Using Noise Gates.
A "Noise Gate" - as many of you will know already - is a device that only
lets sound through if it is louder than a set amount. So in some ways it is a
bit like a dodgy connection lead! - except that:
- You can precisely calibrate when it cuts in
- It doesn't crackle when it cuts out - the sound fades gently away at a
predetermined rate
Gates are used to cut out unwanted background sound when an instrument isn't
playing. Naturally when the instrument does play then you will hear
the background sound switch in as well as the instrument. This doesn't usually
matter, because the playing of the instrument normally masks the background
noise. You could use a gate - for example - to cut out all that amplifier hum
and hiss in-between parts of a guitar solo - it would certainly be objectionable
to have it going on throughout the whole song when the guitar isn't playing.
Of course there is normally some sound quality loss when going through an
analog noise-gate, but this is not usually significant and it shouldn't concern
you. Also, if you are using a compressor on something already, then any built-in
noise gate on that compressor will use the same gain-control circuitry anyway
and so there is no further loss in signal quality - the gate effectively comes
"for free".
If you are using a mastering compressor, then its built-in noise gate is
useful as it can act as a quick-and-easy way of trimming off all that background
noise before the first downbeat of a song. It certainly saves a lot of time by
eliminating the need to edit by hand in a sound editor later!
However, in many individual instrument cases within a multitrack recording,
you can do a better job by hand instead of using a noise gate. Either by erasing
the parts of tracks where the instruments are not meant to be playing (be
careful though!), or simply by automating the mix to mute channels when people
aren't supposed to be playing.
Where noise gates come into their own is when gating a signal that comes and
goes rapidly - and drums are a prime example of this. It would obviously take an
impossible amount of time to try and accurately "erase" the gap between every
signal snare drum beat, and this is especially the kind of situation where the
automatic nature of a noise gate is at its best.
Specifically, there are five uses of noise gates that I can think of (at this
moment) that relate directly to drums:
- Gating sampled drums to shorten their length
- Creating "gated" reverb
- Helping to eliminate "spill"
- Helping to eliminate "resonance"
- Creating a "clinically clean" drum sound
Lets look at each of these in detail.
In each case when gating drums, the "Attack" should normally be set to its
fastest, unless you are using a very, very fast gate which might
introduce an audible "click" when the gates cuts in.
Gating sampled drums to shorten their length
There's little (or no) benefit in gating sampled sounds from a drum machine
simply to remove noise, as the noise is not normally significant (an obvious
exception to this is if you have taken samples from another recording which is
noisy). However, if you are mixing a pre-recorded tape, you can use a noise gate
to shorten the length of (for example) the snare drum. This is something that
you would otherwise not be able to do without access to the original sampler or
drum machine. The "hold" and - particularly - the "release" controls are the
ones that affect this.
Creating "gated" reverb
These days, most digital reverbs have a "gated" reverb preset, so the need to
gate reverb by hand has almost completely disappeared. Previously, you would
have to feed the stereo reverb return through a pair of "linked" noise gates,
and feed the "side chain" of the gate - the signal path that triggers it to open
- from the direct snare sound. You would need to use a noise gate with a precise
"hold" control - which keeps the gate open after it has triggered - in order to
specify how long the "gated" sound lasts. An abrupt "release" setting after a
fairly long "hold" setting usually gives the most dramatic effect. Normally,
exactly one beat or half-beat sounds about right. "Drawmer" noise gates became
famous as the best tool to achieve this in the analog world.
Compressing ambient microphones (if present on the multitrack recording) can
lead to a spectacularly loud sounding drum kit, and gating such a stereo signal
in the same way as gating reverb can heighten the dramatic effect even more.
Helping to eliminate "spill"
With a real drum kit, most microphones will pick up a significant amount of
sound from the other drums. This can severely restrict your ability to EQ one
drum without it affecting the rest of the drum balance. By gating each drum
separately, this gives you more flexibility. But there are a couple of problems.
Firstly, when you EQ the drums, this can easily make the spill from other
drums so loud that the gate "misfires" and opens up when it shouldn't. If you
raise the "threshold" control to compensate, you risk not triggering on
important drum beats. The classic problem case for this, is a loud hi-hat
causing a snare drum gate which has had a lot of high EQ added (for a brighter
snare sound) to open on a hi-hat beat instead of a snare beat. This sounds
particularly loud and objectionable, and really sounds like a mistake
when it happens. One solution to this is to gate the snare before the
equaliser. Some gates also have separate, simple EQ controls on the trigger
signal so you can roll off (for example) the high-frequency troublesome hi-hat
spill which is causing the gate to misfire. However the problem with doing this,
is that it can make the gate open a little bit too late, losing some of the
impact of the snare drum. Sometimes the only way to solve this, is to duplicate
the snare drum track, move the copied track back a few milliseconds, and use
that as the basis for the signal feeding the gates "sidechain". This will make
the gate open up before the snare beat has even happened which is pretty
cool. Some software controlled gates (even hardware-based digital ones) have a
delay available for the direct signal path in order to achieve much the same
result. Naturally in this case, it means that you will have to move the master
track back a few seconds in order to compensate, but on digital multitracks this
is relatively easy, and it avoids the inconvenience of having to prepare a
separate snare track just to trigger the gate. If you do this, remember to jot
this down in the notes that accompany the multitrack, as people using the
multitrack later will simply think that the snare is out of time with everything
else.
Helping to eliminate "resonance"
Related to the problem of "spill" is the problem of "resonance". A real drum
kit has drum skins that resonate in sympathy with each other. For example, every
time that the drummer uses the bass drum, tom-toms will often emit a faint "boo"
sound. Similarly, when the bass drum is used, the snare drum often has an
irritating sympathetic "rattle", which sounds in the final mix like a rattly
bass drum. Gating the drums can help eliminate these types of unwanted noises.
If gating makes the kit sound too clean, it is possible on many gates, to let a
little of the signal through at all times, so that you are not plunged into
total silence when the gates kick in. This is sometimes called "soft gating".
The techniques for gating resonance out of drums are similar to gating
"spill". The main difference is that when equalising the trigger signal (the
"sidechain"), you sometimes have to do the opposite of what you do for
spill. For example, to remove resonance on toms, you often have to use a trigger
signal that has had the top AND the bottom rolled off - otherwise the "boo"
sound of the resonance as well as hi-hat spill might cause the gate to misfire.
By using a combination of high-frequency and low-frequency rolloff (which are
often provided on the gates themselves), it is usually possible to "narrow" the
side chain signal to something that makes the gate trigger at the appropriate
time, without cutting off the front of the beat.
Cautions on gating a "noisy" drum kit
Gating is not the "holy grail" of a getting a clean drum sound. A
clean-sounding drum kit is the best solution. When recording, it is far
better to try and persuade the drummer to investigate and eliminate any
resonance in their kit rather than thinking that it can all be gated "in the
mix". Sometimes, all this takes is careful tuning of the drum kit to avoid
resonance and rattles. Other times, "gaffer" tape has to be stuck on the skins
of the drums in carefully-chosen places to dampen the resonant frequencies and
make the kit sound "tighter". In other cases, you may be regrettably short of
time and have to (unfortunately) fix the sound later.
When resonance and spill are severe, gating can ironically draw attention to
them (in the form of "breathing", which is the name given to the audible effect
of gates and compressors that can obviously be heard working). When resonance
and spill are very severe, it can be beyond the ability of gates to fix the
problem, and you may have to be more imaginative in looking for a solution.
The most common problem caused by severe spill affecting gating, is when toms
that sounded too dull need to be brightened on mixing (just to make the sound
acceptable). If there is a lot of spill from the cymbals onto the drum tracks,
you can find that if you gate the toms, a rather unpleasant effect occurs when a
drum fill follows a loud cymbal crash. What happens is that the cymbals appear
to "pulsate" in a thrashy way every time a tom is hit. This is the sound of the
Cymbal "spill" breaking through, and is very unpleasant. Perhaps you might
decide to take an alternative approach to fixing the problem, such as actually
making the toms sound even duller than they did before, by rolling off
the top end, and then using something like an Aphex Aural Exciter to
re-synthesize the missing top end. That will give you a new top end, without the
same level of cymbal breakthrough.
Often when mixing, I prefer not to gate the toms at all, and if gating would
otherwise be required - I simply automate the mutes on the tom channels so that
they are only switched on when a genuine tom fill occurs. If this makes the drum
kit sound too artificial, I use soft fader movements instead to simply lower the
level of the toms when they are not being played. I prefer to do this rather
than using gates, because otherwise you can find that the whole stereo image of
the drum kit changes and moves when the fills are played. Another solution would
be to gate the toms as a pair rather than individually. Sometimes this works,
but by no means always.
Creating a "clinically clean" drum sound
Finally, even if spill and resonance are not a significant problem in
themselves, you can use gates to eliminate even the merest hint of them, and
gate every single drum down to its bare essential sound. This will give you a
"clinically clean" drum sound, but bear in mind that it will also strip a lot of
"life" out of the sound, and you'll probably need a good deal of very
high-quality short reverb to put some life back in. The result will probably
still sound artificial.
Summary of gating drums
As you can see, gates can be a lot of hard work. If all you are trying to do
is to clean up the drum sound a bit, consider simply muting the toms in-between
fills using mixer automation (mute and un-mute on (e.g.) snare beats to cover
the change in sound). That is frequently all that is required to tidy up the
drum sound. Pan-global gating is not usually required and can sound contrived
and unnatural.
If you can get away without gating or muting any drum tracks at all, then so
much the better.
In any case - as I mentioned at the beginning of this section, make these
decisions about the drum sound at this early stage of the mix and not later. If
you gate drums towards the end of a mix, the whole atmosphere of the mix can
change in a disconcerting way.
Adding in the Padding
Not all songs have a "pad" sound. Some songs work by placing all the
responsibility on the lead instruments. However, a good "pad" sound adds body to
the track and can also hide a multitude of sins in the playing of the other
musical parts - it's a simple and convenient way of making a track sound "full".
A "pad" is a simple musical part playing a straight (often oversimplified)
chord sequence in a middle to low register throughout the entire track. Usually
performed on a guitar or keyboard, the choice of sound is normally warm and
subtle. Pads are usually recorded in stereo - either as a result of the sound
itself (e.g. keyboards), or by the use of effects during recording (stereo
chorus on guitar), or - particularly in the case of guitars, by double-tracking
in stereo with each of two takes panned left and right.
If present, the pad is absolutely key to the sound and feel of the song
because normally every other instrument part was designed and played with the
pad already in place. It really forms the "foundation stone" on which the rest
of the track is built. From time to time as you mix, try muting it and see what
happens. It feels like someone has literally pulled the rug from under your
feet! (interestingly, a friend of mine labels pads as "carpet" on the track
sheet). What remains of the song without the pad in place will sound "suspended"
in space without visible (audible?) means of support, and probably will sound
very strange.
The challenge in getting a good pad sound is making it warm, wide and full,
but yet somehow transparent at the same time. You'll need that transparency
because it leaves space in the mix in which to place the other instruments. If
the pad sounds in any way "stodgy", it will get in the way of the other
instruments that you intend to add to the mix later, so you need to get that
"transparency" correct NOW before you add the other parts in.
The "pad" should feel like a warm blanket wrapped around the song. I don't
think that is a particularly over-the-top analogy - it is certainly how all
"pad" parts sound to me. So thinking "warm blanket" may help you. The pad, in
particular, should fill the sound stage without having a particularly defined
character. If you give the pad too much character then it will distract from all
the other goodies you are going to put into the mix later. Also, because pads
generally play from the start of the song to the end, they can get boring if
they stand out too much. They are there primarily for structural support for the
song.
Stereo Pad Tips:
- Check the pad on headphones. Many keyboard pads are often too wide for
headphones and feel unpleasant and disorienting, so reduce the extremity of
the stereo width, so that you feel there is at least something in the
middle when listening on headphones.
- Some tracks use a double-tracked part for stereo. This often happens on
guitar-based tracks where you get two independent strummy-type parts which are
designed to be placed left and right. Such parts can sound really nice, but
they usually work best if they are NOT panned fully left/right, as the sound
will be too wide, and you won't get the full chorusy sound of the two guitars
interacting together.
- If you really want the two independent guitar parts to be panned
hard left and hard right, then you can thicken them up with a simple
technique. For the left hand guitar, add a totally wet, but simple chorus
effect return panned 50% off centre to the right. Similarly, for the right
hand guitar, add another totally wet, simple chorus effect return panned 50%
off centre to the left. This often works, but is sometimes over-the-top. Swap
the panpot positioning of the chorused counterparts with their direct-sound
partners, and reduce the level of the chorused parts for a more subtle effect.
- If you have just a single pad track in mono, it is usually worth spending
some time with a high-quality chorus unit converting the mono sound to a wide
all-embracing stereo one.
- Specifically for pad parts, if you have two separate instruments playing
"pad" type parts simultaneously, it is often more effective to leave them
panned to the centre to act as one, and use high-quality chorus to create
stereo wideness, rather than simply panning the two parts to different
positions.
- The pad should be stimulating both of your ears independently, but not so
much as to leave a gaping whole in the middle. Reduce the stereo width if in
doubt, and - as mentioned above - check on headphones.
Pad Sound Tips:
- As you can tell, chorus is important to pad parts, but such chorus should
most definitely not be a stereo swirly mess. Really take the
time to experiment with the settings on your chorus unit to see how subtle you
can get it. Keep delay times short, and modulation speeds low and you can get
a chorus that really enhances the high frequency end of a track without it
getting too, ..well.., "chorussy"!
- Some instruments don't suit chorus at all. Piano is a classic. Put chorus
on a piano and it sounds out-of-tune at best, and plain old cheap honky-tonk
at worst. Let your ears be the ruling judge of whether chorus is even required
at all.
- Sucking out a fair bit of 700-800 Hz (or thereabouts), and cutting back a
bit of the low end, on pad-like sounds, with a fairly wide bandwidth EQ,
softens them and gives them the transparent "hi-fi" sound which serves as a
nice backdrop to the rest of the mix. It "restrains" the sound, dampening any
aggression, and the fact that the high frequencies are allowed to rise up
again, can add a silky sheen to both keyboards and guitars alike. Make sure it
doesn't swamp the bottom end of the mix if you do this.
Pad Sound Check:
You've now got drums, bass and pad in place. At this point, the song should
sound amazing. Yup! even with just those three elements, it should really feel
special. The backing track should sound "complete". You should feel like simply
adding the vocal would be enough. This should really be true of any instrument
as you add it into the mix, so I won't say it any more. Don't forget to
double-check the sound with all the remaining elements (from the rough mix) in
place too, or you might have difficulty getting them to "conform" to the mix
later.
Time for a Break
Personally this is my favourite point in the mix, as it is pure and unsullied
by the rest of the twiddly bits coming up later, and you shouldn't feel
particularly tired at this point. You should make a note of this mix - either by
storing it in software or by simply making chinagraph marks to the side of real
faders to mark their positions.
Why? Because later on, if the mix doesn't seem to be working, then this
simple mix is a great reference point to compare each individual instrument to,
in order to diagnose where the problems are.
Also, if you are doing an "extended mix", then this simple drums/bass/pad mix
is a great thing to suddenly drop down to during a breakdown section of such a
remix - perhaps with some dramatic percussion bashing about over the top? (just
a suggestion).
When you have these three (or so) elements in place, then you should be very
careful not to go changing the sound of any of them in future unless absolutely
necessary. The mix of bass, drums, and pad, is really a "signing off" point in
the mixing the song.
Time for you to have some more coffee now...
This is also a good time - if you are using a PC-Based system (such as
n-Track) - To "render" the drums, bass and "pad" sound to a single stereo track.
This should free up some CPU power for you to use on the lead parts. You should
obviously file away the original parts away for safe keeping - perhaps the
easiest way being to save the project as it currently is now to a
safe place, then re-save it to the current working directory before
removing the original separate parts from the current project.
Adding the Main Parts
Probably the first thing we need to get out of the way right now, is a more
detailed examination of the role of EQ in mixing.
The Use of Equalisation
Equalisation - or EQ, to give it its common acronym - is fundamental to
modern recording and mixing. However, its application is still very much hailed
as a "black art" and many engineers don't like discussing openly their use of EQ
in case they give away their own "trade secrets".
To confuse matters further, a lot of the common guidelines established over
the years are touted around by educators as "rules" that can only be broken at
your peril. Rules such as "get the sound right at source", when overemphasised,
can prevent a junior engineer from experimenting with the more subtle
applications of equalisation, to the extent that they "play safe" too often - to
the detriment of their mixes and their education.
This - coupled with the fact that poor EQ practice can ruin a good sound -
can also lead modern record producers to be unnecessarily paranoid about the use
of EQ, and cause them to slap the wrists of engineers who reach for the EQ
controls when the sound is already "basically acceptable" without understanding
that the engineer may have other important reasons to tweak the EQ.
EQ isn't a necessary evil - it is instead, a highly versatile tool,
which can be used for a wide variety of different applications which go far
beyond an engineer simply "playing with the sound". In fact, it may come as a
surprise to you to discover that there are many, many different uses of
EQ, and so to demonstrate this, I will go into some detail on seven of the major
uses of EQ when mixing (and indeed recording):
- Noise Elimination
- Harmonic Damping
- Sound Enhancement
- Bong and Boff
- Distance Placement
- Creating Mix "Space"
- Auto Mix "Levelling"
I'll explain what I mean be these in a minute - but first:
Types of Equaliser
There are many different kinds of equaliser available to do these tasks.
These are (in order of sophistication):
Filters
Simple filters are extremely useful on mixing and recording systems. They are
more correctly known as "High (or Low) Frequency Roll Off Filters", or - with
confusingly opposite names - as "Low (or High) Pass Filters". They either take
the form of a single button to roll off the bottom end, or perhaps instead two
rotary controls marked LF and HF. As you sweep these controls, the top and
bottom end is "rolled away" from the signal. The frequency at which this happens
depends on the position of the control. You can't control how steeply the sound
rolls away - that is always preset. The manual either for your software or
mixing hardware will say what it is preset to. It will be described as something
like "6dB per octave" or "12dB per octave".
Not all mixing desks have simple, dedicated filters. On budget desks, it is
assumed that the task can be done with by careful use of the main EQ, although
this is a little restrictive - because having even just a single button for LF
rolloff is extremely useful - especially when recording.
Shelf Equalisers
The filters described above don't raise or lower the treble and bass in
general terms - the sound is actually trimmed away to nothing the higher or
lower the sound gets. This makes filters good for getting rid of undesirable
sounds outside of the musical range of the instrument (see section on Noise
Reduction Using Equalisation, below).
But, unfortunately, filters are not very good for shaping
the actual musical sound itself. That's where Shelf Equalisers come in. Just
like filters, they come in two types - low frequency and high frequency.
However, instead of having "Frequency" controls, they have "Gain" controls
instead. The gain controls set how much boost or cut is applied to the signal. A
high frequency shelf EQ will normally start working at around 8 to 12 kHz, and a
low frequency shelf EQ works at around 80 to 150 Hz. The point about shelf
equalisers, is that - unlike rolloff filters, they raise all frequencies
above (in the case of high-frequency shelf EQ) or below (in the case of a
low-frequency shelf EQ) by the same amount, which will give you much more
"musical" results than rolloff filters. They are called "shelf" equalisers
because of the shape of their frequency response when drawn on a graph.
The "Bass" and "Treble" controls on ordinary Hi-fi equipment are usually
shelf equalisers with preset, unadjustable frequency points.
Some shelf equalisers allow you to set the frequencies at which they operate
as well.
Sweep Equalisers
The problem with both filters and shelf equalisers, is that they are really
only useful at either end of the audio spectrum.
So what about the middle?
Well, if you think about it, you can boost the middle by using a high
and low shelf EQ to cut the bits either side, or - similarly - you can
cut the middle by boosting the bits either side. But this is not very
satisfactory, and difficult to control quickly and easily.
A sweep equaliser is designed to solve this problem. With a sweep equaliser,
only the parts of the signal surrounding the area selected by the
frequency control are affected - by the amount set using the gain
control. This allows you to boost or cut selected areas of sound quite easily.
It isn't possible to control the width of the sonic "area" affected by the EQ,
and this width varies from manufacturer to manufacturer. Some manufacturers like
to keep the area fairly broad, as this is more musical, but others prefer to
keep it narrow as this is more useful for correcting harmonic problems like
"ringing" on drums or other instruments.
Sweep equalisers normally come in pairs. A pair of sweep equalisers is often
found on some of the better-quality "portastudio" devices, as it allows a great
deal of sonic control for relatively little cost. Some manufacturers give
completely different frequency ranges for each one of the pair, arguing that
this allows for more precise control over the entire range of sound. This is
acceptable providing that the preset "area" or "bandwidth" of the equaliser is
extremely broad. If the bandwidth is smaller ("tighter"), then it is far more
useful for a pair of sweep equalisers to have a generous overlap in their
frequency ranges, so that you can use both of them at the same time in
both the low and upper frequency ranges.
Semi Parametric
To make sweep equalisers more useful, they are sometimes fitted with a single
button that changes the "bandwidth" or "area" over which they are effective.
When one or more sweep equalisers are teamed up with a couple of additional
shelf equalisers for control of the very top and bottom end, the entire assembly
of equalisers is referred to as a semi-parametric EQ.
Fully Parametric
On top-of-the-range mixing desks, you normally have fully parametric
equalisation.
With a fully parametric equaliser, you can control the frequency, the
gain, and - significantly - the bandwidth of the equaliser.
Typically, four of these units are packed together, and a switch on both
the first and last unit, allows them to be optionally used as shelf high
and low frequency EQ's respectively. Naturally, this is expensive.
For most instruments, you don't need this level of control, and semi-parametric
equalisers are fine. Even in a professional set up, you can often get by quite
adequately using semi-parametric equalisers, provided that there are a couple of
plug-in fully parametric equalisers available for troublesome instruments. The
main problem however with a mixing desk with semi-parametric equalisers
throughout, is that the manufacturers preset choice of bandwidth, can result in
the "sound of the desk" colouring the mix overall, to a much greater degree than
it does with fully parametric EQ.
But having said that - in almost all cases - the "sound" of the EQ is
probably still the major distinguishing feature that separates one mixing
console from another.
Graphic Equalisers
These are by and large an overkill They are best reserved for situations
where a number of extremely subtle audio artefacts are already taking place -
such as the equalisation of a recording studio control room's main monitors, or
the delicate final equalisation of a finished mix during the mastering process
when preparing a CD or other release for listening by the public at large.
Graphic equalisers are much less suited to situations where the actual
"correction" required is more "general" such as EQing individual instruments as
part of a mix - although it must be said, with some particularly difficult
sounds, you may occasionally have to resort to using a graphic equaliser to
solve the problem - but not generally.
Passive and Valve Equalisers
It's worth mentioning in passing here, the subject of passive equalisers.
Most equalisers use circuitry that actively boosts or cuts the sound in the
various bands using electronic feedback techniques which can (and often does)
introduce audible "ringing" in the circuitry. Passive equalisers on the other
hand work by already cutting the sound in all frequency bands to
begin with, using simple, unpowered, passive electrical components like
resistors, capacitors and inductors. A single, simply-designed amplifier stage
after the equaliser usually makes up for the loss in signal level, so
that the level is flat when all controls are in their "centre" position.
Therefore, on such an equaliser, when you "boost" a frequency, you're not really
boosting it at all - you're just allowing it to seep through unhindered by the
passive circuitry. Valve equalisers often work in exactly this way.
This gives a much smoother sound. In fact a lot of the "smoothness"
attributed to valve equalisers, often has little to do with the fact that they
have valves in - it is instead due to the fact that the equaliser circuitry is
passive rather than active.
So how do you use these all these different types of equalisers in
practice?
Let me give some practical examples:
Noise Reduction using EQ
The act of eliminating unwanted noise from a signal, is obviously one
application where it is far better to get the sound right at source. Although
this article is about mixing, it is worth mentioning here what the problems are,
and how they are normally solved at the recording stage.
The elimination of rumbles is best achieved by careful microphone placement
and mounting, and the elimination of hum is best achieved by both the careful
choice and placement of the cables used - normally "balanced" cables.
Elimination of high-frequency radio interference is also best achieved by the
careful choice and placement of balanced cables, but because high-frequency
interference is so pervasive, this might not always help. Taxicabs for example
are often a source of RF interference on microphone cables - especially if those
cables are unbalanced.
Hiss originating in microphones can be eliminated by the use of high-output
microphones coupled with high-performance balanced microphone pre-amps which in
the nineties, are now present even in fairly "budget" mixing consoles.
Avoidance of hiss from a recording device (i.e. tape machine) is best
achieved by the use of a good noise-reduction system such as Dolby, or by
recording on a good digital system in the first place - unfortunately, the
digital to analogue convertors on budget recording systems (like some PC sound
cards) don't even come close to the high-performance that high-performance
digital or even top-end analog systems can theoretically achieve.
Also, in practice, avoiding all of these sources of unwanted interference is
not always possible. Even the best isolated city-centre recording studio can
suffer from very low-frequency rumbles caused by nearby traffic and trains, and
the busy lifestyle in a city provides ample sources of radio interference such
as power lines, mobile phones, and taxicabs.
Despite the noble goals of the "purists" of sound recording theory, it isn't
always a practical idea to record all microphones "flat" (without EQ). Although
you might not be able to "hear" low-frequency rumble from nearby traffic and
other extraneous sources, you can certainly "feel" it - especially if your
recording involves a lot of microphones open at once, where "spill" from nearby
sound sources (like a bass guitar amplifier) may be common..
For this reason then, it is almost always a good idea to switch in the
high-pass filter that most modern mixing desks provide when recording from
microphone. The exception to this, is obviously when you are using a microphone
to record a very low-frequency sound such as a bass guitar, a cello, or other
low-frequency instrument, where keeping ultra-low frequencies is paramount.
Similarly - at the top end - any musical instrument which involves an
electrical "pickup" device - such as an electric guitar or electric piano - may
be susceptible to high-frequency radio interference.
When recording electric pianos in particular, I always filter off the very
top-end until I can hear it muffling the sound, and then I open it out again
until I hit the sweet spot where the basic sound remains unaffected by the
filtering. This is based on many sad years experience of recording an otherwise
good "take" that is suddenly spoilt by unexpected subtle interference from a
nearby source. It would be easy of course to "fix it in the mix" later, but
taking such an approach means that you have to suffer unsatisfactory monitor
mixes until main mixdown time. Best to get it right first time, by careful EQ.
Most high-pass and low-pass filters on modern mixing desks have quite a
precise response, and so an acceptable - if overcautious - general approach can
be to align the filters so that they "crop off" ALL sound which is theoretically
outside the frequency range of the instrument being recorded - in much the same
way as a professional photographer "crops" the edges of a photograph, leaving
only the areas of interest as the centre of attention.
Removing hiss from a multitrack recording should be a thing of the past.
Unfortunately, not all recording engineers are as bold as they could be when
setting recording levels, although in all fairness, on a live recording session,
this is often simply due to lack of time and a sensible regard for flexibility
and the need for a "safety margin". Also, it is not at all unusual these days to
be asked to do a "trendy" remix of an old, hissy, multitrack master from many
years ago.
The basic EQ trick under such circumstances is to remove as much noise as
possible from the "obvious" candidates such as bass drum, bass guitar, electric
piano, and poor-quality electric guitar pickups.
Although these instruments have a good deal of high-frequency energy that you
would not want to lose, in practice, most of this "energy" is well below 8 kHz -
a frequency above which noise becomes particularly offensive. Under these
circumstances then, it is perfectly acceptable to roll off as much top-end as
you can, using a steep low-pass filter. It's not "hi-fi", but it is a practical
approach that engineers have used for many years. It's part of the job of
getting good sound.
For the other instruments, you would be best advised to leave them well
alone. In most cases, EQ on the top end of the particular types of instrument
outlined above, combined with some judicious noise-gating on the remaining
tracks, should be more than enough.
If this is not the case, and you still feel that there is too much hiss
overall, then there are three "last-resort" approaches that you can take on the
remaining, troublesome tracks.
Firstly, you can try "single-ended" noise reduction, such as putting a Dolby
or DBX noise reduction unit, switched to "decode", across the offending tracks.
This is a popular trick amongst broadcasters and can be heard on many news
reports. The fact that it can be heard at all, should be enough to put you off
trying. The result usually sounds like the news reporter is fighting to avoid
being suffocated by a pillow. Even boosting the top-end EQ doesn't soften this
impression. It's pretty horrible.
A much more successful technique, is to EQ all of the hiss out of the sound
until it sounds positively muffled, and to then feed the resulting, dead, sound
through an Aphex Aural Exciter, or similar enhancement unit, which will
synthesise a new, clean top end out of the remaining lower-frequency musical
information. This technique works extremely well, and has even been successfully
used on the CD remastering of ancient classical works (although the engineers
involved would probably not admit to it publicly).
Lastly, if a particular track is so badly plagued with noise that it is
beyond the capabilities of the basic tricks outlined above, then all is not
lost. Although time-consuming, the offending track can be fed into a computer
and be subjected to noise reduction software. The noise-reduction results
possible now can be really quite breathtaking - although it must be pointed out
that if you overexert such systems, you can end up with the main sound source
sounding particularly artificial and computer-like (sometimes with interesting
results!). I frequently DO use computer based noise reduction - not on music
sources - but on the location news reports and sound clips from dubious sources
that need cleaning up for radio broadcast.
Harmonic Damping
doink!, doink!, doink!
Know that sound?
doink!, doink!, doink!
You've recorded a real drum kit haven't you?
The characteristic sound of a snare drum going "doink" is enough to make any
engineer feel queasy and go green. It is probably the most often encountered
problem when recording or mixing a real drum kit. The real solution of course,
is to make sure that you have a drum kit that doesn't go "doink!" - but it isn't
always that easy. It is possible to spend considerable time sticking "gaffer
tape" all over the surface of the drum heads, and yet still be left with
unwanted resonance's. Of course a snare drum in particular is an extremely
resonant instrument by definition. You can not only change the sound of the
snare drum by removing unwanted harmonics, but also - using a very
narrow bandwidth EQ - generate harmonics that aren't there in the first place
(because you are really hearing the sound of the equaliser itself
ringing).
Damping unwanted harmonics from all real sound sources by using EQ is
just a fact of life we have to put up with. Sure it's best to get it right at
source - perhaps by changing the instrument in some cases - but there are only
so many hours in the day. When time is restricted (and when isn't it
restricted when the studio costs $1500 a day? - or if you have only limited
spare time to do your recording in?), under these conditions, then quick,
practical solutions are they way forward. Save what precious time you have for
the things that make a real difference - like getting a good performance
from the artist when recording, or getting the delicate mix balance correct.
Another important aspect of sound recording that EQ "purists" fail to take
into account, is that when you record a drum kit, not only is every drum usually
"close miked" (i.e. recorded with the microphone only an inch or so from
the sound source), but those microphones are usually set to a "Cardioid"
response pattern - either in order to "concentrate" on the particular "drum"
being recorded, or simply because it isn't adjustable on that particular
microphone. And what happens with Cardioid microphones?
That's right! It's our old friend (or should that be "enemy"),
"Proximity effect". Every drum (or indeed any other instrument) recorded
close-miked using a cardioid pickup pattern will have an unnaturally high level
of "bass" in its sound. That's why - if you record and mix a real drum kit - you
almost certainly will end up rolling off quite a lot of the bass; not to
"create" a sound that wasn't there originally, but to correct the sound of the
microphone and therefore get the "real sound" back out of the
microphone.
Sound Enhancement
Most musical instrument sounds can be described as containing the following
(in order):
- Sub harmonics (low bass components)
- Fundamental note range
- Upper harmonics
- High harmonics
When using EQ for general "enhancement" of the sound, you are normally
staying well away from the fundamental note range, and are either boosting (or
cutting) frequencies in the other three frequency bands to create general
effects such as these:
- Boosting/cutting the sub harmonics can make a sound warmer or colder
- Boosting/cutting the upper harmonics can make a sound seem louder/softer
without changing the actual level
- Boosting/cutting the high harmonics can make a sound more/less "dazzling".
Surprisingly, this kind of activity rarely results in the instrument "leaping
out" of the mix in its high or low registers as you might expect. Most of the
"interesting" low and high harmonics which can be "excited" with careful EQ are
usually either much higher than the highest tonal "register" in which the
instrument plays, or not quite as low as its low harmonics are. If you do have a
problem, then a bit of judicious "counter-EQ" using a narrow bandwidth on the
edges of the fundamental note range can correct the problem without compromising
the effect overall.
Whatever you do with regard to "sound enhancement", remember that it is how
the sound is perceived as part of the final mix that matters. Don't spend too
long listening to the sound in isolation - the chances are that when you place
it into the mix along with everything else, a lot of what you've done may not be
audible and will need changing anyway - either more or less extreme - for the
overall effect you desire to be heard properly in the final mix.
I'll discuss "Sound enhancement" in more detail in the section "Equalisation
and Processing The Main Parts" coming up in a moment.
Bong, Boff and Sizzle
I'm so sorry - I couldn't find a better title for this section - which is a
great shame because it is actually critically important, so I'd
better explain myself pretty quickly.
I should point out that the previous section regarding "sound enhancement" by
EQ, is really discussing the subject from the point of view of an individual
instrument.
But the sound of the individual instruments alone doesn't make a
great mix.
It is the impact of the whole thing that matters.
Throughout this article, I've tried to emphasise the importance of listening
to each and every part of the song in terms of what it contributes to the mix as
a whole - and a great, energetic mix of a pop song - even a slow ballad -
usually has these following three important parts:
Bong
The "bong" in a mix is the pounding of the rhythm. Note that this does not
mean a bass-heavy mix. It just means that the impact of the relentless
rhythm must be clearly felt in your physical body as you listen to the mix. It
is not just the drums and bass that form the "bong", although they usually are
the primary contributors to it. It is also the pounding of the rhythm of the
other important parts such as piano and guitars. It isn't possible to describe
in words any special "formula" to make this happen, because it is different in
every case. You just need to be aware that it is there and figure out how to
control it (usually by experimentation).
Boff
The "boff" in the mix is usually the offbeat. It is created substantially by
the snare although, again, other instruments can contribute significantly to the
effect - look for syncopated beats that deserve accentuating. The alternate
"bong/boff" of a song is the defining essence of the songs rhythm, and if you
can get this across clearly, then people will get up onto the dance floor, or at
least start tapping their feet when the record is played.
The "bong" in a song should genuinely feel (at loud volume) like it
physically hits you in the midriff or below, whilst the "boff" should physically
hit you in the chest or higher.
Sizzle
Cymbals? Percussion? Yes, these things do contribute significantly to the
"sizzle" of a mix, but not entirely. There are many other contributors to the
very top end of a mix including the lead vocal - and the vocal reverb. A good
top end really shouts "Quality production!". George Michael ballad mixes (such
as "Praying for time") illustrate this kind of thing on vocal reverb (although
I'd be the first to admit that George Michael mixes are probably overdone
in this respect, which tires the ears somewhat). Guitars - especially steel
acoustics - can also create a wonderful "sheen" if the top end is carefully
managed, which can be heard over the top of a mix - even when played on a
low-quality transistor radio.
The careful management of the very top end of a mix is an art in itself, and
is something that deserves special consideration in the mix. But be careful.
It is a dangerous area. Why? Because if you overdo the top end of a mix, your
ears will grow tired very quickly. Not only can this give you a real, physical,
and extremely painful headache when mixing (it can be nasty and extreme, so take
a break immediately if this happens), but what's worse, it makes your ears grow
deaf to the top end, and so you end up actually piling yet more and
more top end on in order to compensate. You can do this without even
realising. This is a well-recognised problem when mixing, and is often the
result of mixing too loud for too long. It is not at all uncommon to listen to
what sounded like a great, powerful and loud mix the night before, and listen to
it the day afterwards and discover that it sounds like a little transistor
radio! A quick glance at the controls on the desk will reveal that all of the
top-end equaliser controls are cranked up way too high. Normally, a
less-than-perfect quick solution is needed to fix it under these circumstances:
Either putting the entire mix through a graphic equaliser, or by going along the
desk channels one at a time and reducing the amount of HF lift that you
inadvertently gave everything previously.
As an aside, it is worth mentioning that if you do get a severe
headache quickly and unexpectedly when mixing, and you are not otherwise ill,
then there is a strong chance that you have got something very wrong with the
top end of your mix, and it will give other people a headache when they listen
to it as well. I've heard this discussed by people many times in the past.
People aren't sure whether it is just the presence of too much top or whether it
is your ears trying to struggle at understanding the unnatural phase distortion
that occurs when overdoing top-end EQ - but people frequently acknowledge that
this is a genuine problem that some mixes have.
Also...
As a final thought - it's also worth mentioning that it is possible to
overdo the aspect of Bong, Boff and Sizzle in a mix. Make sure that the
impact of your mixing technique doesn't overwhelm the basic message and
emotional content of the song. Everything has to be in balance for it to work
properly.
Distance Placement
Here's a very cool mixing trick to remember (and not often
discussed either, which is a shame):
When it comes to distance placement (and the other two subtle effects in this
article - creating mix room, and auto mix levelling) we are now getting into the
realm of subtle equalisation that is rarely documented - but yet frequently used
- by sound engineers.
You may have tried to create positioning and distance in a mix previously
simply by the use of reverb and other room simulation effects. If you've tried
this then you've no doubt also discovered that this doesn't really work very
well, and your mix ends up a soggy mess.
This is partly because the effect of distance placement in EQ is also closely
related to a phenomenon known as "proximity effect" which most directional
microphones and the human ear, and even sound dispersion itself - all exhibit as
a matter of course. I'm using the term "proximity effect" here in a wider sense
than it is normally used. Let me explain further:
When a performer comes close - perhaps even too close - to a microphone (or
your ear!), then two things happen. Firstly, the amount of bass frequencies goes
sky-high (often called bass tip-up), because there is much less bass "loss" at
close range, and - more significantly - the physics of cardioid microphones
overemphasises this effect.
Secondly, the high-frequency content also goes up too - because distance of
sound in air tends to absorb high frequencies.
As a sound engineer, you can exploit this phenomenon in your mixes. By
unnaturally boosting both the high and low frequencies using the EQ controls in
your mixing console, you can create a sound that appears to be much closer than
it really is.
However, this proximity effect is overused in many recordings today, leading
to mixes that sound too "hi-fi".
Remember, it is also possible to do the complete opposite, and thin
instruments out a bit (shelve off some bass and a little off the top end), in
order to push them back into the mix and seem smaller and further away.
The real secret of mixing, is in the "light and shade" - contrasting
one part against another. By carefully using EQ in the form of subtle high and
low balancing - coupled with very short reverb - as a tool for instrument
placement, using the illusion of distance, you can make your mixes sound
significantly "larger" overall than if you try and make every part simply
sound "big" in its own right.
Big only seems "big" because other things seem small - it's easy to forget
this in the excitement of mixing.
Creating Mix Room
I'm not a big fan of this, even though I know that other people use it a lot.
This is really something that came out of the "Tamla Motown" mixing scene,
when people first realised that EQ was just as much a creative tool as a
technical one.
The idea behind "creating mix room" is that, when you can't hear an
instrument properly, then look for groups of other instruments that may be
combining together to obscure the instrument you can't hear. Then as a group,
you subtractively equalise those parts collectively in order to "carve
out some room" for the instrument you can't hear enough of.
You can either "carve out" this space by routing the other instruments
through an audio subgroup on the desk, and equalising that, or instead (and much
better, although it takes a bit longer), by applying the same subtractive EQ
using the individual equalisers patched across each instrument separately.
I'm quite set against this approach, as it seems philosophically extremely
unsound - and too much of a dodgy "quick fix" - for my liking..
It may have been appropriate in the days of Tamla Motown, when radios were of
poor quality, and people had to use every trick in the book in order to get
their mixes to sound good compared to others, but I think it is less valid now
we are in the digital age. The overall effect sounds artificial to me, and
probably to anyone else listening on the high-quality digital playback equipment
that is ubiquitous in this new millennium.
Notwithstanding the above, I do very occasionally use this technique as a
last resort, if the instruments in the mix appear to be conspiring to cover up
the lead vocal, and drastic action is deemed necessary.
Auto Mix Levelling
I love this technique! - but yet I've never seen it documented!!
Every mix engineer is familiar with the problem of individual parts of a mix
suddenly "leaping out" on certain notes and "taking over" a mix. Modern mixing
consoles provide automation to cope with this. But is this really the solution?
If you've had much experience of this at all, then you will recognise that it is
usually similar "sections" of the song, and similar note-ranges, that cause the
problem repeatedly.
Although you can resort to automated faders to solve this problem by
brute-force, (and it may take you hours to get the levels right), I seriously
urge you to experiment with letting the EQ do the work for you. Not only can it
save considerable time programming mix automation, but it can also give you much
better sonic results with less effort.
I'll explain more about the use of EQ for what I call "Auto Mix Levelling" in
the section called "Refining The Main Mix Levels" below. For the time being
though - lets return to mixing our track. That is, after all, what this article
is meant to be about, and we've deviated from it for too long...
Equalisation and Processing The Main Parts
As a summary of the above, and using the above-mentioned techniques, I'll
describe how you apply them to each track, along with other techniques such as
compression.
Firstly, for each instrument, see whether you need to do any "safety" EQ. By
safety EQ I mean rolling off either or both of the top and bottom end, using
"rolloff" filters in order to protect yourself against certain undesirable
things.
Shouldn't you do this only if necessary? Is it right to always punch
in the filters and restrict the sonic range when there might not be a
problem at all?
Perhaps - it depends how much time you have. Ah - that magic word "time"
again! Under tight time constraints it might take too long to go through every
track and listen out for specific faults. Generally speaking, time is always in
short supply in the recording studio, and in any case there is a lot to be said
for working quickly (to save your ears getting tired), and so it would not be
unusual to find a mix engineer who always switches in the filters as a matter of
course. I don't feel that it is "wrong" provided that it has no significant
effect on the desired portion of the sound.
After you've "trimmed" the frequency range with filters, and removed any
unwanted, troublesome harmonics, then here's the fun part: Focussing on just one
instrument, start playing with the EQ on that sound. Now before people jump all
over me, I need to make it absolutely clear that I'm not
for one moment claiming or recommending that you will actually use the EQ
on anything and everything at all!
Instead, what you are doing at this point, is not using the EQ as a tool to
change the sound (not yet anyway), instead you are temporarily using the EQ as a
tool to explore the sound - like a microscope. This is unfortunately much harder
to do using a "soft" (screen based) control interface, than it is when using
real knobs. If you are using a PC-based multitrack recording and editing
package, if it is possible to control the EQ using some kind of MIDI controller
with real knobs on it, you may find this quicker and more intuitive to use than
trying to do everything with the mouse alone.
With each sound, use the EQ to find the thing that "characterises" that
particular instrument. "Tune in" and "narrow down" what the interesting
harmonics are. Then you can decide whether you need to boost it, subdue it, or
perhaps even leave it alone completely. Don't EQ something for the sake
of it - make sure you understand why what you are doing works from an
audio perspective.
Don't "over characterise" the instrument, unless you intend to place it far
back in the mix and you find it becomes indistinct at that volume. Instead - for
parts that are playing independent musical parts - try to accentuate the "voice"
of each instrument such that it simply retains it's own place in the mix whilst
still being sufficiently big to fill it. Separate the parts out by using the
"distance placement" technique described earlier.
For parts that are playing the same part together, it is often
a good idea to not make them sound independent, but instead to make them
combine into one, single, "bigger" sound. Separating them can, in any case, be
difficult as the ear will be trying to merge them. There's little point fighting
against your ears and brain and trying to convince them that the two parts are
distinct. From a musical perspective they are not, and they have usually been
actively designed that way, so that they can be combined into one new,
bigger sound.
Doing this requires a little effort, but can yield eerily effective results.
Think of George Bensons distinctive "scat" vocals that accompany some of his
guitar solo parts, and how the vocal and guitar blend together in order to form
a bizarre new guitar sound. Similarly, mixing synth sounds with (e.g.) live
strings can be much more effective and bold-sounding if you try and form them
together into a new type of string sound, rather than leaving it sounding like a
silly little synth playing over a massive orchestra.
It's hard to discuss in mere text, what you do when working with each of the
main sounds, and I was going to add some brief observations on the kind of thing
you can expect to find when working with some of the more common "main parts",
but it is impossible to give each instrument adequate coverage (I tried and got
exhausted after just two inadequate examples), and this article is already long
enough anyway. More importantly it would be missing the point of this article to
talk about specific lead instruments when it is general
mixing principles that I am trying to explain here.
I'd summarise the EQ process - regardless of what instruments you are dealing
with, as using the EQ initially as a kind of "microscope" that lets you examine
the sound in great detail for a minute or two. Once you are familiar with the
bits that constitute that particular sound, you can then decide how you
want to deal with them - and that might not necessarily involve EQ.
For example, by playing with the EQ as a listening tool you
might discover that an instrument (perhaps a guitar) potentially has a lot of
"attack" to it, that isn't yet being realised. How should you bring out that
"attack"? Perhaps you might decide to use the EQ - but that can often
turn the sound - particularly guitars - into a thin sound, without much body or
depth. You might instead decide to use some compression with a slow attack in
order to accentuate the attack on the instrument.
Track Sharing
Finally, if certain tracks have more than one instrument part on them
("track-sharing" is a common practice if the number of tracks you have is
limited), then it is extremely likely that the different parts will need
different levels, EQ, and effects on them (although if you've read my article on
setting recording levels, you will know that I am an advocate of recording
multiple things on the same track at their correct relative levels, so that
rough "monitor mixing" during production is much easier).
In order to sort out different parts on the same track, you have two options.
You can either (a) automate your mixing desk so that the correct settings "kick
in" at the right part of the song, or you can (b) duplicate the track and use
different settings with the same track coming through two different channels. On
a conventional analog system, option (b) is easy - you just use a patch-cord to
plug the track into two different channels at once. On some high-end PC software
systems you can do this too (in software), but not always. In some PC systems
you might end up having to duplicate the track in order to get it to come up two
different mixing channels.
Option (b) is by far the easiest in terms of the amount of work you need to
do. Although "automating" the same channel so that it suddenly changes settings
at the relevant part of the song seems "clever" and "neat" and is therefore
appealing, it is also a lot more time consuming and can often be quiet
difficult. By putting the same sound through two different channels you can play
around manually, without automation, to your hearts content which is much
easier, and the only automation that you need to worry about is the automated
"mute" that switches from one channel to the other at the relevant parts of the
song.
On an entirely PC based system however, you may find that option (a) is
better, as it might make more efficient use of your CPU's resources, because
each extra track or channel tends to use up more CPU power. One workaround for
this CPU power consumption problem might be to "render" the settings for the two
parts and mix them down to one track, saving the originals in case you need them
in future.
Summary of Adding Main Parts
The key thing when adding the main parts is to not take a
"prescription" approach that blindly follows any "rules" you think I've outlined
above. You should make sure that you really understand what each of the
parts are "saying" and how they interact with each other. That will help you
decide both the sound and the stereo positioning. Use your ears, and look
at the controls only if you think you've done something wrong or if you
want to remember the settings for a future session, and remember -
there are no rules - it is really only how the thing sounds in the end that
matters after all this - not the theory of how you did it - and above all,
please remember that the above comments are just intended as helpful hints and
suggestions, and feel free to disagree and go against them as you see fit.
Perhaps you might have totally different approaches in mind, so feel free to
experiment as much as you like. Remember though, as time goes on your
ears will get more and more tired and you will be less able to make
sensible decisions, so work as quickly as you can, and don't spend too long on
any one instrument - it will drive you to the point of mental breakdown if you
do.
Also, don't be afraid to use quite extreme compression on some of the lead
parts if you genuinely believe it sounds right to do so. I've often been amazed
at how much compression some parts seem to require, but yet in the
overall sound of the mix, heavy compression if often not particularly noticeable
(unlike final "mix compression" which is very audible if overdone). Make
sure that the compressor isn't permanently compressing though - otherwise you're
not getting the best out of it. On the quiet sections of a performance there
should be little or no "Gain reduction" showing (it's obviously very helpful if
the compressor has a "gain reduction" meter). If the "gain reduction" lights are
always on, then you have almost certainly got the "Threshold" control set
way too low - unless you are deliberately using the compressor to add "punch" in
which case its excusable. Otherwise, a compressor with the threshold set too low
is starting to act more and more like a simple volume control and is a waste of
time.
Adding Percussion
You don't have to add percussion after the main parts, sometimes it
makes sense to do so before - when you've got drums, bass, and any pad parts in
place. It depends on the song. Personally I often like to do percussion later on
in the mix because you can get a better perspective of what the percussion is
really adding to the mix. It also gives you a break after doing the drums and
bass which have probably already given your ears quite a pounding.
The guidelines are similar to the lead parts; listen to what each part is
"saying" and that will help you get your stereo placing. Some things will be
intended to be almost part of the drum kit (such as cabasas, tambourines, and
maracas, which often work in conjunction with the hi-hat). Other things are
quite separate (like timbales) and deserve to be featured for only very short
stretches at a time before they become boring.
You don't have to use all the percussion when mixing - in fact it is
generally best not to do so.
The reason for this, is that when recording percussion, people tend to be
overgenerous. They put lots and lots in "just in case" on the grounds that "it
can always be taken out in the mix later". This isn't unreasonable, so bear it
in mind, and consider using automated mutes on the mixing system to just bring
in the percussion at particular sections that need a little more "colour" adding
to them.
When EQing percussion, remember that if you want to get more top end,
removing the low and mid will give you a more smoother sounding top end than
simply cranking the high-frequency EQ up. It isn't that one is necessarily
better than the other - they just produce different results.
For example, things that go throughout the entire song, like perhaps congas,
cabasas, maracas and the like, normally respond to subtractive EQ
(removing low and mid, rather than just boosting high) for a smooth sound -
otherwise they tire and strain the ears. But things which are featured only
briefly - such as timbales - benefit from the extra "thwack" that high-boost
alone provides. Additionally, in the special case of something reinforcing an
important drum beat - such as tambourine beating in time with the snare - such a
sound can benefit from the extra energy that pure high-frequency boost gives,
and make it stand out against the drum it is competing with.
Generally speaking, for most percussion it is not at all unusual to have to
remove a fair bit of low-end to get the percussion to "cut through" the mix.
Bongos and congas - no matter how well recorded - usually need to be "thinned
out", in order to be properly heard on a busy pop mix.
Our old friend - the "small" reverb - is of particular importance when mixing
percussion. To get a truly spectacular effect, try being generous with the small
reverb, and try making the percussion sound like it is "outside" of the bounds
of the rest of the mix - so that it sounds, further back, yet bigger, and
"surrounding" the rest of the mix instead of being in the middle of it (unless
of course, you are trying to get a 70's disco sound, in which case leave the
percussion fairly dry). If you do this "wide spacing" using short reverb, it is
often most effective if it is used only on some parts of the mix (like a
percussion break) - if it is like this for the duration of the track it can be
tiring on the ears, and distracting to the rest of the mix.
Balancing the levels of the percussion is a tricky business, and best done at
fairly quiet levels on small monitor loudspeakers, otherwise there is a risk
that one thing (like a handclap or tambourine) will dominate the final mix.
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