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Behind
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Mountain
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Behind The Mountain

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Behind The Mountain - Recording Tips II
Back to Page I

A Compression
Kickstart
by
Ken Lanyon (Slider)
Compression has to be one of the most confusing and elusive
effects out there. It's easy to know you need it just by
watching your meters, but what does each knob and button really
do and how does it all work? This article should answer those
questions, and will touch on the "whens" and "whys" of
compression.
Let me first start by explaining the basics of dynamic ranges in
recording. First, we have the noise floor. This is the lowest
level, where tape hiss and electrical hum reside at. Next we
have the nominal level, which is the level that is best for
recording your incoming signal in order to minimize distortion
and overcome the noise floor. The distance between the noise
floor and the nominal level is called the signal-to-noise ratio.
Next is the maximum level, which is where distortion occurs at
when your incoming level reaches it. This is the highest level
in the total dynamic range. Distortion is something that you
definitely want to avoid unless you are versed in the skills of
good tape saturation (sometimes engineers will try to slightly
distort the signal by pushing it over the maximum level because
this will give a stronger sound to an originally weak one.
However, in digital recording, any distortion due to overpeaking
is distasteful.). Now the difference between the nominal level
and the maximum level is referred to as your headroom. This is
your safety zone, and this is needed to account for some stray
peaks here and there without hitting the maximum level. And to
wrap this up...the whole thing, from noise floor to the maximum
level is called the dynamic range.
Okay, lets cover how compressors work. Imagine a recording
scenario where you are starting to record some tracks on your
multitrack recorder. You have set a good recording level for
your instrument which is at or near the nominal level, but you
notice that the incoming signal occasionally jumps up into the
red. That is typically going to be the nature of either the
instrument, your playing, or both. So, you don't want those
distortions going to tape and ruining an otherwise fine
performance. This is where the compressor comes in handy.
The Alesis Company a while back issued a brochure on how
compressors work, and it gives the analogy of the compressor
acting like your own dedicated engineer for that one track. It
will monitor all the incoming signals and then act like it is
pulling down the fader the instant that high volume peak occurs.
In a more technical explanation, what the compressor is actually
doing is reading the incoming signals, and then according to the
compression ratio that you set, it knocks the hot signal down by
that ratio. This allows you to keep the level down to one that
is manageable and recordable, without the wild peaks.
Compression ratio you ask? Well, let me explain the 5 main
controls. First, we have the threshold. Think of this as the
decibel level where the compression will start working. I
visualize the threshold as a line that is lowered onto the
offending noise peak, and the lower the threshold level, the
more the incoming signal will be compressed. This is because
more of the noise peak is now ABOVE the threshold level so there
is more to squash.
Next we have the ratio settings. This knob has different ratios
on it like 2:1, 3:1, 4:1, and usually any combination in
between. Okay, assume you set your ratio to 3:1. What this does
is that for every 3dB your incoming signal goes over your
threshold line, the compressor will allow only 1dB to pass. The
level still goes over the threshold, but assuming that you set
the threshold low enough and used an appropriate ratio, the peak
will never reach the maximum level and distort. This is also due
to the amount of headroom you have. Typically, I set my ratio
first, and then use the threshold knob to find the point that
the incoming levels are being compressed. This is done while
watching the meters on the mixer, and you will see the offending
peaks all falling within the same lower range which is nearer to
the nominal level. Keep in mind that if your incoming signal is
lower than the threshold level, (or the threshold is set too
high), then none of the signal will be affected.
Next we have the attack parameter. Think of this as how fast the
compressor acts on the peaks once they pass the threshold. Some
instruments have a really quick attack sound as soon as they are
played, and most peaks arise from this attack. Therefore, on
instruments like bass and kick drums, you would want to set a
quick attack.
The release parameter works by setting how fast the compressor
lets go of the incoming signal once it has gone below the
threshold level (where the signal doesn't need to be compressed
anymore.) You could set the release to fast and cut off a signal
quickly, or set it to slow which results in a longer sustain.
Many guitar players use this to sustain their notes.
The last main function is the output setting. Typically, when
you lower the threshold and the compressor kicks in to squash
the signal, your nominal level will be lowered slightly
depending on the amount of compression being used. You can then
use the output knob to bring the input level back up to nominal.
Be careful though, because by raising your signal back to the
nominal level, you are also increasing the noise floor due to
added noise from within the compressor itself. You may want to
increase the trim on your channel or master fader so more pure
signal is getting to the compressor instead. Everytime you patch
your signal through another pathway (such as another processor),
you are also adding the inherent noise of that pathway.
There is one other feature that not all compressors have, and
this is the option to compress with "hard knee" or "soft knee".
Hard knee is where the signal is compressed the moment it goes
above the threshold to the full extent of the ratio that is set.
Soft knee is where the compression is applied more softly so as
not to sound so abrupt. This is similar to using the attack
knob, and I use hard knee compression on signals like bass and
kickdrum.
Hooking up a compressor is a simple procedure involving an
insert cable. This is a Y configuration cable with one 1/4" TRS
connector that splits out to two 1/4" connectors. One of these
connectors is an RS and the other the TS. (I should mention here
that TRS stands for TIP -RING-SLEEVE, with the tip being the
send and the ring being the return. This way, the TRS connector
allows signals to go both ways, and the TS connector allows on
signals to send FROM the compressor to the mixer while the RS
connector returns the signal from the mixer to the compressor.)
The TRS end is plugged into the insert jack on one channel of
your mixer, the TS to the compressor send, and RS to the
compressor return. This creates a loop where the original signal
leaves the mixer, goes to the compressor, is then compressed,
and finally returns to the mixer.
As for using compression, that is a matter of personal
preference. I use it only when needed. Unless I am going for a
certain type of effect by heavil y compressing the signal, then
I use it only for stray peaks, since putting a signal that isn't
peaking through a compressor will only introduce more noise.
Some people think that even though the signal is peaking out
during recording, they can compress the signal in the mix and it
will be the same. I used to think that myself but I realize now
that when you put a distorting signal to tape, the damage is
already done to that signal's sound. The track is already
saturated with distortion and no amount of compression during
the mix will make it sound as if it were compressed during
tracking. That is why you should definitely fix stray peaks with
the compressor when recording. Also, final mixes may also need a
little compression even if you used it on tracks during
recording. This is due to the summation of all the track
signals.
The following are just suggestions of where to start setting
your parameters for certain instruments. As I mentioned earlier,
how YOU want to use compression is your personal preference.
Bass: Try starting out with a ratio of 4:1, and a fast attack
and release. I usually use the hard-knee type of compression
here since bass is such an attack-oriented instrument. But if
you were playing smooth jazz bass, then you may want to try
soft-knee. It depends on the sound you are trying to get.
Guitar: This depends on the type of sound you are using, but a
good general place to start is 2:1 on acoustic, and maybe 3:1 on
distorted guitar (although you may need 4:1 here.) To get a good
sustain, try a 4:1 ratio, use a fast attack and slow release.
Then play the note you want to sustain, and raise the ratio
until the sustain is as long as you want it.
Drums: Drum signals are often compressed due to their
hard-hitting attack volumes. If nothing else, compress the snare
drum, because each hit will likely peak higher than other hits.
Try starting out with a ratio of 3:1, and use a fast attack and
release. If the signal is still peaking, try using a ratio of
4:1. This method could also be applied to the toms. As for
cymbal hits, try starting with a 2:1 ratio (moving to 3:1 if
needed), using a fast attack and a slow release (to preserve the
natural decay time of cymbals).
Vocals: As with drums, compression is also commonly used on
vocals. The ratio to start at varies for each singer, since some
may be very strong and loud singers, and others quieter, having
a smaller dynamic range. Try starting out with a 2:1 ratio, with
a fast attack, and medium to slow release. Keep increasing your
ratio until you get your peaks under control.
Compression is not typically something that can be heard. You
can hear it if you really spank all the knobs to full-on, but
usually that technique is used more for an effect, rather than
to control the level of the individual signal. Compression
should be applied and monitored by using the peak display meters
on your compressor or mixer. As I mentioned earlier, compression
is something of an art, and you will have to play with it to
find your personal preferences, so don't be afraid to tweak all
the knobs to find out how they affect your sounds. Just remember
that mastering compression techniques will help to make all of
your recordings sound more professional.
(c) 2000, Ken Lanyon,
All rights reserved.
(You are allowed to copy and use this essay for your own
non-professional use. You are prohibited from distributing
copies to others for a fee or for no-charge. You may not publish
or quote this essay without obtaining the written permission of
the author.)

An Introduction To Mastering
by
Stephen J. Baldassarre (Silent Bob)
As a mastering engineer, many people have asked me about the
importance of mastering. However, in order to thoroughly
describe the importance of mastering, I must first describe some
of the equipment and processes available to a typical mastering
engineer.
The equipment used by mastering engineers is very specialized
and precise. Most people have dynamic compressors in their
studios but the compressors used in mastering are a bit more
complicated. For instance, I use compression that can control
high and low frequencies independently. It can catch peaks in
the audio signal instantly or before the peaks even occur. This
compression uses joint stereo operation which means that if a
peak occurs on one channel of the stereo mix, both channels
(right and left audio channels) with be attenuated equally. This
is important because if only one channel is attenuated, there
will be a sudden loss in one channel's volume which will
interfere with the soundscape. Joint stereo operation also
prevents stereo separation from deteriorating as compression is
increased.
Most people are also very familiar with equalizers or EQ. The EQ
used in mastering can affect both right and left channels
independently or identically. This is useful if the right and
left channels have significantly different frequency content or
if there is an error in one channel and not the other (if it
ain't broke, don't fix it). Also, I can use EQ from a ten-band
analogue EQ all the way to 2,400 band digital FFT filters. FFT
means Fast Fourier Transform, which is a method of processing a
digital signal using discrete amounts of delay to control
independent bands of frequencies Why so many bands? Precision,
that's why. I've mastered songs with high pitched ringing going
on throughout caused by substandard equipment or from having a
computer too close to the recording gear. Normal EQ could
eliminate such sounds but would cause severe interference with
the rest of the program material making it sound unnatural. The
digital EQ is so precise that it can eliminate the ringing
without any audible effect on the program material. It can also
be used for split seconds to reduce bum notes or add a little
accent to certain instruments without affecting the surrounding
material. This is very useful for increasing clarity and overall
impact of the sound.
Nonlinear editing tools such as a software controlled hard drive
system are also important for removing sections of sound for the
purpose of making different versions of songs for radio or album
cuts, CD singles etc. Fixing bad "punch-in" glitches, and
cleaning up fades are also advantages of nonlinear editing
tools. The same tools are used to put the songs or other
material in the correct order and set the correct timing between
tracks on CDs. Dynamics can also be added with great precision
to program material using a nonlinear editing system to increase
the impact of the sound. One other real advantage of a nonlinear
system is the ability to reduce transients (occasional sudden
volume peaks), which prevent the overall volume of the material
from being increased. After stray transients have been removed,
the signal can usually be boosted 3-9dB louder than before.
Noise reduction is also a very handy tool in mastering. The same
FFT filter used for EQ can also be used to remove AC hum, tape
hiss (to a limited extent) or other unwanted noises such as
clicks and pops. If there is noise in a particular track like AC
hum, a segment of the track containing only noise can be sampled
in the FFT as a profile for noise reduction. This profile is
applied over the entire selection and (hopefully) attenuates the
noise. This is incredibly useful for restoring older recordings,
but many new projects I've worked on have also benefited from
this process.
Mastering engineers also have the ability to widen the stereo
field of recordings, even if they were originally recorded in
mono. Granted, if you send a mono recording to a mastering
house, they cannot, for instance, pan the guitar to the right
and the keyboard to the left, but they can add stereo space that
was not there originally. If the recording is done in stereo but
just does not have the aural space it needs, then the stereo
field can be accented, creating an improved soundscape. There
are several methods of doing this that can only be done in the
digital domain, but some methods are done using specialized
analogue processors.
One of the last mastering tricks I should mention is time
stretching. A song's tempo can be increased or decreased without
affecting the pitch of the song. This is important for making
radio edits of songs, as radio programmers have a tendency to
speed up songs in order to fit more commercials into the day.
The tempo of the song can be decreased so when the radio station
speeds it up, it will have the tempo it was originally intended
to have. There isn't a large demand for this process, but some
people wanting to make their tunes more danceable or to cheat
the radio stations like to have this option.
So when people ask me what the importance of mastering is, I
could sum it up into just a few short statements. Mastering
increases the impact and clarity of the material. It is the
final polishing an album as a whole receives before it is
released to the public. Final touches on fades, song order and
volume are all made here as well as some correctional touch-ups.
Who should have their stuff mastered? Anybody looking for a more
professional sound in their work should have their material
mastered. Mastering is a key process in bringing recordings up
to commercial standards. Home-recorded demos all the way to
industrial studio recordings can benefit from mastering, which
is why I stress the importance of it so much. Industrial studios
have their material mastered religiously to gain that extra
edge. Many audiophiles have their material mastered to compete
with the industrial studios, and musicians with homemade demos
may have it done just to increase the impact of their sound for
promotional use. So mastering can serve anybody who is looking
for a more professional sound in their music. For audiophiles,
it is a great help for achieving the perfect sound. For
industrial studios, it is a step all to important to skip.
(c) 2000, Stephen J. Baldassarre,
All rights reserved.
(You
are allowed to copy and use this essay for your own
non-professional use. You are prohibited from distributing
copies to others for a fee or for no-charge. You may not publish
or quote this essay without obtaining the written permission of
the author.)
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