The ‘art’ of EQ
The “art” of EQ
by Aaron Trumm
EQ can be used in a variety of situations, from live sound to recording to tape
to mixing down. Mainly, it should be used to enhance signals that have some
problem. The golden rule of EQ is less is more. If something seems fine without
it, I avoid EQing it at all. Then, if I do use it, I try to remain subtle. My
personal golden rule is nearly never EQ signals going to tape (as in a
multitracking situation). I always try to get the original sound on tape, then
I can mess with it later. Putting EQ (or any other effect) on tape usually just
leads to trouble. The other rule (the silver rule 🙂 ) is cutting is almost
always better than boosting, especially when fixing problems. For example if a
guitar sounds too thin, first try cutting high frequencies and boosting the
gain a bit, instead of boosting the lows. The more clutter you can remove from
a mix, the better. A better example is I very often cut a bit of high away from
hats. Another example is, many times you may not hear something well in a
mix…You might try cutting some frequencies in a different track that seems to
be interfering, rather than boosting in the track you want to bring out. With
these basic rules in mind, I’ll tell you my rules when I enter a mixdown session:
3. Rule Of Opposites: Usually, tracks with high sounds, (a
high guitar, hats) need cutting in high frequencies and boosting in lower, and
vice-versa. This is really only a starting guide, not a rule. Also, sounds that
interfere with eachother can be separated in a mix by EQing them in opposite
4. Bass usually needs a boost in the mid range somewhere and
sometimes the high. This way it can cut through and be heard on smaller
5. Kick drums usually need that same mid and/or high boost on
a subtle level so they too can cut through on smaller speakers. For hip-hop,
kick needs a low end boost, but NOT TOO MUCH.
6. Snare drums always sound warmer with a boost in the low-mid
range and some cut of the highs. An annoying CRACK can be softened with this
high cut. Sometimes I boost the lows in snares to make them even fatter. But it
really depends on the snare sound. The rule of opposites usually applies here.
Snare sounds that were thin to begin with I usually warm up a bit, and heafty
snare sounds I might thin out a bit.
7. Hats almost never need any EQ if they’re recorded clean.
Usually an EQing for my hat tracks is to cut highs to get rid of an annoying
8. Guitars are simaler to snares for me. A thin original
guitar might need boosting in mids and lows (depending on what the desired
sound is, and what else is present in the mix) or a heafty guitar might need to
be thinned out a little by cutting lows and low-mids.
9. Vocals usually like to have a boost in the mids or
high-mids, but it depends on the voice. Vocals nearly always get lost amongst
guitars…a good way to deal with this is the rule of opposites. Boost mids in
the vocals and cut them in the guitar, or something similar. Vocals can also
have annoying hiss or sibilance, and sometimes cutting high frequencies can
10. Strings, and more specifically good string patches from a
synth, usually need little EQ. If they are merely a support player, I may thin
them out a tiny bit, or if they are meant to be present, I may thicken them in
the mids a little (or sometimes the opposite…this stuff is highly
subjective). But they usually work well left alone. Really clean piano or
keyboard synth patches are the same way.
11. I like to leave reverb returns alone, but if the reverb
becomes annoying and noisy, cutting some high can soften it up a bit…same
12. Extreme EQ setting create sounds of their own. Experiment.
But for a non-novel track, be subtle.
13. AC hum from a track can almost always be fixed by cutting 60
Hz all the way off. (Sometimes this can take away from bass or kick sounds, but
I believe that most frequencies audible in a song are above 60 Hz).
14. Play with EQ settings thoroughly to find appropriate
15. I don’t mix horns too often, but when I do, I like to leave
them alone. Clean horn tracks usually seem fine to me.
16. NEVER EVER EVER force yourself to EQ a track that sounds
fine, just because you think you should use the full capabilities of the
studio. NEVER NEVER NEVER!
If anyone out there has rules
they use for their mixes, especially for instruments I don’t mention or use
much, send ’em along. 🙂
If you have questions, or have
noticed I have left something out, or misspelled, or mis-explained, or (god
forbid! hehe) I’m wrong, mail me at ő£ŌÜő¨őĽőľőĪ!
ŌÉőĶőĽ 20320m1215u ;őĻőīőŅőīőĶőĮőļŌĄő∑Ōā.or
Manny Rettinger at ő£ŌÜő¨őĽőľőĪ!
ŌÉőĶőĽ 20320m1215u ;őĻőīőŅőīőĶőĮőļŌĄő∑Ōā..
ŌÉőĶőĽ 20320m1215u ;őĻőīőŅőīőĶőĮőļŌĄő∑Ōā.
A Basic Guide for EQing
by Devin Devore of ő£ŌÜő¨őĽőľőĪ!
ŌÉőĶőĽ 20320m1215u ;őĻőīőŅőīőĶőĮőļŌĄő∑Ōā.
Dating as far back as the 1930’s, the equaliser is
the oldest and probably the most extensively used signal processing device
availible to the recording or sound reinforcement engineer. Today there are
many types of equilisers availible, and these vary greatly in sophistication,
from the simple bass and treble tone control of the fifties to advanced
equipment like the modern multi-band graphic equaliser and the more complex
parametric types. Basically, an equaliser consists of a number of electronic
filters which allow frequency response of a sound system or signal chain to be
altered. Over the past half century, equalisers design has grown increasingly
sophisticated. Designs began with the basic ‘shelving filter’, but have since
evolved to meet the requirements of today’s audio industry.
Understanding EQ and its Effects on Signals
There are two areas of equalisation that I want to
cover. Those two areas are vocals and music. I’d like to discuss the different
effects of frequencies within audio signals. What do certain frequencies do for
sound and how we understand those sounds. Why are some sound harsh? Why do
things sound muddy? Why can’t I understand the vocals? I’ll try and answer all
of these question and hopefully bring some light to the voo-doo world of EQ.
Roughly speaking, the speech spectrum may be
divided into three main frequency bands corresponding to the speech components
known as fundamentals, vowels, and consonants.
Speech fundamentals occur over a fairly limited
range between about 125Hz and 250Hz. The fundamental region is important in
that it allows us to tell who is speaking, and its clear transmission is
therefore essential as far as voice quality is concerned.
Vowels essentially contain the maximum energy and
power of the voice, occurring over the range of 350Hz to 2000Hz. Consonants
occuring over the range of 1500Hz to 4000Hz contain little energy but are
essential to intelligibility.
For example, the frequency range from 63 to 500Hz
carries 60% of the power of the voice and yet contributes only 5% to the
intelligibility. The 500Hz to 1KHz region produces 35% of the intelligibility,
while the range from 1 to 8KHz produces just 5% of the power but 60% of the
By rolling off the low frequencies and accentuating
the range from 1 to 5KHz, the intelligibility and clarity can be improved.
Here are some of the effect EQ can have in regards
to intelligibilty. Boosting the low frequencies from 100 to 250Hz makes a vocal
boomy or chesty. A cut in the 150 to 500Hz area will make it boxy, hollow, or
tubelike. Dips around 500 to 1Khz produce hardness, while peaks about 1 and
3Khz produce a hard metallic nasal quality. Dips around 2 to 5KHz reduce
intelligibilty and make vocals woolly and lifeless. Peaks in the 4 to 10KHz
produce sibilance and a gritty quality.
Effects of Equalisation on Vocals
For the best control over any audio signal, fully
parametric EQ’s are the best way to go.
160 to 250
315 to 500
Sense of power in some outstanding bass singers.
Important to voice quality
630 to 1K
Important for a natural sound. Too much boost in
315 to 1K range produces a
honky, telephone-like quality.
1.25 to 4K
5 to 8K
Accentuation of vocals
Important to vocal intelligibility. Too much
boost between 2 and 4KHz
can mask certain vocal sounds such as ‘m’, ‘b’,
‘v’. Too much boost between
1 and 4KHz can produce ‘listening fatigue’.
Vocals can be highlighted at the 3KHz
area and at the same time dipping the instruments
at the same frequency.
Accentuation of vocals.
The range from 1.25 to 8K governs the clarity of
Too much in this area can cause sibilance.
Miking instruments is an art … and equalisers can
often times be used to help an engineer get the sound he is looking for. Many
instruments have complex sounds with radiating patterns that make it almost
impossible to capture when close miking. An equaliser can compensate for these
imbalances by accenting some frequencies and rolling off others. The goal is to
capture the sounds as natural as possible and use equalisers to strighten out
any non-linear qualities to the tones.
Clarity of many instruments can be improved by
boosting their harmonics. In fact, the ear in many cases actually fills in
hard-to-hear fundamental notes of sounds, provided the harmonics are clear.
Drums are one instrument that can be effectively lifted and cleaned up simply
by rolling off the bass giving way to more harmonic tones.
Here are a few ideas on what different frequencies
do to sounds and their effects on our ears.
These frequencies give music a sense of power. If
over emphasised they can make things muddy and dull. Will also cloudy up some
80Hz to 125Hz
Too much in this area produces excessive ‘boom’.
160Hz to 250Hz
This is the problem area of a lot of mixes. To
much of this area can take away from the power of a mix but is still needed
for warmth. 160Hz is a pet-peeve frequency of mine. Also, the fundemental of
bass guitar and other bass instruments sit here.
300Hz to 500Hz
Fundamentals of string and percussion
400Hz to 1K
Fundamentals and harmonics of strings, keyboards
and percussion. This is probably the most important area when trying to
control or shape to a natural sound. The ‘voice’ of an instrument is in the
To much in this area can make
instruments sound horn-like.
800Hz to 4K
This is a good range to accentuate instruments or
warm them up. Too much in this area can produce ‘listening fatigue’. Boosts
in the 1K to 2K range can make instruments sound tinny.
4K to 10K
Accentuation of percussion, cymbals, and snare
Playing with 5K makes the
overall sound more distant or transparent.
8K to 20K
This area is often what defines the quality of a
recording or mix. This area can also help define depth and ‘air’ to mix. Too
much can take away from the natural sense of a mix by becoming shrill and
Here are a few other pin point frequencies to start
with for different instruments. In a live sound situation, I might event pre
set the console’s eq to these frequencies to help save time once the sound
check is under way. These aren’t the answers to everything… just a place to
Besides the usual cuts in the 200Hz to 400 area,
some tighter Q cuts at 160Hz, 800Hz and 1.3k may help. The point of these cuts
makes for space for the fundamental tones of a bass guitar or stand up. I have
also found a high pass filter at 50Hz will help tighten up the kick along with
giving your compressor a signal it can deal with musically. 5K to 7K for snap.
The snare drum is an instrument that can really be
clouded by having too much low end. Frequencies under about 150Hz are really
un-usable for modern mixing styles. I would suggest a high pass filter in this
case. Most snares are out front enough so a few cuts might be all that is
needed. I like to start with 400Hz, 800Hz, and some 1.3K. This are just
frequencies to play with. Doesn’t mean you will use all. If the snare is too
transparent in the mix but I like the level it is at, a cut at 5K can give it a
little more distance and that might mean a little boost at 10K to brighten it
High hats have very little low end information. I
high pass at 200Hz can clean up a lot of un-usable mud in regards to mic bleed.
The mid tones are the most important to a high hat. This will mean the 400Hz to
1K area but I’ve found the 600Hz to 800Hz area to be the most effective. To
brighten up high hats, a shelving filter at 12.5K does nicely.
Again, the focus here is control. Most toms could
use a cut in the 300Hz to 800Hz area. And there is nothing real usable under
100Hz for a tom… unless you are going for a special effect. Too much low end
cloud up harmonics and the natural tones of the instrument. Think color not big
In my opinion, drum over heads are the most
important mics on a drum kit. They are the ones that really define the sound of
the drums. That also give the kit some ambience and space. These mics usually
need a cut in the 400Hz area and can use a good rolling off at about 150Hz.
Again, they are not used for power…. these mics ‘are’ the color of your drum
sound. Roll off anything that will mask harmonic content or make your drums
sound dull. Cuts at 800Hz can bring more focus to these mics and a little boost
of a shelving filter at 12.5K can bring some air to the tones as well.
Bass guitar puts out all the frequencies that you
really don’t want on every other instrument. The clearity of bass is defined a
lot at 800Hz. Too much low end can mask the clearity of a bass line. I’ve heard
other say that the best way to shape the bass tone is to roll off everything
below 150Hz, mold the mids into the tone you are looking for, then slowly roll
the low end back in until the power and body is there you are looking for. If
the bass isn’t defined enough, there is probably too much low end and not
enough mid range clearity. Think of sounds in a linear fashion, like on a graph.
If there is too much bass and no clearity, you would see a bump in the low end
masking the top end. The use of EQ can fix those abnormalities.
These instruments all have fundamentals in the mid
range. Rolling off low end that is not needed or usable is a good idea. Even if
you feel you can’t really hear the low end, it still is doing something to the
mix. Low end on these instruments give what I call support. The tone is in the
mids. 400Hz and 800Hz are usually a point of interest as are the upper mids or
1K to 5K. Anything above that just adds brightness. Remember to look at
perspective though. Is a kick brighter than a vocal? Is a piano bright than a
vocal? Is a cymbal brighter than a vocal?
Equalisers are one of the most over looked and
mis-used pieces of gear in the audio industry. By understanding equalisers
better, an engineer can control and get the results he or she is looking for.
The key to EQ’ing is knowing how to get the results you are looking for. Also,
knowing if its a mic character or mic placement problem. EQ can’t fix
everything. It can only change what signal its working with. Equalisers are
also a lot more effective taking away things in the signal than replacing what
was never there.
Reverb is an important studio tool. It can be used
to add realistic ambience to a sound that was recorded in a dead, dry room, or
to electronic or synth sounds. About everything we hear has some reverb to it,
so when we hear an untreated sound, it sounds uncomfortable, and unnatural.
Back in the sixties and seventies before there was
digital reverb, studios used plate reverb. They would hang a thin piece of
metal inside some frame work, and vibrate it using a voice-coil assembly. Then
they would mic the metal plate with contact mics and feed that back into the
mixer. The only problem with this method was that it sounded metallic and
bright. After so many years of hearing this, people were used to it, and the
new digital reverbs sounded strange to them. Now, digital reverb units repeat
little fragments of the sound wave thousands of times to recreate
reverberation. Most reverb units have hall sounds, room sounds, and, of course,
plate sounds which are great for drums.
Rules for Using Reverb
sounds the best when used in sparingly. Don’t swamp tracks in it. Use the least
possible to get the desired effect. The best engineers know when they have used
Sounds with a
lot of bass, such as the kick drum or bass guitar are best left with little or
no reverb. If you do use it, keep it short and bright, or cut the low
frequencies on the reverb return. Otherwise you’ll have a big mess before you
more reverb you use, the farther away a sound will seem. This can be used to
push certain things back in the mix such as backing vocals, but once again,
don’t load it on.
your effects unit will allow you to use many different types of reverb in one
mix. Theres nothing wrong with using a couple of different reverb styles all
within the same mix, it will just sound more interesting to the ear.
Bright Plates, nonlinear
Between 1.1 and 2.5 seconds
Around 20 milliseconds
Plate or short hall
Between 2 and 3 seconds
Between 20 and 60 milliseconds
Hall or concert hall
Between 2 and 4 seconds
Between 5 and 50 milliseconds
Room or Plate
Between 1.5 and 3.5 seconds
Between 20 and 50 milliseconds
Plate or Bright hall
Between 1 and 2.5 seconds
Between 20 and 80 milliseconds
10 Steps to a better Mix.
By: Howard Mangrum
The following is a ten step procedure for the
mixing of a song. These steps can be varied in any way necessary to accommodate
the themes or concepts of the song or materials to be mixed. Please be aware
that the detail of each step can change depending on your equipment and the
song. Sometime the song may not be fully developed and attempting to mixdown
will make this evident, one of the reasons why even the professionals do rough
mixes. Final mixes are best approached when your ears are fresh, not at the end
of an all day tracking session. After mixing various projects you will develop
your own procedure and you can feel free to throw this out, I mean store this
for further reference along with all of the other bad song ideas that your
friends have come up with.
Normalize & Mute
Normalize each track by panning to the center, take
the EQ section out or verify all settings are zeroed (this may be in the 12
o’clock position), and turning down (off) all Aux Sends so that there are no
effects. Pull all faders down (some people mute each channel, then un-mute them
individually as they proceed).
(This is a good place to play a reference CD, to
help ensure the monitoring system is performing properly and you have a good
referenced starting point for your mix.)
Review any notes taken during the tracking process
and your preproduction notes. Setup the signal routing scheme, configure patch
bays. Compressors and noise gate can be patched in and normalized so they have
little or no effect (put device into bypass if possible, set noise gates to a
low threshold, etc.). It should be possible to assign outboard effects
(reverbs) to the various tracks at this point based on the song concept and
basic ideas of the sonic landscape.
(It is perfectly acceptable to determine the
concepts and sonic landscape as you progress through these mixing steps, this
is art and there are no rules, just guidelines or opinion.)
Set the tape deck to play the song in loop-mode if
possible. This allows the following steps to be completed in a continuous
Critique & EQ
Critique each track individually. Start by soloing
(un-muting or only bringing up one fader) each track to ensure proper gain
setting by observing your level indicators. Setup noise gates and compressors
if necessary. Perform your first rough EQ, do this EQ as fast as possible,
don’t spend more than a few minutes per track
(the point of dimensioning returns is close at hand
during this first pass, the perceived frequency distribution will shift/change,
as all tracks are mix together)
(Periodically switch the EQ section out and back in
to help ensure you are making improvements.)
(General approach to EQ; if you have a parametric
with tunable ‘Q’ use it to fix frequency problems, get rid of the bad sound.
Use shelving EQ’s to do gross adjustments to the sound, since they effect a
large range of frequencies.)
(It is usually better to cut, so work to cut the
bad and if you have remaining control use this to enhance.)
(Adjustment to the EQ/frequency content can make
dramatic changes to the gain structure of your signals/sounds, be sure to keep
an eye on this and make adjustment accordingly)
Bring up each track to start building your mix. The
order should mimic the priority of each track, this depends on the style of the
music and your personal tastes.
(It is standard practice to start with the
foundation, such things as drums & bass)
(This is were you start to build your sonic
landscape, you determine which sound should be out front and which sound should
be in the background.)
(If necessary this is a good place to draw a sketch
of your stage setup of the band, to help visualize your sonic landscape)
Re-EQ tracks where necessary. Listen for too much
sound (muddy) in each frequency range, where, you have instruments or sounds
that are in the same basic frequency range and may conflict or mask each other.
Pan tracks/sounds to complete the setting of your
(This step is done in conjunction with re-EQ, step
5. The overlapping frequencies maybe less offending after panning)
(Periodically monitor your constructed sonic
landscape in mono to ensure that phase cancellation and sound masking are not
going to cause you any problems, your mix should stand-up in mono as well as
stereo, with only the basic imaging shifting.)
Setup the reverb and other effects. When applying
reverb, keep your sonic landscape in mind. You are setting the outer-boundaries
of your sonic landscape at this point.
(It is easy to over use reverb and other effects,
generally turn-up the effect to a point where they become dominate then back
them off to they just meld into the background)
(Be sure to keep a written record of which effects
are used where and the programs of the effect units, with any special settings
and/or signal routings that have been employed)
Listen to the mix and ensure you can hear each
sound and the over-all balance between each sound is correctly portioned.
(This is a good point to perform a rough mix to
tape and playing it on a secondary monitoring system to help gain a second
prospective and ensure the main monitoring system is not leading you down the
wrong path. Studio monitors can reduce the perceived impact of various settings
and the amount of such things as reverb.)
Map out any move that maybe necessary, such as:
(map the move to a tape counter and/or a smpte time
readout, keep a written list of these moves)
(learn to perform the moves on-the-beat, tap your
foot and count)
Practice the mix, learn to play the console/mix
like an instrument. When you are confident with your mix start recording it to
your mixdown deck.
(It is usually good to perform a few mixes, like
any performance each will be different and one will usually be preferable.)
(It is common at this point for you to realize that
you have not determined how the song should start or end, map these moves out
as above. Be sure to allow for some pre-roll & post roll time)
(Performing a mix should be similar to performing
on an instrument where moves and other events happen on the beats of the song,
your mixing moves should have rhythm to them.)
Roger Nichols Recording Guide
people (musicians) will be in the recording room and how will they be arranged
Instruments will they be playing and what special requirements need to be met
How big is the
room (or rooms). If sharing an isolation room, consider grouping of instruments
for least adverse leakage.
between instruments should be considered. Is some of what is being recorded
going to be replaced (stand in vocalist, not the real solo, etc.) Determine how
best to isolate the instruments (baffles, Tube Traps, blankets, foam, plywood).
You can never
have too many cables or adapters. All cables must have previously been
ascertained to be in proper working order. Cables that have been previously
suspect and checked to find nothing wrong should be labeled as such until they
have successfully worked in a session. (this is in case of a cable problem, the
first cable to check would be a previously faulty cable.) Anticipate problems
as much as possible.
microphones. What mics are available for the session? What mics are
specifically requested by the client? Are there notes from previous sessions
with the same musicians that pointed out a unique requirement or a mic that
worked rather well in a particular situation.
be matched. (Lo, Hi, Inline Xformer) Thin sounding microphones (reduced low
frequency response) usually means that the impedance is not matched properly.
Connections must be matched. (XLR, 1/4″, DIN, Teuschel). Polarity must be
matched (Pin 2 hot – Pin 3 hot?)
requirements must be ascertained. If mics are split to multiple consoles, such
as live performances, only one console should provide phantom power. Make sure
that mic splitter will pass phantom (some will not). If console does not
provide the proper phantom voltage, use external pass through phantom power
modules. If the microphone has it’s own power supply, make sure that phantom is
turned off to that mic, otherwise distortion and noise may result. (Guaranteed
in some instances)
phantom on with unbalanced microphone.
Try direct box
for synths and electric instruments. Try different direct boxes (they are like
microphones and have a coloration of their own). Active (phantom powered)
direct boxes may not have ground isolation capabilities and may cause ground
loops (which result in a buzz or hum). Some consoles will let you
pluginstruments in directly. (check for impedance matching.)
acoustic instrument can be added in with the microphone sound.
must be chosen properly for the job. Understand proximity effect in
placement may not be the same each time you record in a similar situation. It
may depend on the individual player or instrument.
reflections off of music stands (use foam or towel) when recording vocals.
Listen for extraneous noises from squeeky chairs or rattling instruments.
Speakers and monitoring:
What do they
sound like. Have you heard these speakers before in a different environment?
Does the control room color the sound of the speakers so that you must
compensate for that difference?
speakers in control room may effect the way they sound (experiment with
Try not to use
speakers (instead of headphones) for monitoring in the studio during recording.
The leakage will hamper an otherwise good recording. If this must be done,
there are methods whereby two speakers are fed a MONO signal and placed out of
phase. The microphone is then placed in the phase null between the speakers.
Extreme caution should be taken when employing this method.
distribution system for multiple headphones, don’t just parallel a bunch of
headphones from the console or cassette machine headphone outputs.
Me” headphone distribution systems for individual mixes to each musician
can help the recording process immensly.
volume control. Don’t clip the input. In some situations it may be desirable to
set all faders to “zero” and establish the initial recording level
with the input trims. This provides an instant graphical representation if
microphone levels change drastically during the recording (the fader is pulled
down or up from it’s reference)
sound through the console, select the correctrouting path, with inserts
With no EQ,
listen. Does it sound good, is it muffled, scratchy, far away,or boomy?
Is the mic
facing the wrong way, (this happens often) or are you listening to wrong input.
mismatch between the console input and the source. (line, mic, instrument)
connection, patch bay, patched in wrong hole, patchbay normal not broken- mic
going too many places at once
unbalanced – pin2 vs. pin 3 (unbal pin 3 at one end + unbal pin 2 = SHORT)
– change try another
technique or position – try something else, face Mecca.
Move the mic a
little – start with close micing, then move the mic away
guitars, pianos, Bass, Standup bass, Drums
Go in the room
and listen to the instrument with a finger in one ear.
Ask the player
– chances are he has recorded this instrument before and has some idea as a
If he says
“This is what I do all the time and it always sounded good before”
then there is probably something else wrong.
input. Monitor through the machine (good idea in case you are overloading the
machine input) make sure that whole signal path is working right. (what you see
on the meter may not be what you think is going there.
output of machine with no music playing. Listen for hums, crackles or buzzes.
If the meter is reading something, then there is probably a hum or other noise
that you didn’t notice.
What kind of
metering? Digital metering is the most accurate. Peak meters second best Analog
VU meters, depend on what music is playing (click, hi hat, organ, etc.)
Percussive instruments should indicate lower on the VU meter for proper
to make sure that you are using the correct tape for the machine. Bias, tape
stiffness & head wrap.
reduction dbX Dolby A, B, C, S, SR
noise reduction on the SMPTE track. Make the SMPTE track one of the edge
tracks. (cross-talk). Don’t use noise reduction on digital recordings.
a nice feature. Also autopunch, cycling etc.
to clean the machine. Digital machines need cleaning too. Follow manufacturers
machine in plenty of time before the song begins. Allows machine to get up to
speed. Allows plenty of SMPTE for future lockups.
machine keep running a little while after the take. In case you want to add
something at the end, or cross-fade into the next tune, or?
Make sure you
have enough tape for the take you are about to record. If what you are
recording is longer than a reel of tape, plan a break in the music for changing
tape or get a second machine for A/B rolling. (the second machine is placed
into record before the first machine runs out of tape.
Must be able
to monitor output of machine
Try not to use
speakers to monitor during overdubs
Test punch in
capabilities of machine Punch during sustained playing & punch right on
beat. Play back. See if there is glitch and see if there is any delay. If
delay, modify punch technique accordingly.
If big glitch,
don’t punch during sustains or punch on back beat or someplace that will mask
Effects – EQ:
change the tonal characteristics of the audio. They have at least bass and
treble controls. Most desirable is four band sweepable parametric EQ.
equalizer. Usually 5 to 31 frequency bands, each fixed in frequency. Usually
with slide pots to show a graphic representation of the frequency curve.
Tuning EQ by
Use EQ to:
low listening levels
Make the blend
between different instruments more pleasing
bad frequency response in some device
effects like telephone voice
apparent leakage between instruments
Effects – Compressors & Limiters:
keep levels more constant by automatically detecting level changes above a set
level and riding the gain.
compressors on individual instruments, not mix. It will be less audible.
settings determine the “punchiness” of the instrument. Peaks get
through before the compressor actually clamps down. Faster attack will make for
a smoother sound.
faster than compressors and are there to LIMIT the amount of signal passing.
These are usually there to protect equipment such as radio transmitters or
speakers from overloading.
Effects – Noise Gate:
work like a soft on-off switch. As the level of the sound gets below a set
point, the signal is turned off, blocking any residual noise that may creep
through. If not set properly they can be worse than the noise.
Effects – Delays & Echoes:
A delay by
itself has no effect but to delay the signal. When the delay is heard mixed
with the original signal, we have a sometimes more interesting sound.
reverb units control the amount of feedback sent to the input of the delay as
well as the number of taps off of the delay line. These signals mix together to
form artificial reverberation as found in different size enclosed spaces.
(recording the same instrument playing the same part twice) can be simulated by
using a delay of from 9 to 30 milliseconds. This fattens up vocals and
instruments and can make it sound like there was more than one instrument
playing the same part.
can also add fake ambiance to a recording that was too dead sounding.
caused by modulating the delay time. This modulation causes a change in pitch
as well as a change in the delay time. This produces a wavy effect in the
effects are created by using a delay of 10 to 20 milliseconds and changing the
delay amount slowly between those two parameters. The delayed signal mixes with
the original signal and some of the frequencies are out of phase with each
other and cancel or augment each other. A change in delay time changes the frequency
that is affected.
Harmonizers – Octave dividers – Aural Exciters:
are used for pitch shifting effects. They can be used to fix bad notes in some
cases, or to add harmonies in other cases.
dividers add an additional tone one to two octaves below the original signal.
This can fatten up an otherwise wimpy bass or kick drum.
work by adding slight distortion and phase shift to the signal. This can
brighten up an otherwise dull sounding instrument. They usually work the best
if there is a rich overtone sequence present in the original sound. The work
great on snare drums, background vocals, and string pads.
machines do as little as possible. If you have to combine vocals – record a
bunch, combine to one track – record next bunch – combine to next track. At all
costs avoid bouncing to adjacent tracks (feedback). Watch out for track next to
digital and analog machine. Record vocals on ADAT & combine to one track on
analog deck. Same quality as one original recording on analog machine. Adjacent
tracks not a worry on digital machines.
multiple tracks and combining to make one master track. If you can do it on the
digital deck, it is better. If you must do it on analog deck, try not to do it
If the way you
work is to try 4 takes, then comp, try 4 more then comp again. Don’t use the
comp track as a component and bounce to new track, try to take any pieces and
comp them into the existing comp track. That way comp track will never be more
than one generation down. (Make a safety track if you have trouble punching in
tracks. Erase unwanted material (with the supervision of the producer). Mixing
will be easier
Make a cue
sheet reminding you when to make what moves
Different DAT machines use different reference level.
reference level mean? analog vs. digital.
recording, “Zero” is a level reference at which there is 3% harmonic
distortion. Above this level there will be more distortion but a better signal
to noise ratio. Audio contains peaks which may be above this zero reference by
as much as 20dB. Analog tape compresses this information and records it with
more harmonic distortion, but for the small instance that the peak lasts, this
may not be a problem. If recordings are made at a lower level, the distortion
figures are lower, but the signal is dropping into the noise floor of the tape.
recording, “Zero” is the level above which no additional information
can be recorded. This results in hard clipping of the sound. Anything above
“Zero” is not recorded. A reference level of 18dB below
“Zero” allows room for peaks in the audio to be recorded without
clipping. Because the noise floor is so low in digital (98dB below
“Zero”) having a reference at -18dB does not really effect the
quality of the recording.
use too much of a good thing. Use just enough to provide the ambience or effect
you have empty tracks available on the multi track tape, record the effects to
free up equipment for something else, or to save time in re-mixing.
on record and playback – different ratios) Effects on vocals should be kept to
stereo placement should be determined by the final destination of the mix (TV,
video game, Surround Sound, CD, CD Rom). Keep in mind the center buildup
phenomenon. Avoid placing something all the way to one side. (keep in mind
stereo listening and being able to hear from opposite side of the room)
Analog 2 track
Otari, Ampex, Sony, Studer
30 ips vs. 15
ips vs. 7 1/2 ips
1/2 inch vs.
Dolby vs. Non
dbX and other
Emphasis on or
built in converters
Type of DAT
tape Computer backup DAT tape, Apogee, HHB.
3hour tapes unless machine is designed for it
machines. Use DAT cleaning tapes properly.
wears the heads.
Carver, Yamaha, Studer, Phillips, Micromega.
Mixing back to
two tracks of multi-track
48kHz or 44.1kHz? Stuck with whatever multi is.
Sample Rate Conversion:
To get from
one sample rate to the other or VSO final mix?
Analog out –
To change the
arrangement of the song
all of the tunes in order for distribution or going to mastering
editing (try to keep blood to a minimum). Razor blade editing can be performed
on reel to reel digital tapes under certain circumstances. Special precautions
need to be adhered to and a bad edit may not be reparable.
editing Akai, Sound Tools, Sonic Solutions, Turtle Beach Roland, SADiE, RADAR,
editors AKAI, Sony PCM 9000
Sony, Otari, Fostex Music editing vs. assembly editing
DAT is a NO-NO unless plenty of time between cuts.
vinyl records (South America etc.)
the correct order with proper spacing
Don’t do pause
edits on DAT machines unless 5 seconds around edit
(If that is
the only way, let mastering do it)
levels (If you don’t do it, Mastering will have to)
EQ All of the
selections should have similar tonal quality
When you are
done, Make a Digital Copy. Don’t send your only tape
can accept CD-R as master. It must not be Multi-Session
accept Sony 1630
will accept DAT( not if they have to edit)
accurate timing sheet (where you want each cut to start
Make them send
you a ref (plant or mastering facility)
done (eq, levels, editing) copy DAT to 1630
PQ codes on
tape? or PQ time sheet. Music @ 3:00 into tape
If you can
afford it, good idea to let mastering facility EQ and level correct your tape.
You want your product to be competitive with everyone else so it has to sound
as good. Third party reference is good.
breaks for cassette. Second side should be shortest
labeling of boxes and track sheets.
& J cards
notes. Which mic on which instrument.
was used to print to tape
What was the
interface was used to drive sequencer
What kind of
direct box was used through what preamp?
delayed? if so, which delay and by how much?
What kind of
tape was used and what was the machine set up for
What was the
reference level for recording.
tape was used to set up machine
tape and levels were used for Mix?
units were used and what were the settings?
settings for vocals or whatever?
If you printed
alternate mixes, what were the differences?
printed at different levels or different VSO settings?
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