander with a high
ratio.

Expander parameters

The parameters found in an expander are and function mostly the same as those in a compressor.
The threshold once again determines the input level at which the expander will activate. This happens
when the signal is loud enough to cross this threshold level (upward expansion) or quiet enough to fall
below it (downward expansion).

Ratio, however, acts a bit differently. In a standard expander (which is upward), an expansion ratio of
1:x amplifies the signal to a level of x dB above the threshold for every 1 dB it crosses.

Again, let’s look at an example. Say we had an upward expander with a ratio of 1:3 and a threshold set
at 0 dB. If the incoming signal were at 1 dB (1 dB above the threshold), the signal would be amplified to
3 dB at the output. If the signal were at 2 dB (2 dB above the threshold), the signal would be amplified to
6 dB at the output. The louder parts of the signal are now louder.

In a downward expander, a ratio of x:1 attenuates signal to a level of x dB below the threshold for every
1 dB it drops below the threshold.

Say we had a downward expander with a ratio of 1:3 and a threshold set at 0 dB. If the incoming signal
were at -1 dB (1 dB below the threshold), the signal would be attenuated to -3 dB at the output. If the
signal were at -2 dB (2 dB below the threshold), the signal would be attenuated to -6 dB at the output.
The quieter parts of the signal are now quieter.

With the threshold, ratio, and input level, we can determine the output level using this equation (this
works for downward and upward expansion):

Knee, attack time, and release time for expanders would all work the same as in compressors.

Makeup gain is only really necessary for upward expansion. As louder parts become louder, the signal
will be louder after the expander than before, which can eventually lead to distortion of your gain-
staging is off. The makeup gain can be used to attenuate the signal, returning the louder parts to their
previous level.

Downward expansion does not require makeup gain, as the quiet parts will simply be quieter.

Why use an expander?

An expander can be used to achieve the opposite result of a compressor, expanding the dynamic range
rather than compressing it. Therefore, expanders are best used when you want to have a wider dynamic
range.

Expanders can be used to make instrumental or vocal performances a bit more varied in volume. This
can be very useful if you want a more organic sound. This can, however, reduce presence in the mix. It
can also potentially cause unnatural pumping, as these expansions in dynamic range are caused by
mathematical processes ignorant to musical phrasing.
One of the main uses of expanders is in mixing a recorded drum kit. Each drum is individually mic'd,
allowing each to have a separate channel on the mixer. However, total isolation is difficult, and bits of
the other drums are bound to bleed through into other microphones.

An expander can be used, for example, to decrease the volume of the hat in the snare mic. As the hat
will be further away from the mic than the snare, it will be quieter than the snare when picked up by the
snare mic. Therefore, downward expansion can be used to attenuate it.

With the same logic, you can use expanders to remove reverb from drums. The reverb signal will be
lower than the threshold, causing it to be attenuated in between the drum hits. Listen to the example
below:

Gates

What do gates do?

Our last dynamics processor is a gate, which is essentially the extreme version of a downward expander.
Gates provide a floor level which signal must cross to get through the gate. If the signal is too quiet to
reach this floor, it will be attenuated to silence.

Let’s see how a gate simply acts as an expander with a high ratio.

For example, let’s say that we have an expander with a ratio of 1:2 (not very high). We send three
signals through it, at levels of 2 dB, 4 dB, and 8 dB below the threshold.

With this ratio, the expander would output signals at levels of 4 dB, 8 dB, and 16 dB below the
threshold. The signals’ levels are further apart but are all still relatively close.

However, if we turned the ratio up to 1:4 (very high), the expander would output signals at levels of 8
dB, 16 dB, and 32 dB below the threshold. These signals are now much further apart and much closer to
being inaudible.

Eventually, as the ratio increases, any signal will be greatly attenuated, and all signal that passes through
the gate will have to cross this floor level.

iZotope Nectar Gate module

Gate parameters
Every gate will have at least three parameters: threshold, attack time, and release time. These all
function the same as in compressors and expanders.

Some gates will also have a hold parameter, causing the gate to remain open for a period of time (in ms)
after the signal has dropped below the threshold and before the release phase begins.

Some gates will also offer the ability to have the gate close at a different level than the threshold, which
is only used for opening the gate. This parameter is often called the close or return level.

And just like the fact that some limiters have an adjustable ceiling, some gates will have an
adjustable floor level. This is the level that signal will remain at while the gate is closed, and can be
increased up from -∞ dB.

Now that we know a gate is essentially an expander with a high ratio, take a look at our expansion
output level equation again:

As the ratio increases, that total fraction will become larger and larger. Therefore, the equation will
eventually become this:

As expected, as the ratio increases, the output level for a signal that falls below the threshold will
become quieter and quieter. Eventually, the signal will not be able to pass if it is below the threshold—a
gate.

Why use a gate?

Gates are mainly used to cut out audio when it’s quiet and unneeded. This can be on a vocal to
eliminate breaths, or can be used as an expander to isolate louder signals in a recording (e.g. isolating
drums). Gates are equally useful in a studio recording or live sound context.

They can also be used in a similarly creative way as the expander example above. However, a gate would
cause that static sample to pump harder and therefore have a more difficult time blending with the
drums. Therefore, you may just want to use an expander for that.

Conclusion

Dynamic range is a major aspect of any sound’s identity. Additionally, the level balance between
elements over the course of a track is super important for mixing. As a result, audio dynamics processors
like compressors, limiters, expanders, and gates are invaluable tools for any producer or mix engineer.
With the four of them, you should be able to shape a sound’s dynamics in any way you’d like.

Signal Processing For The Home Studio Owner: Part 2, Gates, Delay, and Reverb

Part 1 of this series focused on Compressors, Limiters, and EQ. Part 2 explores Noise Gates, Delay, and
Reverb.

In addition to your microphones, DAW/console, and room, an essential part of any home studio set-up is
your signal processing gear. From the dynamics control of compressors and limiters to the effects
processing of reverb and delay, these tools are necessary to create a professional-sounding final product.
But for the new engineer, these effects can be fairly mysterious, and a tendency to overuse plug-ins and
outboard gear is commonplace, especially for someone just learning the nuances of the art of recording.

How can you best use your typical signal processing plug-ins to enhance and optimize your recording?
Understanding how the dynamic control processors like compressors, limiters, EQs, and gates function,
and knowing how to use multi-effects such as delays and reverbs to perfection will make you a better
producer and engineer. It’s also important to remember that signal processing tools are just that – tools.
There are no rules stating you can’t use them in different or novel ways to create new sounds. But before
doing that, it makes sense to learn about the basic parameters of each and the functions they were
invented to serve.

Noise gate

The floor tom was close mic’d, and now, listening back critically, the drum’s nearly five-second release
time blurs the tom’s definition. You can live with some of the release but you want to clearly hear the
attack of each hit on the floor tom. The best signal processor to help solve this is a noise gate. Noise
gates are part of the dynamic processing family of plug-ins. Like compressors and limiters, the noise gate
has a user-definable threshold, provides variable gain reduction, and offers attack, hold, and release
time parameters.

Gates function by setting a threshold level that determines the amount of amplitude required to open
the gate, then letting the audio pass through to the gate’s output. Any amplitude level below the
threshold value will not open the gate – so the gated track remains silent. On this particular song, there
are a few breaks that Doug leads into with the floor tom, but it rings on too long. Loop the phrase so it
plays continuously, then insert the noise gate on the floor tom track and set the threshold level to the
point at which the hit on the tom just barely opens the gate. Now adjust the attack, hold, and release
parameters to achieve the desired floor tom effect reducing the long decay.

Noise gates are very useful when you need to eliminate any unwanted incidental sounds that may have
been recorded. For instance, use one on vocals to eliminate breathing sounds between lyrical phrases,
or on a distorted lead guitar to eliminate overdrive noise between lead passages. Noise gates could even
be tried on the stereo mix bus output to really tighten the breaks in the song.

Noise gates can also create their own problems, since everything recorded on the track you are gating is
eliminated according to the gate’s envelope, including any ambient leakage. This can sometimes cause a
perceptible and distracting dropout within the song’s mix. As with all signal processing, use your own
ears to decide how much noise gating is useful in your mix.

Delay and Reverb

So far the plug-ins mentioned in this article are generally considered to be dynamic, having to do with
varying or controlling amplitude in real-time. The next type of signal processing is time-based
processing, and we’ll focus on the bread and butter effects of delay and reverb.

In our example, the bass player recorded her parts directly into the DAW interface via direct box. Her
Fender Precision Bass sounded great, and with a little EQ and compression, the track is all set. The guitar
player tracked the leads with his guitar processing pedals, but recorded the rhythm guitars direct and
dry. Now you are faced with the challenge of giving life to his rhythm guitar parts.

Let’s start with delay. A delay is a time-based processor that generates discrete wave fronts of the input
signal according to the delay time. Delay settings of 250 to 500ms will create rhythmic interest while
smaller times such as 20 to 80ms can create a sense of depth. You can also create echo effects by
increasing the amount of feedback, a parameter that returns the output of the delay circuit back into
itself.

Many delays provide rhythmic note values, such as whole, half, quarter, eighths, etc., and offer a sync
option that times the delay precisely to the tempo of the original track. The delay also has low and high-
cut filter parameters, so you can change the frequency content of the delay generation when feedback
is used. You can also modulate the delay time using the depth and rate parameters, and create variable
moving rhythmic echoes.

Here’s one practical approach, assuming there are two rhythm guitar tracks. Start by bussing one
rhythm guitar to an Aux Track and insert a medium delay. Set the delay time to 40ms and pan the Aux
Track to the right, leaving the original rhythm guitar in the left channel, creating a delayed stereo spatial
spread.

For the second rhythm guitar track, a long stereo delay provides a good option. For this plug-in, a stereo
Aux Track is required, or if inserting on a mono audio track, you can automatically convert it to a stereo
track. For the most part, the controls are the same as the first delay used, but now there are separate
left and right channel parameters on the delay itself, allowing you to create complex rhythmic and
spatial movement in the stereo field.

Finally, you turn to the last solo instrument, a melodica, a three and a half octave reed instrument
played by blowing into its mouthpiece and fingering its keys. It’s a warm sounding instrument that
requires some compression but generally sounds fine. Here the decision is to add Reverb.
Reverb (short for reverberation) is one of the oldest and most widely-used time-based effects. It can add
lush ambient room sound to any instrument. Like delays, reverbs generate multiple wave fronts, but
there are a large number of fronts and the time differential between each front is extremely short. It’s
easiest to think of these fronts as reflections of the original sound, like the way an instrument sounds
when played in a well-designed concert hall. The sound generated by the instrument moves out in all
directions. It comes directly toward the listener but it also hits the floor, walls and ceiling. The sound
reflections from these surfaces return back to the listener slightly delayed from the original sound,
depending on the size and depth of the space. Of course the reflections off the floor, walls and ceiling
also continue to bounce off of the surfaces in the space and listeners perceive all those reflections at
slightly different times, creating the perception of a spacious concert hall.

Today’s reverbs emulate a wide variety of acoustical spaces. Some of the most common environments
include Hall, Room, Church, Club, and Stage. Some reverb plug-ins offer additional emulations taken
from the analog reverb days such as Plate, Spring, and Chamber. In all cases there are a few common
parameters that can be selected and adjusted to good effect.

Reverb type refers to the room being emulated (hall, room, arena, church). Reverb size refers to how
large of a space you can create. You might have a large room, a small church, or a medium hall. Diffusion
is a parameter that determines how far apart each reflection spreads out from the instrument, giving a
sense of depth of the enclosure. Reverb Decay adjusts how fast the reflections die out after the initial
attack of the sound. Pre-delay is an important parameter that determines the time differential between
the direct sound and the point at which listeners perceive the reverb reflections. Finally, most reverbs
have low and high cut filters that can reduce or increase harmonic partials as a part of the reverb’s
reflections. These filters are very useful to create transparency within the reverb process.

It is important to remember, the best sounding reverb is the one that enhances the sound without being
too noticeable. For the melodica solo, the large hall setting, a pre-delay of 40ms, a wide diffusion and
cutting high frequencies at 8kHz, results in a dreamy-sounding solo for this tune. Everything is now set
to begin the final mix, with signal processing tools helping to address the issues that would have made
this EP project sound less polished.

Final Thoughts
When using plug-in processing it is critical to keep in mind the style of project on which you are working,
the type of instruments you will be recording, how they will be recorded, and what kind of plug-in
processing will help when it comes time to mix. As you get more familiar with how signal processors
work, listen to some of your favorite recordings and try to reverse engineer what types of processors
and settings may have been used.

Addendum: The Lowdown on Impulse Reverbs

The various reverbs known as impulse reverbs offer a different approach to time domain processing.
These plug-in effects processors provide reverberation, but their processing method is very different
from the traditional reverb plug-in. Impulse reverbs, although they have many of the same parameters
mentioned above, access a complete library of sampled sonic spaces known as impulses, taken from
various rooms known for their unique acoustical characteristics found all over the world.

Famous concert halls, cathedrals, and classic recording studio tracking rooms are just some of the
options available when shopping for impulse reverbs. Instead of algorithms that emulate or calculate
the dimensions of a hall, church, room, the impulse reverb actually loads the acoustic signature of a
given space with all the actual time variables included. This results in a totally convincing audio
reverberant spatial environment.

Addendum: Using Auxiliary Tracks to Preserve Your Computer’s CPU Power

It should be noted that reverb and delay plug-in processing demands more CPU power and larger
amounts of RAM in order to accommodate the time differential between the input of the unprocessed
signal and the output of the plug-in processor. Because of this time differential, something known as
latency (the time difference between the audio signal written to the drive, passing through the CPU,
getting processed, and then returning to the audio output) can result in slight phase anomalies. Most
professional audio recording applications compensate for latency by imperceptibly delaying the
returning audio to the track output. The user is never aware of the delay and for the most part operates
the application as usual. However, it is important to keep in mind the total number of time-based
processors you insert on any given audio track.

The more time-based plug-ins inserted on a track, the greater the amount of compensation required.
For this reason, when using time-based processing, you may wish to create an Auxiliary track (Aux Track)
as you are setting up your mix and insert the time-based processors on the Aux Track and then bus the
audio via the individual channel sends to the Aux Track. You can set up one for reverb, another for delay
and so forth. In this way, the original audio track with the recorded instrument information is separate
from all the time-based plug-in processors, minimizing the need for latency compensation of the audio
track. Doing so will result in a sharper and more defined audio image throughout your mix.