Introduction to Audio Console

Types of Consoles

Master Modules

Input Modules Group Modules

Functions of a Console
The mixer is the central device in any sound studio. Although you can do a lot
without it, sooner or later you are going to want to bring all of your materials
together to make a piece of music, and for that the mixer is essential.

The mixer is also the most formidable looking device in the studio. Pictures of
recording studios always show the mixer because there is nothing more impressive
than a couple of acres of knobs. The functions of a mixer are actually quite simple;
all you are doing is combining a few channels of signal into one or more
outputs. Along the way, the signals are amplified and equalized if necessary. The
mixer looks complicated only because these functions are repeated many times.

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The functions of a mixer are simple:

1. Process input signals with amplification and EQ and other parallel or serial
signal processing;
2. Combine those signals in a variety of ways.
3. Route the combined signals to an output (i.e. speaker, tape recorder, etc)

Types of Consoles
There are two basic architectures in the world of consoles. They are called split
and in-line designs. On a split console each I/O does ONE job. The signal
comes in the channel, either line or mic, then goes to the multi-track or speakers
depending on if you're recording or mixing. On an inline console, each I/O does
two jobs. You are provided with two inputs and two outputs thus two faders. So
you could use one path to go to the multi-track and the other path to listen to
that signal from the multi-track. You'd have two faders so you could set optimum
level to tape and then optimum mix level in the control room. All this on one
vertical module. This saves a lot of space.

Although most mixers have similar features and organization, there is some
variety to meet special purposes. This is generally a matter of which features
are duplicated. Thus a "stage mixer" has several auxilary outputs, a "studio
mixer" has very low noise, a "broadcast mixer" has cue functions and stereo
faders, etc. The basic layout of the mixer is described in terms of the number of
main inputs and outputs, such as 8X4. A very popular configuration is 8 or 12
inputs with four subgroups and a stereo master (8X4X2).

A mixer is really a traffic manager. A signal is connected to an input, and you

steer it to one of several possible outputs. Some mixers have several stages of
mixing, where inputs are mixed to submixes, or GROUPS, and then the groups
are further mixed to a stereo output. For instance, you might have five mics on a
drum set, and group these so one fader controls all drums. Trombone, tuba and
trumpet might go to a group for horns, etc. Groups usually have the same
assignment flexibility as individual inputs.

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There are several parallel mixers, (called busses), to allow different
combinations of inputs at once. You might send piccolo, clarinet and
bass to the right channel, clarinet and xylophone to the left channel,
and the clarinet and tuba to a reverb unit. The output of the reverb may
then be brought back and sent to the left and right channels in equal
amounts.

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Microscoping Signal Path

When you bring a microphone into a console

one important thing happens. The level of
the mic is changed from mic level to line
level. This is because the level that a
microphone puts out, either a condenser or
a dynamic mic, is too weak to record. It must
be boosted to line level by the mic pre-amp.
This pre-amp is built into each I/O and is the
first thing that the microphone sees when
entering the console. Once the signal is
boosted to line level it's speaking the
language of the console, tape machines,
outboard gear etc. Now your path is to the
tape machine, not to the speakers just yet.
So on a console, one of the ways to get to
the tape machine is to go through a fader to
a buss (another way is a direct patch and
we'll discuss that later). A buss is a common
signal line and is much like a city bus. It
takes a number of signals (people, if you
will) to a prescribed destination (tape track).
By assigning a signal to a buss you assign it
to it's numbered tape track. Note here that
you still haven't heard the signal. Your goal
at this point is to get the optimum level onto tape. Also note that the fader on the
I/O you're using to buss to tape is now called a channel fader. This name can
change with the job assigned to it. If you're monitoring the tape machine with that
fader, it becomes the monitor fader.

The Tape Return

Now after the level has been set you're ready to listen to it. The signal is routed
back to the console again and is this time brought to a different I/O into the line
input. The jump from mic to line has already been made and you can come into
the console with less boost. This I/O is strictly used for monitoring your signal
and setting optimum level to the speakers and two-track. Each I/O gets to the
speakers through the stereo buss which is another conveyance that sums all the
monitor I/Os to stereo. Stereo is our standard way to listen at this point in time

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(soon to change, keep your eyes here for an explanation in a future feature).You
can clearly now that the path is a two part process. See the drawing below:

To sum up: Signal path on a recording or overdub session is a two-fold

job. One fader (channel fader) is sent via the multi-track buss to the
tape machine. Then this signal is brought back through the console
through another fader (monitor fader) and is sent to the stereo buss so
we can monitor it on speakers in stereo. This same signal is
simultaneously sent to the two-track (DAT etc.). Next week we get into
console design specifics and explore more about signal path. Signal
path is the key to clean recordings.

When you look closely at a mixer you will usually see three kinds of
module: INPUT MODULES, GROUP MODULES, and MASTER
MODULES.

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Modules and Components of Audio Consoles

Input/ Output Modules

INPUT MODULES contain circuits that modify the signals before they
are mixed. Usually the following are available:

MICROPHONE PREAMPS boost the signal from microphones to

levels compatible with the rest of the studio. The quality of the preamps
separates the tools from the toys in the mixer world. (So-called
keyboard mixers do not have mic preamps.)

PADS or ATTENUATORS. A pad is a simple push button that cuts the

microphone signal by a fixed amount, usually 20 db. An attenuator is
another name for the same thing, and sometimes offers a choice of
attenuation. This is necessary because when you put mics close to an
instrument you often get an output strong enough to overload the mic
preamp.

MIC/LINE SWITCHES allow you to choose either a microphone or

signal from the patch bay as the input. Sometimes a third position
selects the outputs of a multitrack tape deck.
TRIM adjusts the signal coming from the mic preamp and/or the line
input to be compatible with the rest of the mixer. This should be set so
that there is no distortion when the fader is all the way up.

FILTERS and EQ adjust the frequency response of the input modules.

These may range from a simple high cut to several parametric
sections. EQ adds a lot to the cost of the mixer. There is often an EQ

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BYPASS, because even the best eqs always change the signal
slightly.

PAN KNOBS and ASSIGN BUTTONS make the actual connections

between the input modules and the group or master modules. In most
modern mixers the PAN function gives a continous sweep from one
output to another, with various combinations possible depending on
how the ASSIGN switches are set. Some mixers allow an input to be
assigned directly to the stereo master , others only allow assignment to
the groups.
FADERS conveniently control the relative levels of the various input
signals. This is where you control the balance of the mix.

EFFECTS, FOLDBACK, ECHO or AUX are all examples of auxiliary

SENDS, which mix signals from the input modules to special outputs
independently of what is happening on the main busses. There is a
trend in modern mixers to add a switch which connects a tape output
to the cue mix, to reduce patching with multitrack tape decks.
Auxilary sends may be PREFADER, meaning the signal will get
through even if the fader for that input is down, or POSTFADER, where
the fader controls the amount of signal sent.

SOLO is a button that connects the input module output to the monitor
output (see below) to the exclusion of all else. This is very useful for

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checking on the operation of a single microphone. PFL or prefade
listen, is similar to solo, except that the signal is monitored at full
volume. On some broadcast consoles, this feature is called CUE, and
may be triggered by pulling back on the fader.

Connections
Each input module has its own set of connections. There are line
and/or microphone INPUTS, and possibly these outputs:
DIRECT OUT carries the input signal as it appears at the assignment
switches, processed by the eq and fader. This allows you to use the
functions of the input module as if it were a totally independent device.
This is a common way to connect mics to each track of a multi-track
tape deck, and is useful any time you need a line level microphone
signal for electronic music tricks.

A BREAK POINT or INSERTION POINT is a direct out normalled right

back into the module. If there is no plug in the jack nothing is affected.
If a plug is inserted into this jack, the signal comes out of the module
instead of to the fader. This allows the insertion of outboard processors
such as delays or noise gates between the mic preamp and the mix
buss. These are usually before the fader. The signal must eventually

Pre/ Post

be patched back into the module-- sometimes this is handled by the

same jack with a stereo type plug.

Functions of a typical input module

Busses
The circuitry that combines signals from the input modules is usually
referred to as a buss. (A buss is literally a wire that connects to several
points-- modern mixing circuits are actually somewhat more
complicated than that.) Most mixers have several busses to provide a

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wide variety of features. The labeling of each buss is according to the
manufacturer's notion of which features sell mixers, but each buss
works the same way electrically. If two busses are provided for a
function, they are often labeled left and right. If there are more than
two, they are numbered.
Usually, each buss has a master level control somewhere on the mixer
and an output on the patch bay. There will often be direct inputs to the
busses (inputs in addition to the input module connections) to allow the
interconnection of two mixers or the return of processed signals from
effect devices. (If such an input has a level control, it is usually called
ECHO RETURN or a similar name.) Since many signals come together
on the buss, the possibility of overload is great-- to prevent this, meters
are usually provided to indicate buss levels.
BUSS INPUTS are extra connections from the patch bay directly to the
buss. They are intended to be used to connect two mixers together for
complex operations, but they are handy any time "just one more input"
is required.
Some common buss names are:
Main Busses
MAIN BUSSES are intended to be the most used busses. Each has its
own output module to give control of at least level, and often EQ.

Echo Busses
ECHO BUSSES are meant to be used with a reverberation device, in
conjunction with ECHO RETURN on the master or group modules, but
they may be used any time you need an extra output.

Effects
EFFECTS means the same as echo.

Cue
The CUE buss is usually connected before the faders (prefader) on the
input modules. This mix allows the operator to listen to channels that
are off as far as the main mix is concerned. This began as a feature on
broadcast consoles, where the DJ has to find cuts on the records
without sending the sounds over the air.

Monitor
STAGE MIX or STUDIO MONITOR is a mix intended to be sent to the
performers so they can hear what is going on. In a live performance,
the musicians often have on-stage speakers to hear each other. A

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musician playing with prerecorded tracks will wear headphones and
will need a mix of the recorded material and his own playing.

Each buss is part of a mix or output module. These may be very simple
or elaborate, depending on the presumed function of the buss.

Group Modules
Group modules may function as output modules or may be
intermediate control points in complex mixes. In large mixers, the subs
may even be used as simple input modules. (This adds flexibility- a 12
input stereo out mixer with 8 submaster modules may be used as a 20
to 2 or a 12 to 10 mixer depending on requirements.)
There should be level and pan controls on each module. These feed
the stereo mix (if there is one) or the monitor speakers. There will also
often be some kind of effects return.

Functions of a Group Module

Output Modules
The Output Modules (often known simply as "the mix", "main", or
"program") combine signals from the inputs and/or groups into a final
stereo mix. Not much is required at this point except a pair of faders,
so you often find other controls on the spare panel space. These might

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include the master levels for auxiliary sends, talkback functions, or the
control room monitor controls (see below).
If a mixer does not have a stereo master module, you can usually get
the same function by recording the monitor outputs.

Monitor Systems

MONITORING allows the operator to listen to what is going on.

Monitoring has no effect on the program going through the mixer, it is
simply the connection of selected signals to the control room speakers
or headphones. On the more complex boards, there is provision for
listening to practically any point in the system, from a single input
module (SOLO) to a special mix of the outputs with provisional reverb.
There will often also be aux or tape inputs to allow monitoring of
devices outside the mixer. Most recording studio control rooms are set
up so everything is controlled at the board.
Monitoring has no effect on the group or stereo outputs, which means
you can usually check on various signals without disturbing whatever is
being recorded.

Typical monitor systems have two sets of outputs: CONTROL ROOM

monitors connected to the speakers above the console, and STUDIO
monitors, connected to speakers in the studio proper. There will be
level controls for each.

Common Monitor Functions

The BUSS MATRIX is a group of knobs that determine how the busses
will be monitored. Usually, there is a level and pan control for each
buss, which connect that buss to the stereo monitor. Since this does

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not affect any outputs, it is possible to monitor with a stereo placement
totally different from what is being recorded.

TAPE/BUSS switches select either the internal mixer signals or the

output of a multitrack tape deck for monitoring. This is the equivalent of
the tape/source switch on tape decks or home stereos, as it allows the
quick comparison of the recorded to original sounds.
Consoles that are designed for use with 24 track (or more) decks often
have the tape monitor controls on the input modules. Then flipping a
switch connects the deck for mixdown. This is known as an IN-LINE
setup.

SOLO. As mentioned earlier, this is activated by buttons on the input

modules. When pressed, only the selected inputs are heard. A
variation of this, called PREFADE LISTEN or PFL bypasses the input
faders. There is usually a light to indicate solo is active because it is
often hard to notice a single pressed button. There should be a special
knob to set the level of solo monitoring so you don't mess up any
settings trying to check a quiet source.

METERING. VU meters may be connected to various parts of the

mixer. Meters must be available for the groups and main outputs, can
often be switched to the echo outputs, and are sometimes provided for
the inputs (although we usually have to make do with an overload
light).

TALKBACK allows the console operator to speak to the musicians in

the studio. When the button is pushed, the control room speakers are
shut off and a small microphone built into the console is activated; that
mic is connected to the studio speakers. Shutting off the control room
speakers prevents feedback through the recording mics.

SLATE is a similar function that allows the engineer to record his voice
on the tape. "Slating" is the announcement of take number and other
information on the tape. This is very useful when the tape is played
later.

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TONE provides a sine tone of calibrated amplitude on the main
busses. This simplifies the adjustment of levels on the tape recorders
(set levels so the VU meters on the deck match the board), and
provides a reference for later playback. The tone is usually available at
the patch bay as well.

MONITOR FUNCTIONS OF A MIXER

Q&A
Question:
What are your goals when you bring a microphone or line level signal
into a console?
Answer:
To get it on tape (hard drive etc.) or to the speakers for listening.

This Q&A just summed up basic console signal path. You either want
to record a signal or monitor it. It's either/or because the level that you
would send to tape would not necessarily be the optimum level for a
mix in the control room. Therefore it's a two step process to record and
monitor a signal. This gives you individual control over control room
volume and level to your headphones

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Features and Functions of Audio Consoles
EQ Section | Why Use EQ?
EQ should be your last line of defense when recording. Microphone
positioning is a better solution to frequency problems. For instance, next
time you're recording an acoustic guitar, put your head in and around
where the player is and notice how the sound changes. Up near the
neck you get more of the string sounds and it's a bit thinner. As you
travel down near the sound-hole it gets fuller and at the tail of the guitar,
high frequency falls off and you get more lows. Where your mic is
placed will determine how it sounds in the control room. Also keep in
mind that you're listening in stereo, if you put one mic up and expect it
to sound like it did when you were in the room with the player you'll be
disappointed. Even when you're going to one track of the multi-track,
put up two mics and buss them both to one track. This gives you the
benefits of stereo miking even when you can't afford to lay it down that
way. There are times however that EQ is your only way to shape your
sound and add the ingredients necessary to have it come out well.

Types of EQ
Although there are other kinds, the following three types of EQ are the
most common that you'll encounter at the console. Each number below
also represents the amount of controls you'll have to adjust.

Type 1
This type is called Tone
Control or Shelving EQ. You
get one control, boost and cut
of a fixed frequency. You'll
see this type of EQ every day
when you get in your car.
Bass and treble control on
your car radio is a basic tone
control. These are most likely
factory-set at 10k at the top
and 100hZ at the bottom.

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Type 2
This kind is called
Sweepable EQ. You get two
controls, boost and cut and a
frequency sweep control.
The sweep varies, sometime
it covers a small area but
mostly you'll find that
manufacturers cover a large
piece of tonal real-estate,
especially in lower priced
consoles. Common labeling might be 4k to 18k at the top and .5
(meaning 50hZ) up to 8k in the bottom band. By the way, when you
hear the term "bands" of EQ, that is referring to how many different
volume/sweep control combinations you get on a console. For instance
a four-band EQ would cover lows, low-mids, high-mids and high
frequency, all separately.

Type 3
This is called Parametric EQ. You get three controls, boost and cut,
frequency sweep, and "Q" control. "Q" stands for bandwidth, it refers to
how big of a piece of frequency real-estate you're affecting when you
boost or cut. These are called wide or narrow "Q" (narrow is
sometimes called high "Q"). What the "Q" determines is how many
frequencies around your target frequency you're affecting. The drawing
below illustrates what I'm talking about:

Summary
These three types are basic. Some manufacturers might give you an
either or "Q" control on a sweepable EQ. This would be called semi-
parametric because you only have narrow or wide options and not a
sweepable "Q". Next week we'll cover shelving EQ and some basic
EQing scenarios.

Shelving EQ
One option you'll often find on the upper and lower band of a console's
EQ section is the ability to shelve the EQ. Shelving refers to the ability
to boost or cut a certain frequency and then all the frequencies beyond
that, either at the top or bottom of the spectrum. (see the illustration
below) Shelving is a switchable option and not the norm for most EQs.
Most types of EQ, as a default, let you boost and cut at a wide "Q".
This uses the bell shape that you saw in the artwork of last week's
feature. Overall, it's the smoothest way to boost or cut. Instead of using
a wide "Q" you may want to create some "air" in certain mix

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 ingredients. The term air describes an overall high frequency sheen
that shelving creates. This is very useful for lead vocals and perhaps a
solo instrument that you want to give a special quality to. Overused it
makes your mix top-heavy. Used wisely it gives focus to important mix
ingredients like drum overheads, certain percussion and vocals.

If you cut the low frequencies of certain mix ingredients you eliminate
unwanted low frequency clutter in a mix. For instance, let's say you
have a home studio that's not completely isolated from outside noise
and you live near the freeway (I actually worked in such a place).
Every time you record vocals or anything using a microphone you'll get
a lot of rumble and low frequency hash on tape. This is further
accentuated by using a digital multi-track because of the low noise
floor. Because these machines are so clean it leaves that low
frequency hash more exposed. By shelving out the low end where it's
not needed, say on lead and background vocals you clean up the
problem. For instance, you wouldn't shelve 50hZ out of a kick drum or
bass track but on a vocal or sax solo it would be fine. You're not using
that part of the spectrum anyway. I assisted for one famous engineer
who as a rule would go through and shelve out the low frequency on
any track where he wasn't using it. Not a bad practice especially in
today's digital world.

Parametric vs Semi-parametric
One other distinction that I want to mention here is parametric vs semi-
parametric EQs. The "Q" control option of a fully parametric EQ allows you to
adjust the bandwidth continually. Sometimes to save money a manufacturer will
give you an either/or bandwidth control. You can either choose wide "Q" or
narrow "Q". This is done with a button or incorporated into a push/pull frequency
sweep knob. This is called semi-parametric because of the either/or nature of it
as opposed to a true sweepable "Q" you'd find in a fully parametric EQ.

Assignment Section | Getting on the Buss

We've established that the basic building block of a console is called an I/O
(input/output). The input section is where you'd plug in a microphone or line level
device into the console. How you get out of either the channel side (going to the
tape machine) or monitor side (going to the speakers/2-track) of the console is
called a buss. A buss is a common signal line that carries the signals you assign
to it. You can equate it to a city bus, it's a conveyance where you can transport
one or many people (signals) to a destination. How you assign it is by using a
buss matrix, to get to the multi-track, or by injecting individual monitor faders into
the stereo buss to get to the speakers/two-track.

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The Buss Matrix
When you see the specs for a console and it says 8-buss, they are speaking of
the routing capabilities of the buss matrix (also known as the multi-track buss).
Each buss of the matrix corresponds directly to a track on the tape machine. If
you had an 8 buss console and one ADAT, it would be perfect because you'd
have a dedicated buss to each track. If you had a 4 buss console and an ADAT
you'd only be capable of bussing to 4 of the 8 tracks at any one time.

A Bussing Scenario

A Buss
Matrix

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If you had to record a drum kit with an 8 buss console, one of the ways you might
route the signal is as follows:

You can see that every mic but the tom mics maintain a one to one relationship
with the tape machine. To save tracks I chose to record the three toms down to
two stereo tracks. This is the beauty and strong point of the bussing matrix. You
can take any number of signals and bounce them down to stereo or mono. By
manipulating the pan feature of the matrix I can assign Tom 1 completely to track
4 (panned even), Tom 2 to both tracks 4 and 5 (panned center), and Tom3 to
track 5 (panned odd). This way I free up an extra track for whatever else I'm
recording at the time, bass for instance. This panning feature is the pan on the
channel fader or sometimes a dedicated buss pan up near the matrix.

Faders
The Stereo Buss Master Fader
There are three kinds of faders that you will run into on a console.
One of these is the Stereo Buss Master fader. This is one stereo
or two mono faders that control the level of the stereo buss. Think
of the stereo buss as the collection point that all the monitor faders
feed on their way to the speakers and two-track. When you hear a
song fade out on a CD it's the Stereo Buss Master fader that the
engineer used to do the fade.
It's very important when building a mix that you make a special
effort to "fill up" the stereo buss. You would determine this by
looking at the stereo buss meters. Too little level and you'll have to
make it up at the two track or crank your speakers in the room.
This would add unnecessary noise from your power amp or two
track's input. Too much level on the stereo buss and you'll use up
the headroom of the board and cause distortion. You want to hit
the stereo buss so that on the peaks it's in the red at about +2. Not all meters are
created equal however. Analog meters are my favorite (the kind with moving
needles) They average the peaks and give you a good idea of how hard you can
hit a console or a tape machine. In general digital meters, even in VU mode, are
much more responsive to level and give you the impression that you're using
more headroom than you actually are. It's a good practice to compare any digital
meters to an analog meter just to see where you stand.

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The Monitor Fader
If you read last weeks feature, you'll see that the monitor fader is
the fader that receives the line inputs from the multi-track tape
recorder. If you've used up all your multi-track outputs and have
faders left over, you can use these extra faders to bring back more
line level signals. These might be reverb returns, or live midi
instruments that you're running virtual and synced along with your
mix.
On any console, the job you give the fader determines the name of
that fader. If you're using it for a tape return or for some other line
level input that's not going to the multi-track to record, then it's
called a monitor fader. If it's going to the multi-track to be recorded
it's called.........

The Channel Fader

The channel fader takes in a line or mic level signal and sends it to
the multi-track tape machine. This is accomplished by a direct patch
or by using the multi-track busses. The channel fader is usually set
at zero which is called unity gain. The reason you set it at zero and
leave it there is so that you can use the mic preamp's gain control to
set your level to tape. For the most part, anytime you have two
opportunities to add gain along a signal path you'll try and optimize
all but one by running the unneeded stages at unity gain.

Auxiliary Sends
Auxiliaries, aux sends, echo sends, and cue sends, these are four names for the
same thing. Simply put, an auxiliary send lets you send off a signal from the
console, either pre or post fader, to any device that will accept a line level signal
like a reverb or delay or headphone amp. All this without effecting your original
signal one bit. One of the two main things you'll use an auxiliary send for,
is..........
Parallel Effects Processors
The following devices fall under the category of parallel effects.
• Reverb
• Delay

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• Pitch Shifter
• Flanger
• Chorus
• Phase Shifter

The best way to use these

types of effects is to use them
in a parallel fashion. To see
what I mean by that, look at
the drawing below:

When you're effects are used

in this fashion, it optimizes
them in a number of ways.
Because all I/Os on the
console have aux sends, you
can send any number of
signals to the same effects
device. (for the BEST possible mix it's not advisable to send everything to one
reverb, more on that during our upcoming series on mixing). You also have your
original signal untouched and the treated signal coming back in stereo on two
additional I/Os. This lets you set a wet/dry relationship that can stay this way
during the whole mix. In addition you can maximize your signal to noise ratio.
You do this by using up the headroom of the effect so you don't have to ride the
output at a hot level. This makes the output stage of the device much quieter.
On the other hand, If you chose to hook up a reverb in a serial fashion (like most
guitar stomp boxes) you'd be at the mercy of the device as far as noise goes.
This is because your whole signal would be traveling in and out of the device, so
if the device is noisy then so is your original signal. In addition, you wouldn't be
able to use the same device for a number of signals, but only for that one signal
you're sending through your processor. The other thing you're going to use your
auxiliary send for is........

Headphone Mixes
A lot of the time, it's not practical to send your stereo buss mix to the band when
you're tracking or overdubbing. This is because you may have special requests
from the band for specific things in and out of their mix. For instance the
drummer may not want drums in his phones because they are so loud in the
room, or you may not want to listen to an annoying click track in the control room.
So you can use the auxiliary sends to build a mix from individual I/Os just as you
would with a fader. Then you'd send the aux master output to the headphone
amp and you'd have an independent mix for the band. You can then change your
mix in the control room and not effect the bands headphone mix.

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Pre Vs. Post
One of the options you have when setting up auxiliary sends is whether to send
them before or after your fader. This is called Pre or Post and is represented by a
button near the auxiliaries. It can be a button for each separate aux send or for
groups of two or more. The reasons you'll set these up in either fashion are
simple. Here's the basic guidlines: Headphones - Pre, Effects - Post. Let's take
these one at a time.

Headphone Setup
The great thing about using auxiliary sends for headphone mixes is that if you set
them up Pre-fader, you can have a completely independent mix going out to the
studio. This means that no amount of manipulation to your mix in the control
room will change the headphone mix. If the drummer wants no drums and all
bass and guitar in his phones then you can give it to him. You can use individual
mono or stereo pairs of auxes to send as many mixes to the studio as you have
headphone amps. Very versatile. If they're all set-up Pre-fader you've got total
independence. You go about building a mix from the auxes just like you'd do it
from faders. First you'd put on headphones with the feed that the studio is getting
or choose to listen to the auxes in the control room monitors. Then you go
channel by channel just as you would with faders, and add each mix ingredient to
the headphone mix, including reverb. It's easy to take requests and make tweaks
along the way and you don't have to change your own mix. This is great when
the band has to listen to a click track when playing. Because they have their own
mix you can give them the click and just have the instruments etc. up in the
control room.

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Effects Setup
On the other hand, when you're setting up auxes to send to parallel effects
processors, the smart choice is to use the auxes Post-fader. This way the effect
will follow the action of your fader. For instance, let's say you've chosen a few
reverbs and ambient effects for your drum kit. You set them up PRE-fader. Now
you decide that you want to fade the drums early and just have the percussion
take the song out in the fade. Because the effects are before your fader, when
you fade the drums you'll have drum reverb still coming out into the stereo mix.
Not something you'd want. If you'd of set them up Post-fader, the effects would
maintain the same wet/dry relationship that you had set up in your mix and the
reverb would fade with the drums.

If you keep these guidelines in mind when setting up reverbs and headphone
mixes, auxiliary sends need not be intimidating or mysterious.

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Semi Professional Audio Consoles & Mixing
Techniques

What is a Mic PreAmp?

The first thing a microphone sees when it enters the console is the mic preamp.
The reason you need a mic pre is to boost the microphone's signal from mic level
to line level. Line level is the language that the console, tape machines and all
the outboard gear in the studio speaks. Once you've boosted the mic to line level
you can send it's signal throughout the control room to interface with any line
level device.
The mic preamp is your master volume control to set level to the multi-track tape
machine. Even though your signal passes through a number of other gain stages
such as the channel fader and multi-track buss trim, you still want to use the mic
pre to set your level to tape.
If you're using a direct box to record bass or guitar, you must plug the output of
the DI (direct box) into a mic preamp. This is because the output of the DI is at
mic level. A bass or guitar, just as a microphone, can't speak the language of the
studio without being translated. A guitar's output is unbalanced and high
impedance. The input of the console wants to see a low impedance, balanced
signal. The DI translates the signal to low impedance, balanced mic level signal
so you can plug it right into the console's mic preamp. There's a rhyme you can
use to remember - the hi and lo impedance puzzle. Hi into Lo won't go, Lo into Hi
will fly. This simply means that Hi impedance into Lo impedance doesn't work but
Lo impedance into Hi impedance will.

Padding the Input

A mic pre sometimes comes with a pad. This is a button that resides near the mic
preamp. It's simply a one-shot volume reduction control. If you engage the pad, it
reduces the input volume BEFORE the mic pre by a pre-determined amount,
usually -10dB. This is used if the input from the mic is so hot that you're
overloading the mic pre and causing distortion. If you use the pad you can lower
the volume and gain back some headroom in the console.
Some condenser mics come with a pad. This is a switch on the mic that says 0 in
one position and -10 in the other. This pad is usually before the mics
amplification stage. If you're overloading the mic's internal amp then you can pad
the signal so it's clean coming out of the mic. This signal may still be too strong
for the mic preamp meaning you'll have to pad it again at the console. Kick and
snare drums are notorious for having the kinds of level that will overload
microphones and consoles. If you're hearing distortion when you're recording
instruments with high transients, pad them either at the console or the
microphone and you should hear them improve immediately.

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Different Ways to The Same Places
The features of the console series put together the nuts and bolts of the console
and how to work quickly to get the desired goal accomplished. Here are some
advanced tricks you can use to get around and through the console in other
ways.

The Versatile Channel Fader

Most inline consoles will let you use the small fader in two ways during mixdown.
Normally the small fader (channel fader) is after the mic pre-amp and before the
multi-track buss. The mic must pass through here on it's way to the multi-track.
However, during mixdown this fader can often be used as an additional monitor
fader. For instance, you could assign it line input status (most consoles will let
you toggle the channel fader from mic to line) and then route additional tape
returns or reverb returns into it. Once you patch these returns you simply assign
it to the stereo buss with all the other mix faders. How this stereo buss
assignment happens varies from console to console. Some manufacturers give
you the option of routing to the multi-track busses or the Stereo buss via a
button. This is the most flexible of options because you can configure the fader
as a monitor or channel fader easily. However all is not lost if this is not available.
Remember that this fader still routes to the multi-track buss. If you patch a few
multi-track busses into a few monitor faders, then assign the channel faders to
those busses you can take the back door to the stereo buss. (see the diagram
below).

It's important to use this subgrouping function in groups of two odd/even busses.
These will be patched to your two monitor faders and hard panned left and right.
This way you can assign your channel faders to these two multi-busses and your
panning will be reflected in the Stereo buss. For instance, patch buss 1 and 2 to
two open faders. Pan these faders hard left and right. Now patch a line input into
your channel fader and in turn assign it to buss 1/2 and pan it like you'd want to
hear it in the stereo buss. Make sure your monitor faders are not muted. You
should now be hearing your channel fader's signal in the stereo buss and the
channel pan should reflect what you're hearing.

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Subgrouping Instruments
Last week we discovered how to route the channel fader to the stereo buss using
the multi-track busses. This allows you to use the channel fader as an additional
line return for more tape tracks or reverb returns. Now let's talk about the
advantages of subgrouping groups of instruments for use during mixdown.
When I mix I like to start with things laid out in groups on the console. For
instance, I'll cross-patch the drums and percussion so they're together, then
groups of keyboards then guitars and background vocals. I'll leave a spot free
with a few faders open for the lead vocals and automated reverb sends right in
the center of the console. This way when I'm ready to make my vocal moves in
the mix, my head is right between both speakers. This is optimum for being able
to hear nuance of levels or other things that might mask your vocal.
Once I've got all my patching, levels, EQ and other things all setup, I'll subgroup
certain instruments together. This does not negate you're individual control of
level but enables you to make master group moves with backgrounds, drums,
percussion etc if you need to. For instance I'll group the drum kit to two faders (or
a stereo fader if it's available), keyboards to another and guitars to another (all in
stereo if that's my plan). Then it's easy to make master group moves with or
without automation.

How to Get There

Of course each console is different and may have grouping functions that you
can easily implement. (for instance SSL has dedicated stereo group faders that
are easily assignable). It's much the same as we did with the channel fader in
last weeks feature. (I'm taking for granted that the console we're working with
does not have a grouping function and will allow you to send the monitor fader to
the multi-busses). First designate a gang of faders that you can use as group

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masters. These should be easily reachable so when you make your moves
you're not off-axis from the speakers center image. Patch an odd even multi-track
buss out combo to two faders and label it GROUP 1 (or with the name of the
instruments you're grouping, drumkit for instance). This patch might look like
multi-track buss outs to Console line inputs. Then patch another two busses to
another two faders etc... as many as you need. Then assign these subgroup
master faders to the stereo buss. Now simply take the groups of tape returns you

wish to group, de-assign them from the stereo buss and re-assign them to your
group faders via the multi-track busses that you chose. The odd/even buss
assignment should follow your monitor pan (that depends on the console) and
should be in the same stereo perspective as your original channels. Provided of

course that your group faders are panned hard left and right (don't forget that or
you're mix will be in mono). See the diagram below to further visualize what's
going on.

Endless Auxes
Console manufacturers with flexibility in mind will let you source the channel
fader's input from the monitor fader of the same I/O. This simply means that the
fader which you are listening to in the stereo buss also goes to the channel fader
and stops there. We know by now that the channel fader can usually be assigned
to the stereo buss or the multi-track buss (on most inline consoles). Since we're
through recording we don't need the multi-track busses to go to the tape
machine, so we can use those same busses to go other places.

Sending the Multi-Track Busses Elsewhere

The beauty of a patch bay is the ability to send line level signal anywhere. In this
case we're going to patch an output of a multi-track buss to the input of a reverb
or other effect. You can now see that the channel fader is just another post-fader

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send that travels via the multi-track buss to an effect. This way you can use the
channel fader as a volume control just as you would an aux send. In one case
you patch the output of the aux send master to the input of an effect, in the case
of the channel fader you assign the output of a chosen multi-track buss to the
effect.

How to Make it Work For You

Here's the working scenario. You use up all your aux sends and need an
additional send for a particular reverb. You'd patch the output of buss 1 (this can
be any buss but for our demo it's buss 1), to the input of an effect. You then bring
back the output of the effect to two free inputs on the console and send those to
the stereo buss. Let's say for our purpose we want to use this effect on some
background vocals and guitar. You'd then assign the input of the channel faders
on the background and guitar I/Os from the monitor fader. You're channel fader is
now getting a post-fader feed from the monitor fader of the same console I/O.
You then assign the channel fader to buss 1 and it becomes a master (post-
fader) volume control that sends to your reverb. See the diagram below to further
illustrate this:

Direct Outputs
Direct outs are an alternate way to get your signal from the channel fader to the
multi-track tape machine. You already know that the multi-track buss is a way to
send individual and groups of signals to one or more tracks. If you're signal-to-
track relationship is one-to-one, using a direct output is a cleaner way to get your
signal from the console to the multi-track. You're not even limited to going to the
same number track as the I/O you're using. If the direct output is represented on
the patch bay you can assign it on the individual I/O and then patch it to any track
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you like (or plug directly from the console to the tape machine if you need to).
You just have to remember that you can't mix two signals using direct outs, it's
strictly one signal to one track.

Knowing Your Path

When you use the multi-track buss you're going through yet another amp on the
console. This ads noise and is another gain stage to manage. When you use a
direct out you eliminate another gain stage. The more gain stages you can
eliminate in your signal path the cleaner your signal will be. Let's explore for a
second just how many gain stages are in a single I/O on a console. A gain stage
is every time you have the opportunity to ad gain, and noise. If your console has
4 bands of EQ and 4 Aux sends, that's 8 chances to ad gain right there. Add the
mic preamp, channel fader, monitor fader, multi-track buss, stereo buss
(summing amp) and line trim you have a total of 14 chances to mismanage your
gain structure.

Keeping Your Signal Clean

When you're tracking and mixing, having gain stages working efficiently with
each other and other amps outside the console should be a big part of your
focus. For instance if you're barely sending signal out through an auxiliary to a
reverb and you must boost the gain of the output in order to hear the effect,
you're adding unnecessary noise. It would be better to hit the input of the reverb
harder by boosting the aux send to the effect and downplay the noisy output
stage of the effect.

Keeping it Clean
Every audio device, be it a fader, line trim, auxiliary send, or audio tape has
what's called a noise floor. This is the inherent noise of a device. It might seem
that digital recorders have no noise but in actuality they just have a very low
noise floor. So everyone's in the same noisy club. Along this noisy pathway
comes your signal. Every opportunity to add gain along this pathway is an
opportunity to add noise to your once pristine track. It's inevitable that some
noise will creep along but if you manage the gain properly this will be masked or
greatly understated by the fact that your signal is printed nice and strong.

The Box (Dynamic Window)

I like to think of recording as fitting the signal
in a box. At the bottom of the box is the
noise floor. At the top is distortion. The
closer you can get your signal to distortion
without going over, the cleaner it will be.
This is because you're away from the
dreaded noise floor.

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As we said the more opportunities you have to add gain, the more you can
degrade your signal. There is a concept at your disposal that can greatly help
you along your signal path. It's called.......

Unity Gain
Unity gain is the point at which you're neither adding or subtracting level from
your signal. This of course means that you have to have the means to add gain,
namely a volume control. This could be a fader or a volume pot. Unity gain is
usually marked as ZERO on the fader or pot. Some pots have no scale but are
detented. This is a notch you can feel when you turn the knob. This detent is
sometimes zero. However, some manufacturers choose to keep us all in the
dark. If the manual for the device doesn't give you a clue, I usually guess that
unity is about the 2 O'clock position, or number 7 on a one to ten scale.
Let's trace a bit of our signal from mic to multi-track and see how we can use
unity gain to keep our signal cleaner. The signal will flow as follows:
• Mic
• Mic Preamp
• Channel Fader
• Multi-track Buss
• Multi-track machine

There are three opportunities to add gain here. The mic pre, channel fader and
buss trim. There are amps in all three of these gain stages, all with noise floors
and distortion. Think of the box. The best possible scenario would be to eliminate
two of the three by using the unity gain concept, then using only one as a master
volume. In this case the mic pre is the best choice for the master volume.
Therefore, you'd set your channel fader and buss trim at ZERO and not add any
gain there. Then you'd use the mic pre as the volume control that adjusts your
level to tape.

Keeping it Clean - The Tape Return

Getting your signal clean to tape is only half the battle. To mix properly and get
the most out of your console you need to manage your gain all the way to your
speakers and two-track tape machine. Getting in the habit of doing this time and
time again is to your benefit.
Let's trace our path from the tape machine to the speakers and two-track. When
the signal enters the console the first thing it usually sees is the line trim. This is
another amp you encounter before the monitor fader. From the line trim you go
through the auxiliary sends and EQ stages and then to the fader. In order to hear
all the faders on the board they must spill into the summing amp. This amp sums
all the different faders to stereo, and then they're routed to the various two-tracks
and speakers in the room. To get to the summing amp the faders travel along the
Stereo Buss. A buss is a common signal line. On some consoles the monitor
faders normally go to the stereo buss. On others you must assign the fader to the
stereo buss. There is then a master control called a Stereo Buss Master Fader.
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It's usually set aside and is a different color to tell it apart from the other faders.
So here's the breakdown of signal flow from tape:
• Multi-track
• Line Trim (console input section)
• Monitor Fader
• Stereo Buss
• Stereo Buss Master Fader
• Speakers/Two-track

More About Unity Gain

We talked about eliminating one or more gain stages by setting them to unity
gain. There are three gain stages in our example above. To properly manage the
gain on the tape return path you should set your line trim and your Stereo Buss
Master to unity gain (unity gain on the stereo buss master fader is marked as
zero, see your manual for unity gain on your line trim). Then use your individual
faders to adjust volume on the console. Your goal is to "fill up" the stereo buss
and not overload it. How you gauge this is to watch your stereo buss meters as
you mix. Your goal is to use up the headroom in the stereo buss and thus stay
away from the noise floor .

Things that you should not do!

What you don't want to do for instance is over-do it on your monitor faders and
then adjust your Stereo buss master level down 5 dB or so. This is not smart gain
management. You're adding noise at the monitor fader and not utilizing your
headroom at the buss master. Keeping all these gain structures in a row will keep
your tracks sounding clean by minimizing console noise.

Clean Effects Sends

Auxiliaries are most often used for headphone mixes and parallel effects devices.
Because of the abundance of gain stages when using effects, we'll concentrate
on that side of the auxiliaries' dual personality.First let's count the number of gain
stages that your signal will encounter in route from the auxiliary and back again:

• Auxiliary Send (individual sends on each I/O)

• Auxiliary Master
• The Effect's Input Setting
• Any Internal Level Setting the Effect May Have (software)
• Output Setting
• Console Fader (return level)

There are six opportunities to add gain at these stages. We know from past
features that the best way to manage gain is to employ the concept of Unity Gain
for the stages that aren't directly being used to manipulate gain. In this case the

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individual auxiliary send and return fader will be the way we manipulate gain
directly to and from the reverb. All others will be set to an optimum setting.

The Game Plan

Before we get to the individual settings let's discuss some theory here. The
problem with effects is that they create a lot of noise in the process of effecting
the signal. This is why we use them in a parallel fashion as opposed to plugging
our signal directly through them. If we did plug through them, we'd be at the
mercy of the device as far as regulating our noise level as opposed to signal
level. This is called Signal to Noise Ratio. The greater the ratio the happier we
are as engineers. So, the basic plan is to hit the unit hard at the input, (without
overloading it of course) and then turn down the output to keep the noise at bay.

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 Just Doing It
• Aux Send
The individual send is used to set the level to the effect. The thing to watch when
you're sending the signal to the effect is the input meter of the effect. Hit it hard
but not too hard, leave yourself some headroom for other signals you may send
later in the mix.

• Aux Master and Effect Input

The aux master should be set to unity which is usually about 7 or 3/4 the way up
on your rotary pot. The effect's input setting should also be unity. This is tricky
because most manufacturers don't give you a clue. Once again I err on the side
of caution and use the "7 or 3/4 way up" rule that we mentioned a moment ago.

• Internal Level
Next, page through your editable parameters for any software based level
settings in your effect. It depends on how this sounds when you do it, but I
usually boost it a bit and see what happens. Listen to your outputs and if there's
too much noise added then back off. For instance, on Yamaha SPX 900, I go in
and crank the software level of the individual effect and it works great. Be careful
but see what you can get away with.

• Effect Output
Output settings can be a dedicated knob or not exist at all, it depends on the unit.
On the Ensoniq DP-4 there is a dedicated input and output knob. Once again
experiment. The output might be an attenuator and in that case should be wide
open. Reading the manuals or questioning the manufacturer via email or a
support line would really help here and I highly recommend it.

• Return Settings
Your return level then becomes your savior because you can ride them down in
the mix and bring the noise floor way down. This stage should not be run at unity
for that reason.

Interconnection
In a large pro studio with lots of gear, you will need to be able to connect all of it
to each other at some time or another. It's not practical to run cables all around
through the studio from here to there. So a lot of time and money is spent to build
the facility so you can run cable under the floor and through the walls. This is so
you can plug in a mic on the other side of the glass and be able to hear it and
route it to your console. All this without jumping through a bunch of hoops to get
it done. In addition, you need to be able to access the gear in the outboard rack
and the multi-track tape machines. So, for practical reasons, all the ins and outs
of all the equipment in the studio terminates at the patch bay. This way you could
make a copy of a DAT to a cassette as easily as taking two patch cables and

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connecting their inputs and outputs, even though they may be at different ends of
your rack.
It is relatively expensive but makes your work much easier and is the way that all
major studios operate. Even in a home studio, someday you may decide that you
need a patch bay....even a small one, so here's how it all works.

Patchbays
Patchbays by nature are
groups of inputs and
outputs. They are
constructed into groups of
two rows stacked on top of
each other. The top row
being an output and the
bottom row being an input
(some bays are just the
opposite). Patch cables are
expensive, and so you don't
have to make a lot of
common patches there are
connections that are
normalled.

A normal is a connection
that is hard wired at the
back of the patch bay so you don't have to patch it at the front. These are ins and
outs that you want to be connected all the time. For instance, buss outputs to
multi-track inputs, or multi-track outputs to monitor fader inputs. When they are
hard wired in this fashion they are said to be normalled. This is a flexible setup by
design, you can do something called breaking the normal by simply routing a
new signal (via a patch cable) to any input. This disconnects the normalled signal
to the input you're plugged into, and inserts the signal where you've patched
from.

Two Kinds of Normals

There are two kinds of normals, they are called Full and Half normals. This has to
do with how the patch behaves when you plug into an input or an output. For
instance, on a half normalled patch bay, if you plug a cable in any input, you'll
break the normal and be able to route your signal there. But, on a full normalled
patch bay, if you plug into any input or output you break the normal.

Personally, I like to work with half normalled patch bays. The reason being is that
there are some situations where you want to have your signal going to two

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places. To be able to make that double connection happen with one patch is a
good thing.

What's a Mult?
If you're in a situation where you're working on a patch bay that's fully normalled,
and you must split your signal to two places, then you've got a problem to solve.
If you remember last feature, when you plug a cable in any output on a fully
normalled patch bay, you lose the normalled signal. The way to overcome this is
to use a handy set of patch points called a mult. A mult is a passive splitter.
Passive meaning not powered. The way the mult is laid out is usually 3 or 4 side
by side patch points. It's wired from the back in such a way that when you plug a
signal in any one of the four, the other three become outputs. In other words
there is no set input, any of the 3 or 4 points will work as an input and the other
three become the outs. The mult is usually labeled as below:

How to use a Mult

Let's say you need to split off your signal form one output to two inputs. Say it's
tape track out 15 and you want it to go to inputs 15 and 16 on the console. Using
the mult, you could take output fifteen (remember if we were using the fully
normalled patch bay this would kill your normal to input 15 below), and go into
the mult. Then take any two outs from the mult and go to inputs 15 and 16 on
your console. The mult just solved our normaling problem.

Other Mult Uses

Most patch bays whether full or half normalled will have a number of mults for
you to use. They are handy things to have around and I say the more the better.
You could use a mult to:

1. Take your multi-track buss output and mult it to three tape machine inputs.
This way if you're doing overdubs on repeated tracks (this happens a lot when
doing vocals) all you have to do is flip one track out of record ready and flip
the other in. The signal is already multed to the new track and there's no need
to re-buss or re-route your signal.

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2. If you're trying to get a mono auxiliary out into a stereo effects processor's
inputs. You can go aux out into mult in, and take two outputs from the mult
into left and right of your effects device. For effects, most of the time you don't
need true stereo ins anyway so there's no need to tie up two auxiliaries.

3. Say you want to bring back a stereo track to the console in two places so you
can treat it differently in different parts of the song. Here you'd use two mults
to maintain discrete stereo left and right outputs. You'd take the left side into a
mult and then bring it back to console inputs 17 and 19, then take the right
side and split it twice from another mult into faders 18 and 20. This way you'd
have the two stereo groups side by side and you could EQ, Pan and treat the
reverb differently on both sets. Muting one when you needed the other and
vice versa.

What are Dynamic Processors?

Dynamic processors are devices that change the gain of your signal. For
instance, that would be a compressor, limiter, or a gate. When we talk about
console dynamics we're talking about a console coming loaded with these
devices on the individual channels and perhaps the stereo buss as well. It used
to be you had to spend big dollars to get this option. Traditionally SSL and Neve
"V" series consoles have a dynamics section built in as standard equipment.
These consoles sell in the hundreds of thousands of dollars range.

In recent years consoles in the $25,000 and up

price range have begun to offer dynamics as a
feature. The AMEK BIG for instance comes with
Virtual Dynamics. This is a software based limiter,
compressor, gate or panner that is available on
the first 24 channels and is a great leap in
technology for this price range.

How Do They Work?

In a previous feature we discussed parallel effects
processors and how they work, dynamic
processors work as serial devices. This means
instead of a piece of your signal going off and being processed then coming back
separately, your whole signal goes through the device in a serial fashion and
comes out altered.

So, unlike parallel processors, if you lose the connection to your serial device you
lose your signal altogether.

Why Use a Compressor?

There are two reasons you'd want to use a compressor on different mix
ingredients. You'd want to limit the dynamic range of something to be able to give

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it more prominence in the mix or to even out an uneven performance. This allows
you to lower the overall gain and yet hear the entire performance, thus bringing
up the apparent loudness of the signal. You'd want to pick and choose what you
want to compress based on the performance, musical style and what kind of
sound you're going for. Compressors can be overdone so use them sparingly to
maintain an open and transparent feel to the track. Also just because you have
compressors on every channel of a console doesn't mean that you'd use this
same type of compressor on all the things needing it. Use outboard compressors
for the important items like bass and lead vocals.

Why Use a Gate?

To clean up those spots in between parts of a performance where tape noise and
other spillover sounds like headphone leakage resides, you'd want to use a gate.
This will tend to open up a track by cleaning up all these unwanted lurking
sounds. I use gates on mix items where there are holes or spaces in the
performance. On kick and snare drums especially. Something that is constantly
playing does not need a gate because there are no holes to clean up.

The Starting Line

Just like you do certain things to setup your car when you start it, (seatbelt on,
car in park or neutral if it's a stick etc.) there is a basic starting point that you can
use when setting up compressors and gates. This makes it much easier to get
the device to do what you want it to do right off the bat. If your device has been
used in a previous session and is already setup for something different than the
instrument you're going to use it on, it can be a challenge to get it going. Not that
I'm not one for challenges, but if you can start out the device fresh your job will
be easier. The idea is to use the following setups as a starting point and then
tweak your signal until you get exactly what you want.

Compressors
Quite simply the routine for setting up a compressor is as follows:

• Threshold at zero
• Ratio at 2:1
• Output at zero

I'm purposefully using a very simple compressor such as the DBX-160X as a

template here. Other compressors may have more features but if you can start
out with this one and can master the basic parameters then you can easily jump
to something more complex using the same principles. Before we run through
what each parameter does let's explore compressors for a second. How
compressors work is that all audio below the threshold is untouched and all audio
above the threshold is compressed according to the ratio. The ratio works like
this: 2:1 would mean that for every 2dB over the threshold you'd get 1dB back,
this is called gain reduction. If you flip 2:1 over you get 1/2, this means you'd get
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half your level back of the signal over the threshold. 3:1, 4:1 etc all work the
same, you're just getting less back compared to what you put in. Various
compressors sound harsh or transparent based on the attack and release
settings. If you don't have attack and release settings these parameters are said
to be fixed.

Gates
Gates work by opening up and letting audio pass when the threshold is crossed
and closing when the signal falls below the threshold. Basic gate setup is as
follows:

• Threshold at zero
• Attack at the quickest setting (usually
turned left)
• Hold at the quickest setting (usually turned left)
• Release at the quickest setting (usually turned left)
• Range set to the lowest possible dB for maximum gating

Threshold of course is where your gate opens up, Attack is how fast it opens,
Hold is how long it stays open after the attack, Release is how it tails off (gradual
or quickly) and range is how low in level the gated signal will be heard.

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References

Borwick, John (ed.), Sound Recording Practice, (4th ed.), Oxford: Oxford University
Press, 1996.

Ballou, Glen (ed.), Handbook for Sound Engineers: The New Audio Cyclopedia,
(2nd ed.) Indiana: Howard W. Sams & Co., 1991.

Huber, David M. & Robert E. Runstein, Modern RecordingTechniques, (3rd ed.),

Indiana: Howard W. Sams & Co., 1992.

Davis, Don & Carolyn Davis, Sound System Engineering, (2nd ed.), Indiana:
Howard W. Sams & Co., 1994.

Davis, Gary & Ralph Jones, The Sound Reinforcement Handbook, (2nd ed.),
Milwaukee: Hal-Leonard Corporation, 1990.

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