SAE Recording Guide

Index
Drums and Percussion
Vocals and Harmonies
Stringed Instruments
Effects
Welcome to the SAE
Equipment
Mixing
reference material center
Acoustics
Absorbers
Construction
currently available: reference material - Audio Studies
Fitting out/Electrics
This section, written in an approchable and non-academic style, presents
Studio Plans
basic information about all aspects of recording from the point of view
STC Chart of an experienced recording engineer in Australia.
Helmholtz Calculator Multimedia and Digital Film studies will soon be available.
Tempo Chart
please register HERE to receive update information.
Reverb Calculator
Links
INDEX
A F P
Absorbers - Fader - channel Panel Absorbers

introduction Fittings - control room Panel Absorbers -
Acoustic Bass Flangers Variable
Acoustic Guitars Fletcher Munson Curve Parametric Equaliser
Miking Flexible Channel Patch Bays
Acoustic Guitars Floors Peak Meters
Tracking Fundamental Percussion - recording
Acoustic Hangers Frequency Calculator Phase - microphones
Acoustics - Phasers
Introduction G Piano - recording
Air Conditioning Playing in the control
Ambience Mikes - Gates room
drums Grand Piano - Polar Patterns -
Analogue Recorders - Recording microphones
Alignment Graphic Equaliser Power and lights
Alignment Tapes Guitars - Acoustic Power Conditioning
Automation Guitars - Electric Pressure Zone
Auxilary Sends Guitars - Bass Microphone - PZM
Auxiliary Send - Guitar lines - between Proximity Effect
mixing rooms
Azumith R
H
B Rear Reflections
Hard Disk Recorders Recorders
Balanced/unbalanced Harmony Vocals Reverberation
circuits 1 Harp Reverberation -
Balanced/unbalanced Helmholtz Resonator Calculation
2 Helmholtz Resonator Ribbon Microphone
Banjo Calculator Room Modes
Bass Guitars High Frequency Routing Switcher -
Bias - recorders Absorbers console
Bi & Tri Amping
Bit Rate S
Bongos I
Sampling Rate
C Inserts Skew - recorder head
Isolation alignment
Cable Ducts Insulco absorption Sound Pressure Level -
Car - sound factors chart SPL
Ceilings - Sound Transmission
construction J Class - STC
Ceilings - sealing Snare Drum - miking
existing Speakers
K Speaker Leads
Chorus Effect
Coefficient of Standing Waves
Kick Drum - recording
Absorption Staggered Studs
Kick Drum - Tuning
Coefficient of Star Earthing
Absorption Chart STC Chart
Compression L Stereo Image
Concrete Floors
Lighting
Condensor T
Microphone Limiters
Consoles Low frequency
Tambourines/shakers
Congas Absorbers
etc.
Cymbals and hihats Low-Mid Absorbers
Telephone lines
Tempo/Delay chart
D M Three phase power
Timber Floors
De-Esser Mandolin Toms - miking
De-Earthing Meters
Microphone Preamps
Degaussing Heads U
Delay Microphones -
Delay - setting Common types
Upright Piano -
quickly Microphones - Links to
recording
Diffusion manufacturers
Digital Recorders Microphones for drums
Mixing V
Digital Sound
Direct Boxes A Mounting Strips -
effects Valve Microphones
Direct Boxes B
Monitoring - console Variable Panel
Dobro
Monitoring Level Absorbers
Doors - construction
MS Stereo - miking Vibes and marimba
Drums - mike
Violin Family
placement Vocals - Recording
N
Drums - tracking Vocals - multitracking
Drums - Equalisation Vocals - Harmony
Noise Gates
Drums - Tuning VU Meters
Dynamic Microphone
Dynamic Range O
W
Organ - recording
E Output Groups -
Walls - Construction
Windows and Doors
console
Earthing - star Wind Shields -
Overdubbing - console
Electret Microphones microphones
Operation
Electric Guitars - Wrap - recorder head
recording alignment
Effects Units -
mounting Z
Equalisation
Equaliser - console Zenith - recorder head
Expanders alignment
ABSORBERS
By using different construction techniques it is possible to treat the walls so they absorb sound
at various frequencies. The following pages deal with each of the frequency bands-high, mid and
low. By using these construction techniques you can dramatically change the acoustics of a
room. High frequencies are the easiest to absorb and it gets harder as the frequency lowers.
Most home studio enthusiasts only seem to treat the high frequencies in the room yet it is the
mid and low frequencies that cause all the room problems.
If you study the treatment for the low and mid frequencies you will notice that the high
frequencies are not affected by this construction and it is possible to have a room where all
frequencies have a similar reverberation time which is the idea of the whole exercise.
RECORDING OTHER STRINGED INSTRUMENTS
The Harp
The harp is the ultimate stringed instrument and I sympathise with all the harp players as it is
the devil of an instrument to tune. They say mischievously that harp players spend half the time
tuning up and the other half playing out of tune! (sorry harp players)
There are basically two ways to record a harp. If it's on its own as an overdub a simple high
quality Condensor mike 30 - 60cm(1 -2 ft) from the instrument aimed at the striking point
(hands) will cover it fully. You can also use two mikes as a stereo pair similar to the position A
in pianos aimed at the top and bottom strings respectively and as close as the player finds
comfortable.
I used to often have a harp within a big band/strings situation where it was so quiet relative to
the rest of the brass etc. that a mike in this position was 80% spill so I had to find a better way.
What I found is the other way of recording a harp using the sound board as the source. Like the
holes in the frame of a grand piano the harp has a series of sound holes down the back. If you
get a quality mike and wrap it in a cloth you can jam it into the centre hole. The cloth will hold it
tight and stop any handling noise and you will get a good level signal that with a little top EQ
will work very well and have a lot less spill and you'll be able to mix those beautiful glissandos
over the brass. A combination of both mike positions will give a fuller richer sound if used
together.
The Banjo and Mandolin

The banjo and mandolin are similar to the acoustic guitar and both have the same points -
strike area, bridge and sound board - thus the same mike positions apply. I don't like to get too
close to a mandolin or banjo because their sound doesn't fully develop until around a foot or two
away.
The Dobro
The Dobro and lap steel both have a recording problem mainly the string noise as the slider
moves up and down the strings. If you think of them both as guitars on their backs and place
the mike in the acoustic guitar position B where the mike points back towards the striking
position you can put the fingerboard off axis to the mike and thus reduce the slide noise. You
can also mike the sound board and bridge as per the acoustic guitar.
The Violin Family

The violin, cello viola etc. are all the same except different sizes.
They all have a strike point (bow area), a bridge and a sound hole and soundboard. For the
violin and viola the typical miking is to place a quality mike 30 - 60cm(1ft - 2ft) above the
instrument pointing down aimed at the strike area. This gives a balance between the bow,
bridge and soundboard sounds. One technique I have tried is to mike the violin from underneath
as well as overhead with the bottom mike in the opposite position to the overhead mike and
phase reversed. By adding a little of the under mike you can add body and warmth to the sound
because you are adding more of the soundboard sound. It can also be said that a violin player
should be on a reflective floor as opposed to carpet because the sound emanating from the
soundboard will reflect back off the floor and add to the fullness of the sound.
The cello is the same with the mike out in front of the instrument pointing at the bow area.
Additional close mikes near the bridge and the sound board/hole can be used for effect if
required. Again a reflective floor is recommended.
The Acoustic Bass
I have singled out the acoustic bass because it is one of the hardest instruments to record in my
opinion. Being a classic stringed instrument it has all the sound areas - bow area, bridge, and
soundboard and soundhole. It depends on the style of music as to how you mike it but the
hardest is the straight plucked jazz bass. The traditional technique is to use a good mike
(preferably with a large capsule like a U87 or U49 and put it about 5 - 10cm(2" - 4") from the
bridge. This will emphasise the attack of the fingers with the added hardness that the bridge
sound has. Another mike can also be added that is aimed at the sound hole which will
emphasise the warmth and lower frequencies. A mix of these two should cover it nicely. Many
bass players have an electric pickup on their bass and a combination of direct pickup and mike
works well as the pickup adds presence.
Be very careful about the low end of the sound. It may sound silly but quite often to get a good
bass sound you have to remove bass from the signal. A low end rolloff from around 80 - 100Hz
can stop the bass from sounding muddy or a dip around the low mids at 200 - 300Hz will also
work. There is a lot of energy in the low end of an acoustic bass and reasonable compression
can help to contain it.
RECORDING ACOUSTIC GUITARS
The acoustic guitars is the classic stretched string instrument as it has all the positions where the sound varies. These
are the main positions that can be used to record an acoustic guitar.
Lets look at the different positions.
Position A
This is the standard crossed pair stereo miking position. This gives an overall sound of a guitar, it is not a tight presence
sound like position B but has the combination of the striking sound, the bridge sound and the body sound. The stereo
effect is not very wide but if you want the ambience of the room (like in a full strum rhythm track) it can be appropriate.
This can also be a position for a single mike.
Position B
Position B is the most popular mike position and the one I recommend for normal acoustic guitar recording. The mike is
placed about 15cm(6") from the guitar pointing at the end of the finger board but not directly at the sound hole. The
pick sound is emphasised in this position giving a nice clean attack to the guitar. This position also has the mike pointing
away from the fretboard so finger noise is reduced.
Position C
This mike is aimed directly at the bridge and is close around 10cm(4"). The sound here is harder sounding as it has less
low frequencies and the mid range sounds are emphasised. If I wish to record a stereo guitar I usually use positions B
and C and pan one mike left and the other right. In these positions the stereo spread is emphasised because
1.
The mikes are about a 30cm(1 foot) apart therefore increasing the difference thus the spread.
2.
The sounds are very different because each mike is recording a different aspect of the guitar sound.
Position D
Position D can be used as an alternative to position C for a warmer stereo sound as it doesn't have the hardness of the
bridge sound yet emphasises the warmer body sound. You must note that the mike position at the rear of the guitar
causes the mike to be 180 degrees out of phase to the mikes in the other positions therefore a phase reversal must be
used.
All Positions!
Why not use all positions? If you are about to record acoustic guitar tracks why not set up mikes in all positions and play
with the balance of each mike to gain the benefits of each. You might have a great stereo spread between positions B
and C yet adding some of position A will add fullness and body, or adding position D panned centre to do the same. Play
around, don't just limit yourself to one position only.
Tracking Techniques.
How we wish the guitar to sound in the track determines how we track it to tape and how many tracks we use. Lets look
at the various ways acoustic guitars are used.
Solo guitar as in folk singer.
Here you can either use position B and have the track in mono or you can create a stereo track using position B and C.
The thing about folk singers is that they sing and play at the same time!! so the guitar mikes are going to pickup the
vocal as well therefore any EQ, Reverb etc. that you put on the guitar will also affect the vocal. To get the minimum spill
of the vocal into the guitar mike I recommend you use position B and raise the mike so it points down at the guitar at
about a 45 degree angle but still in the B position. This tends to put the vocalist off mike to the acoustic guitar mike.
(You can also do the same with the vocal mike by having it pointing up at the singer and away from the guitar.) Another
method I've seen is to place a soft covered sheet of cardboard or timber horizontally above the guitar that divides the
spaces between the guitar and the vocal, but your guitarist must be able to play without seeing their hands!! but it does
work.
If the singer is going to overdub the vocal later then you can afford to make more of the guitar sound by recording it in
stereo but the singer must play the guitar track without singing.
Strummed Rhythm Guitar.
In this situation you may wish to have a single stereo/mono guitar track or you may wish to multitrack the acoustic. I
often double track an acoustic rhythm guitar with one panned left and one right. Another good method is to get the
guitarist to play one part through then to put on a capo and play the same chords but in a different position on the
guitar. This expands the guitar sound and sounds really good. Some people call it "adding a high strung" You play the
first part in say the standard C position and then play the part capoed up to the third fret but play it in the A position.
You can go even further , as I have often, and record two tracks in the C position and then do two tracks in the higher
capoed position. The effect is a wall of acoustic guitars. You can pan the two high strungs left and right and pan the low
strungs half left and right.
LOW FREQUENCY ABSORBERS
Low frequencies are big waves, consider that a 50Hz wave is 6.6m (21' 8'') and a 30Hz wave is
11m (36ft) long! That's 11m peak to peak -There's a lot of guys around here who would love to
surf a wave like that! So to stop it requires special techniques.
There are basically two ways to control low frequencies.
Acoustic Hangers. This is a system of fibre board panels that are wrapped with
insulation and are hung freely using wire or rope. The large hangers 1.8m x 500mm work
in the low frequency range whilst the panels 1.2m x 300mm effect the low mid
frequencies. It is common to have up to a 1.2m space at the rear of the control room
with the large hangers whilst the smaller hangers are effective if suspended in the ceiling
cavity created by a false ceiling.
Panel Absorbers. A panel of plywood or particle board is placed over an air cavity with
insulation glued to the back of the panel. The panel has a resonate frequency and when it
occurs in the room it resonates and the insulation absorbs the energy.
Acoustic Hangers
The above drawing shows the rear of a typical control room design. The fibreboard panels are
suspended from the ceiling with the sizes varying to give a broadband absorption field. They can
also be hung behind a false wall in the studio as in the following drawing.
False Wall with Acoustic Hangers
Panel Absorbers
A panel absorber is created when you place a sheet of plywood or fibreboard, with insulation
glued to the back of it, over an air cavity. The panel will have a resonate frequency of its own,
tap it and you will hear it. When it is placed over a sealed cavity, and insulation is attached to
the back, everytime it hears its own note it resonates and the air in the cavity resonates and
the insulation absorbs the resonance, hence absorbing the frequency! It is important to note
that here we have an absorber that reflects the high frequencies and attenuates the low. With
the hangers all that exposed insulation absorbs the high frequencies as well so the panel
absorber has a place in the studio. The two factors determining the frequency of absorption are:
The mass or density of the panel.

The depth of the air cavity, i.e. depth of the sealed timber frame.
A panel absorber is made like this:

You can apply different shaped front panels
The other great advantage of panel absorbers is that they can have angled or curved fronts so
when mounted on a wall or the ceiling they stop parallel wall interference and prevent standing
waves creating diffusion.
You can even tune this absorber by placing a contact microphone on the plywood panel which is
plugged into a real-time analyser and blasting the panel with white noise or a swept tone with a
speaker. When the frequency = the panel's resonate frequency the panel will vibrate and the
frequency will show up on the real-time analyser. The thicker the plywood the lower the
frequency and the greater the depth (depth of the timber box) from the wall the lower the
frequency. Using fibreboard as an alternative tends to create a low-mid absorber.
You can create a broadband low frequency absorption wall by building a series of sealed boxes
with different depths with each box being only 1m x 1m (3' x 3'). With a variety of different
thickness of plywood you can cover the whole low frequency range. It looks good too. You can
also alternate the fronts between panels and slats. (See helmholtz resonators)
For absorption coefficients and panel thickness check out the absorption coefficient chart.
Variable Panel Absorber

You can create a variable panel absorber by splitting the box into two boxes and placing hinges
on one side so that it opens fully as per the following diagram:
VARIABLE PANEL ABSORBER

The variable panel absorber allows you to change the acoustics in a room. A wall of these
absorbers can quickly change a room's acoustics from live to dead. A variation is to have a slat
resonator in the bottom box so that when the box is opened it reveals a slat resonator so you
end up with a wall of alternating low-mid absorbers and high frequency absorbers. If you can
only afford the space for one studio this is an excellent addition as you can change the room
acoustically to cover all situations.
LOW - MID ABSORBERS
If you look at the absorption coefficients of various materials you will notice that some of the
fibreglass products absorb low-mid frequencies very efficiently as does a panel absorber with a
fibreboard panel instead of a plywood panel. But the best low mid absorber (and the best
looking) is the helmholtz resonator - often called a slat resonator.
THE HELMHOLTZ RESONATOR

The helmholtz resonator (named after a Mr Helmholtz who discovered it) can best be
demonstrated by taking a normal soft drink bottle and blowing over the mouth of the bottle - a
note is produced. Now place some cotton wool in the bottle and try again. You will notice the
note has reduced- well not really, the note is produced but the wool absorbs the resonance and
turn the sound energy into heat! Imagine, if you lined a whole wall with bottles of various sizes,
all filled with insulation material. You would now have a low-mid (200 - 500Hz depending on the
bottle size) absorbing wall that as well as absorbing the low mids would also reflect or diffuse
the high frequencies. I haven't tried it yet but it would be worth trying if you are short of cash
because bottles are cheap. The Romans used to do it using clay jars which they placed around
their theatres.
The helmholtz resonator is often called a slat or slot resonator because you can create a
helmholtz resonator by building a wall with slats of timber separated by slots as in the following
diagram
The timber slats can be either finished or

rough sawn. If the gaps vary say 5mm,
10mm, 15mm,20mm and the wall is
angled as shown below, a broad band
low mid absorber is created that still
keeps the the high frequencies alive.
Remember the cavity behind must be
sealed to an airtight container, like the
bottle.
Further more, our scientists have created a formula with which we can tune the resonator to a
specific frequency. If we vary the depth from the wall, slat width, slot width (and the slat depth)
we can create a wall that is a broadband low-mid frequency absorber. The beautiful thing about
these absorbers is that they still reflect high frequencies, in fact they will diffuse them which is
even better.
The angle to the
wall here can be
either
horizontally, as
shown, or
vertically with
the 100mm at
the bottom of
the wall and the
300mm at the
top.
As you can see a slat wall like this can break up parallel walls thus stopping standing waves.
Because the distance from the front to the back is varying from 300mm to 100mm or around 12
degrees, the wall becomes a broadband absorber. So simple yet so effective! I've seen some
beautiful looking ones where you cut the slots out of a sheet of quality particle board with a
timber veneer.
Another form of helmholtz resonator is created using perforated plywood - i.e. plywood with
hundreds of holes in it. We call it pegboard in Oz, you see it in hardware stores holding up tools
etc. If you place a panel of this over an air cavity like in a panel absorber not only do the
little holes act like bottle necks the whole panel acts as a low frequency panel absorber!
The formula for calculating the helmholtz resonant frequency is:
f = 2160 x sqrt ( r / (( d x D ) + ( r + w )))

Where:
f = resonant frequency in Hertz (Hz)

r = slot width.
w = slat width.
d = effective depth of slot. (1.2 x the actual thickness of the slat)
D = depth of box.
You can open the helmholtz formula as an Excel file and do your own calculations.
You must have Microsoft Excel on your computer for this
file to open.
ABSORPTION COEFFICIENT CHART
I got this chart off the web and it gives you an idea of how the different materials absorb sound
at different frequencies.
Remember that full absorption is 1 whilst full reflection is 0
Absorption coefficients of common building materials and

finishes
Floor materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz
carpet 0.01 0.02 0.06 0.15 0.25 0.45
Concrete (unpainted, rough finish) 0.01 0.02 0.04 0.06 0.08 0.1
Concrete (sealed or painted) 0.01 0.01 0.02 0.02 0.02 0.02
Marble or glazed tile 0.01 0.01 0.01 0.01 0.02 0.02
Vinyl tile or linoleum on concrete 0.02 0.03 0.03 0.03 0.03 0.02
Wood parquet on concrete 0.04 0.04 0.07 0.06 0.06 0.07
Wood flooring on joists 0.15 0.11 0.1 0.07 0.06 0.07
Seating materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz
Benches (wooden, empty) 0.1 0.09 0.08 0.08 0.08 0.08
Benches (wooden, 2/3 occupied) 0.37 0.4 0.47 0.53 0.56 0.53
Benches (wooden, fully occupied) 0.5 0.56 0.66 0.76 0.8 0.76
Benches (cushioned seats and backs,

0.32 0.4 0.42 0.44 0.43 0.48
empty)

0.44 0.56 0.65 0.72 0.72 0.67
2/3 occupied)

0.5 0.64 0.76 0.86 0.86 0.76
fully occupied)
Theater seats (wood, empty) 0.03 0.04 0.05 0.07 0.08 0.08
Theater seats (wood, 2/3 occupied) 0.34 0.21 0.28 0.53 0.56 0.53
Theater seats (wood, fully occupied) 0.5 0.3 0.4 0.76 0.8 0.76
Seats (fabric-upholsterd, empty) 0.49 0.66 0.8 0.88 0.82 0.7
Seats (fabric-upholsterd, fully

0.6 0.74 0.88 0.96 0.93 0.85
occupied)
Reflective wall materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz
Brick (natural) 0.03 0.03 0.03 0.04 0.05 0.07
Brick (painted) 0.01 0.01 0.02 0.02 0.02 0.03
Concrete block (coarse) 0.36 0.44 0.31 0.29 0.39 0.25
Concrete block (painted) 0.1 0.05 0.06 0.07 0.09 0.08
Concrete (poured, rough finish,

0.01 0.02 0.04 0.06 0.08 0.1
unpainted)
Doors (solid wood panels) 0.1 0.07 0.05 0.04 0.04 0.04
Glass (1/4" plate, large pane) 0.18 0.06 0.04 0.03 0.02 0.02
Glass (small pane) 0.04 0.04 0.03 0.03 0.02 0.02
Plasterboard (12mm (1/2") paneling

0.29 0.1 0.06 0.05 0.04 0.04
on studs)
Plaster (gypsum or lime, on

0.01 0.02 0.02 0.03 0.04 0.05
masonry)
Plaster (gypsum or lime, on wood

0.14 0.1 0.06 0.05 0.04 0.04
lath)
Plywood (3mm(1/8") paneling over

0.15 0.25 0.12 0.08 0.08 0.08
31.7mm(1-1/4") airspace)

0.28 0.2 0.1 0.1 0.08 0.08
57.1mm( 2-1/4") airspace)

0.38 0.24 0.17 0.1 0.08 0.05
50mm(2") airspace)
Plywood (5mm(3/16") panel,

25mm(1") fiberglass in 50mm(2") 0.42 0.36 0.19 0.1 0.08 0.05
airspace)
Plywood (6mm(1/4") paneling,

0.3 0.25 0.15 0.1 0.1 0.1
airspace, light bracing)
0.28 0.22 0.17 0.09 0.1 0.11

0.2 0.18 0.15 0.12 0.1 0.1
Absorptive wall materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz
Drapery (10 oz/yd2, 340 g/m2, flat

0.04 0.05 0.11 0.18 0.3 0.35
against wall)

0.05 0.07 0.13 0.22 0.32 0.35
against wall)

0.05 0.12 0.35 0.48 0.38 0.36
against wall)
Drapery (14 oz/yd2, 476 g/m2,

0.07 0.31 0.49 0.75 0.7 0.6
pleated 50%)
Drapery (18 oz/yd2, 612 g/m2,

0.14 0.35 0.53 0.75 0.7 0.6
pleated 50%)
Fiberglass board (25mm(1") thick)

0.06 0.2 0.65 0.9 0.95 0.98
Fiberglass board (50mm(2") thick) 0.18 0.76 0.99 0.99 0.99 0.99
Open brick pattern over 75mm(3")

0.4 0.65 0.85 0.75 0.65 0.6
fiberglass
Pageboard over 25mm(1") fiberglass

0.08 0.32 0.99 0.76 0.34 0.12
board

0.26 0.97 0.99 0.66 0.34 0.14
board

0.49 0.99 0.99 0.69 0.37 0.15
board
Performated metal (13% open, over

0.25 0.64 0.99 0.97 0.88 0.92
50mm(2") fiberglass)
Ceiling material 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz

Plasterboard (12mm(1/2") in
0.15 0.11 0.04 0.04 0.07 0.08
suspended ceiling grid)
Underlay in perforated metal panels

0.51 0.78 0.57 0.77 0.9 0.79
(25mm(1") batts)
Metal deck (perforated

0.19 0.69 0.99 0.88 0.52 0.27
channels,25mm(1") batts)
Metal deck (perforated channels,

0.73 0.99 0.99 0.89 0.52 0.31
75mm(3") batts)
Plaster (gypsum or lime, on

0.01 0.02 0.02 0.03 0.04 0.05
masonary)
Plaster (gypsum or lime, rough finish

0.14 0.1 0.06 0.05 0.04 0.04
or timber lath)
Sprayed cellulose fiber (16mm(5/8")

0.05 0.16 0.44 0.79 0.9 0.91
on solid backing)
Sprayed cellulose fiber (25mm(1") on

0.08 0.29 0.75 0.98 0.93 0.76
solid backing)

0.47 0.9 1.1 1.03 1.05 1.03
timber lath)
Sprayed cellulose fiber (32mm(1-

0.1 0.3 0.73 0.92 0.98 0.98
1/4") on solid backing)

0.7 0.95 1 0.85 0.85 0.9
solid backing)
Wood tongue-and-groove roof

0.24 0.19 0.14 0.08 0.13 0.1
decking
Miscellaneous surface material 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz
People-adults (per 1/10 person) 0.25 0.35 0.42 0.46 0.5 0.5
People-high school students (per

0.22 0.3 0.38 0.42 0.45 0.45
1/10 person)
People-elementary students (per

0.18 0.23 0.28 0.32 0.35 0.35
1/10 person)
Ventilating grilles 0.3 0.4 0.5 0.5 0.5 0.4
Water or ice surface 0.008 0.008 0.013 0.015 0.02 0.025

RT60 relates to intelligibility. Diffractors reduce pronounced reflection by breaking up the sound
wave before reflecting it back. This does not reduce reverberant energy, but does reduce echo
spikes that may otherwise exceed -60db of direct, thus lowering RT60 and improving
intelligibilty, but not necessarily improving the listening environment for music.
ACOUSTICS
So you want to setup a studio so why worry about acoustics
and what are acoustics anyway?
Let me tell you a story as I heard it from the late Dean Jensen who I stayed with in the early
seventies when he was the top techo in LA. --- According to Dean in the early seventies some of
the engineers of The Record Plant (one of LA's top studios ) were sitting on the roof of their
building after a long party. They had taken some speakers from the studio up to the roof for the
party and were discussing the fact that the speakers sounded great on the roof but were pretty
awful downstairs in their control room! Why?? Well, on the roof the speakers were in what is
called an anechoic environment - i.e. no reflections or reverberation! The sound left the
speakers, went past them and didn't come back. Try it yourself - take your favourite speakers
outside and set them up in your backyard or in an open field and have a listen to them.
Suddenly the bottom end will be clean and tight and the top end imaging will be amazing. The
centre will be really tight and defined and you will hear all the mistakes you made in the
recordings. Unfortunately your neighbours won't let you set up a control room in your backyard.
I might mention here that it amazes me how much EQ front of house engineers apply to
speakers when mixing outside concerts where there are no room modes etc. I remember an
engineer being highly offended (and confused da!) when I suggested that JBL must make
terrible speakers when they needed +/- 12db EQ in the open air. (I was the first in Australia to
add 1/3 Octave EQ my studio monitors in 1974 when I got back from LA. Within 6 months I'd
removed them.)
So what is the ideal internal listening environment? Well I reckon if you did a vote you would
find that over 50% would say - In the CAR!!!. I agree, I often check a mix in the car and I know
a lot of other engineers who do the same. So why does the car sound so good? There are a
number of factors and it is these factors that go into making a good listening environment.
The main factors in a car are:

The Shape: There are no parallel walls in a car and what walls there are are thin and
curved.
The Speakers. In a car the speakers are almost always flush mounted. i.e. They a
mounted into a flat surface like below the rear window or in the side door panels. As a
result there are no out of phase signals coming from the rear of the speaker. Also the
rear speakers are mounted in a big cabinet - the boot.
High Frequencies: In the car the windows are the main high frequency reflectors but
they are all at angles (approx. 6 - 12 degrees)and are usually curved as well. The highs
also get diffused evenly throughout the cabin by the dash board. Also the ceiling, sides
and floor are covered in high frequency absorption.
Mid Frequencies: The seats, door panels and passengers are all low mid to high mid
absorbers. Modern cars have deep pile carpet on thick underfelt which also acts as a mid
frequency absorber. Most of the car's acoustic treatment for cutting down engine and
road noise is also on the inside and acts as acoustic treatment for the car stereo.
Low Frequencies: The beautiful thing about cars is the bottom end response. With a
couple of hundred watts a side, a sub-woofer under the seat and the loudness switch on
the bottom end thumps away and sounds great. Actually most of the low end goes
straight through the walls and disappears, consequently it doesn't hum around the
internal body causing phase problems. Any vibration is dampened by the foam lining and
carpet and as far as the low end is concerned the car is equal to open air. Next time you
play a tape/CD in your car get out and listen to what actually leaves the car (most of it!!
especially if the windows are open)
The problem with recording studios is that to keep external sounds out you land up keeping
internal sounds in. People who build studios in the city have to worry about trains underneath,
traffic noise outside, planes flying overhead etc. Obviously the best thing to do is to build it in
the middle of a 50 acre paddock in the country where your only external noise concerns are
birds, wind and rain. Then you can build a simple skin to keep the rain out and allow all the
internal sound to get out so it doesn't muck up the sound within the room.
So how do you create the effect of your car, or the open air, in your studio? - By using
Acoustics!! Treating the walls in your control room and studios so as to control the sound and
thus improve the quality of the sound that you hear and record. I reckon good acoustics can
beat a fancy effect unit any day and they cost about the same.
You can now select any of the topics listed in the adjacent column and progressively gain a good
understanding of the basics of building a quality studio for yourself..
AIR CONDITIONING
If you've built all the rooms of your studio correctly you will have airtight sealed rooms so the
opportunity for external air to enter the building has been totally eliminated. You must therefore
have some form of air-conditioning to not only keep your air temperature constant but to supply
fresh air. An air conditioner consists of three components.
The compressor unit that creates the cool/heat.

The fan unit that moves the air.
The Ducts that distribute the air.
These units can be assembled in three ways
The standard free standing unit that has all units in one.
The split system where the fan unit is in the room and the compressor unit is outside.
The fully external system where the fan and the compressor are external and the air is
circulated via ducting.
The following must be considered.
Number of units. You must have one unit for the control room and a separate unit for
the studio. I've found that when you use one unit for both you land up with a nice
temperature in one room whilst the other room freezes. Never underestimate how much
heat normal recording equipment produces. If you add up the wattage of all your gear
and imagine a heater of the same wattage you will find that you have around a 2000 watt
heater going in your control room all the time whereas in the studio the only heat is the
occasional guitar amp and body heat. Consequently if the thermostat is in the control
room when the control room is fine the studio freezes. I recommend two separate units.
Type of unit. There are three types of air-conditioners available as previously stated.
The main difference here is that only the fully external system can add fresh air.
Noise Factor. Obviously having an air-conditioner grinding away in a studio is not much
good if you are recording quiet instruments but if you've got a metal band in there who
cares!! Lots of home studios have a split system (That's when the compressor is external
and the fan unit is in the room) There is no air exchange and if you are recording quiet
instruments you just turn it off for a while. It is the cheapest system and, to be honest,
the most common. Fresh air is achieved by leaving the doors open when isolation is not
required.
I think that if you check your local building codes you will find that because your rooms are
totally airtight they will require you by law to have a fully ducted system which adds fresh air.
To install this system you will have to breach the air seal that you have carefully created but
there are ways to get around this. The typical ducted system works like this.
TYPICAL AIR-CONDITIONING UNIT
The above drawing shows a typical external system. The compressor and fan are in the unit
externally and the ducts send and return the conditioned air.
The Ducts. The typical duct found in a home air-conditioning unit are OK for the standard home
studio. They normally consist of a aluminium foil pipe wrapped in fibreglass and plastic. The
external ducts on the other hand must be soundproof and are typically made from galvanised
iron boxes lined internally with sound absorption material. Just remember that the larger the
duct the quieter it will be.
The Noise Factor: The slower the air in the duct the quieter the air-conditioning system will
be. Some of the top studios have ducts which are 1 - 2 M (3 - 6ft) wide and the air moves
slowly within the duct. This stops the hiss you get when the air enters the room via a duct.
As stated the seal between the external duct and the internal duct is very important. The
common way is to join the two ducts with a flexible join so that the two ducts aren't
mechanically linked.
FLEXIBLE JOIN BETWEEN DUCTS
The important factor here is to make sure that it is totally sealed. The flexible joint can be made
out of a product that is made from vinyl impregnated with lead. Ask your air-conditioning
company about it, they will know. This gives a flexible soundproof seal.
Fresh Air: I must say something here about adding fresh air to a system. I have found that the
formula used by most companies for adding fresh air doesn't add enough. When you have a
control room with a producer, engineer, musos and a few hangers on the amount of fresh air
added is usually insufficient and by halfway through the session everyone is yawning. They
usually add about 15% fresh air to the system but I would urge you to consider adding up to 25
- 30%. This is more important when you have a fully sealed system in a city building where
there is usually no external windows and the outer office/hallways etc. are fed by the buildings
own system. Creativity requires plenty of fresh air!!!
Obviously the air-conditioning can be extremely expensive.
No doubt your budget will be the determining factor.

MICROPHONE PLACEMENT
Let's take a look at a standard drum kit.
Looking from the top this is the standard layout of right handed drummer. What we have to do now is Mike it! So let's
start by putting two mikes over the top (generally called the "Overheads"). But where? We have to start thinking in
terms of a stereo image right from the start.
The standard panning setup in drum recordings is:
Kick - Centre
Snare Centre
Hats - Half right/right
Cymbals - left - right
Toms - left - centre - right.
But if you look at a kit it isn't really setup like it should sound. The snare is to the right, the toms have no spread etc. In
fact if you were to put up two overheads left and right the stereo image would have the kick centre and the snare to the
right and the toms going from centre to left.
Sound Picture Created
Yet on recordings if you imagine looking at the kit from the front it looks (sounds) like the snare is centre and the toms
spread from left to right. (You will notice that I refer to the imaging as a picture - well that's what it is!) So can we get
the overheads to paint a picture like this? Have a look at this setup which is based around drawing an imaginary line
through the kit which lines up the kick and snare etc.
The microphone placement places a stereo image similar to where you want to go as far as the placement of the
different components.
The mic placement will look like this"

You can now start to hear a stereo image of the kit as you will want to hear it in the mix. If you were to put the mikes
together in a stereo pair but aimed each side of the dividing line you would get a stereo image but a narrow one. The
width increases as you move the two mikes away from each other. The placement in the drawings above are about
normal with enough spread to make the kit have some width. With practice and careful placement of these two mikes
you should be able to get a good balance of the kit. If you added a kick drum you would have a real, open sound of the
kit.
The next step is to mike the individual components so that their position and individual sound can be emphasised.
KICK DRUM
There are three ways of setting up a kick drum
Front and rear skins on.

Front skin with rear skin with hole in it
Front skin only.
These three set-ups create three differing sounds. First you must tune the kick as per the directions in the tuning
drums page.
The first setup with both skins is a thick, solid, round sound with a decay as the drum decays. I believe the best way to
mike up this setup is to use two mikes. One over the pedal and one at the other end like this.
This setup allows you to balance the attack sound of the beater with the decay of the front skin. This miking setup also
brings up and important factor in recording:
Microphone Phase Relationships
So which mike should you phase reverse??. If you look at the microphone over the beater it is pointing downwards like
all the other microphones on the kit will do whereas the microphone on the front skin of the kick drum is facing the
opposite way. Therefore the front skin mike should have the phase reversal. As you can see it is a good idea to reverse
the phase of your kick mike even when you are not using two of them as the normal kick mike setup places the kick
mike out of phase to the rest of the kit mikes.
Similarly, when we get into miking toms and snares top and bottom the bottom mike will require a phase reversal.
The next setup is where the kick drum has a front skin on with a hole in it. Because of the hole you can access the front
skin - thus the attack sound - without having to use a beater mike.
Here the mike is placed inside the drum pointing to where the beater hits so as to get the full impact of the beater. Note
that the mike is still out of phase to all the downward facing mikes on the kit so a phase reversal is preferred. The mike
is also placed off centre within the shell.
Another factor effecting the kick sound is the beater the drummer uses. Beaters vary from soft to hard. Hard beaters
(usually wood) have more impact sound than the softer beaters. Experiment with each and you will hear the difference.
How close to the centre of the skin the beater is placed also varies the sound. Similarly the size of the drum sticks the
drummer uses will also effect the sound - thin sticks aren't going to go boof! no matter how much you EQ them.
Sound Pressure Level:
It should be noted here that the SPL (Sound Pressure Level) created by drums is extreme so you must select a
microphone that can handle high SPL and even then it will output a high voltage into the console. Therefore a
Microphone PAD should be inserted in the console to prevent the front end of the microphone preamplifier distorting. If
your console doesn't have a mike pad switch you should insert one in the microphone lead. Like the phase reversal plug
you can purchase mike pad plugs from your local dealer. A pad of anywhere from 10db - 20db will be required.
A note here about mike pads.
TOMS
The toms are similar to the full kick drum miking in that there is a mike on the impact skin that gets the full attack of
the stick when it hits the drum plus you can also add another optional bottom mike to get the hang of the the drum.
You must again remember the phase relationships here. If you wish to add a bottom mike to the toms you must reverse
it's phase.
RACK TOMS FLOOR TOMS
If your drummer doesn't have a bottom skin on the toms you can use either a top mike or both mikes or you can opt for
just one under mike with a phase reversal naturally. The advantage here is that the under mike is inside the tom which
isolates the mike from the other drum sounds and improves separation.
CYMBALS AND HIHATS
The Cymbals
These are basically covered by the overheads but you might find that the ride cymbal needs a mike of it's own if the
drummer rides it a lot through the chorus. Basically you want the crash cymbals to have a loose sound yet the ride
often is the main drive as it replaces the hihat for the 8 a 16 feels. You must consider this factor when setting up the
overheads. Drummers also accent using the bell of the ride cymbal that can be extremely loud so beware of miking too
close to the bell of the ride cymbal or it will dominate the sound field. Some engineers mike the ride from underneath.
In a complex drum setup with lots of splash and crash cymbals you might like to spot mike certain cymbals but I reckon
that if you've setup your overheads correctly they should cover the full cymbal range.
The Hihats
Like the overheads the hihat also requires a mike with a clean top end so it's usually a condensor mike. I like to hide the
hihat mike from the snare by placing it in a position that is pointed at where the drummer impacts it with his stick but
the hihat is physically between the hihat mike and the snare.
Good separation between the hihat and the snare is desirable so consider the snare when you place the the hihat mike.
Another factor of the hihat is the sound made when they are snapped together. I like to aim the mike so it is pointing at
a point that gets the stick impact as well as the pointing at the edge of the hats as that is where the closing sound
emanates. N.B. If you get too close you will get wind distortion from the hats as they close.
One of the problems you can get is where the drummer has the hihat low to the snare and the toms also low to the
snare. This creates separation problems as well as making it hard to isolate the snare from the tom. There's not much
you can do other than ask the drummer to change. This is not as awesome as it sounds, some drummers have never
considered this aspect of their kit layout and on making the change actually say it's OK and find they easily got used to
it and now prefer it. The same problem can occur with the ride cymbal - some drummers have their ride cymbal almost
touching the floor tom which makes separation hard - I recently had a drummer like that and when I mentioned it he
agreed to change. After the session he remarked that he actually liked the change and would do it in future. Moral of
this story? - don't be afraid to ask!!
SNARE DRUM
Once again, the snare can be miked from the top and the bottom, in fact it is one most often double miked. The bottom
mike on a snare can give the snare more depth but it also gives you control over how much snare crack sound is in the
overall sound. (The snare is actually the stretched wires across the bottom skin and gives the snare it's sound -
otherwise it's just another tom). The snare mike is normally squeezed in between the hihat and the first rack tom and
like the tom mikes is aimed at the main impact area in the centre of the snare.
Side Stick: Often you have a drummer playing a lot of side stick. I have used a separate mike specifically for the side
stick. The side stick action is for the drum stick to hit the rim of the snare drum and the main impact is on the right side
of the snare. As your normal snare mike is placed on the left side it doesn't always pick up the side stick clearly. Not
only does this give you a mike closer to the side stick action it also allows for different EQ and effects for the side stick
sound. You can either track it to a different recording track or you can watch the drummer and switch mikes during
record.
AMBIENCE MIKES
Drums miked close-up don't actually sound very real as their real sound is a combination of various factors. You actually
have to get away from them to get the full sound. A close mike on a snare doesn't really sound like a snare drum (thus
the importance of the overheads) so some engineers add Ambience mikes to allow the freedom to add the distance
sound of the kit when mixing. Naturally the drums must be in their own room for this system to be used or the
ambience mikes will pick up everyone else in the room. Basically ambience mikes are a stereo pair of mikes placed at a
distance (room size limited) from the kit. They can be setup as a crossed pair or moved apart to gain a more ambient
spread. You might like to try using a MS Stereo mike setup. Ambience mikes can also be Gated - so they only
open when the snare is hit for example- and you must have plenty of recording tracks to allow for another stereo pair.
Ambience mikes effect all the kit and push the drum kit back in the sound field so if you want a round tight kick sound
and an ambient snare sound you have to gate them so they are closed for the kick and open for the snare. An engineer
I know used to hang a very directional shotgun mike high above the kit aimed at the snare and use the under snare
mike to trigger a gate that opened it whenever the snare was hit. He would then mix it in with the snare sound and it
gave the snare a natural ambience and was extremely effective.
So now we have set up all the mikes we are ready to start balancing and equalising them
Microphones for drums
The Kick. What are we looking for in a kick drum mike? Firstly and most importantly it must be capable of
withstanding high sound pressure levels!! When a mike is only inches away from a kick drum beater the sound
pressure levels are extremely high at low frequencies. The kick drum mike must be capable of handling the
extreme transients involved. Secondly it must be capable of reproducing very low frequencies.
The two most popular kick mikes are - The AKG D12 and the Beyer M88. The M88 is my favourite. Both these
mikes have an extended bottom end response and can handle the high sound pressure levels associated with kick
drums. On the other hand if the drummer is not hitting too hard you can't beat the Neuman U87 or 49, which are
high quality condenser and have large diaphragms (good for low frequencies) and smooth low end response.
Other mikes are the Shure SM57/58 and the old RE20 which are both capable of withstanding the load.

The Snare. Here we are looking for a mike that will withstand extreme high end transients and has a tight
pattern so as to keep out the high hats and the adjacent toms. The most common snare mikes would have to be
the Shure SM57 and the Sennheiser MD421, followed by the Neuman U87/89 and the AKG 414EB. Others are the
Sennheiser MD441 or the Neuman KM84. I'm always amazed at how many engineers still use the Shure SM57
even though there are lots of other mikes around. The main advantage of the SM57 is that it's a tight mike with a
tight pattern that keeps out the spill from the hi/hat and the toms. They are also extremely reliable and don't
mind being hit by a wayward drummer. I should note here that the difference between the SM57 and the SM 58
is that the SM58 has a permanent wind shield - the microphone section is identical. You can buy a wind shield for
the SM57 (Note the two microphones next time you see a press conference from the White House.)
The Toms. The two main mikes used for toms are the Sennheiser 421 and the Shure SM57. In the studio I like
to use Neuman U87's as they have a beautiful warm bottom end. The Shure SM 57's don't have a lot of bottoms
but if you're tight miked the proximity effect compensates for it and as with the snare their tight pattern helps.
Overheads. Good condensor mikes make the best overheads. There are three main overhead mikes, the
Neuman U87 for warmth, the AKG 414EB and the AKG 451 for crystal clarity. The AKG C1000 and the Roden are
also a good budget condensor overhead mike except I find that both have a slightly tinny top end compared with
the more expensive models. I would say the AKG 451 with a CK1 capsule and 10db pad is the most popular
overhead mike.
Hihats. Condensor mikes with a tight pattern make the best Hihat mikes like the AKG 451 or the Neuman KM84.
Both have a 10db pad option which is handy as the high end transients from a hihat are extreme.
Ambience Microphones. Usually high quality condensor mikes are used here.
TUNING DRUMS
" How do you get a great drum sound?" -- "Get a good drummer!" Like every other instrument,
drums must be properly tuned and good drummers know how to tune their drums. They also
know how to hit them consistently on the right spot so their sound is true. You can tell a good
drummer by looking at the skins. If they are worn in a small circle in the middle - they are good -
whereas if there are stickmarks all over the shop they're not consistent.
The kick, toms and the snare are all tuned the same way. I recommend you take the drums off
the kit to tune them.
Note: When you start with new skins it is a good idea to stretch the skin once it is on the drum.
A drummer suggested to me that he stands on the drum (especially kick drums) and lets his
weight stretch the skin!! He swears by it.
First, make sure the bottom skin is nice and tight. The bottom skin is usually tuned higher than
the top - about a fourth up - every drummer has his own tuning, the main thing is to make sure
that it is even. To tune a skin evenly you must put your finger in the centre of the skin and tap
the outer part near each tuning point.
You will notice that each point produces a slightly different note so you go round tuning each
point so that they all produce the same note. I usually work on the pins opposite each other
because as you tighten one the opposite pin is effected. You do this procedure for both top and
bottom skins. You must also tune each drum relative to the other drum so that the high toms
progress down to the low toms.
Once you've tuned each drum mount them back onto the kit. Now if you hit each tom a pure
note will sound. If you now take one of the lower tuning pegs (the one closest to you) and start
to lower (unscrew) it as you keep hitting the drum you will find a point where the skin hangs out
and even appears to drop in pitch. A straight tom sounds like doom whereas one with one pin
detuned sounds like dooommmmmmmm with the mmmmmmm dropping in pitch and the whole
note lasts longer. That's how you get the t-tooo t- tooo t-tooooo fill sound because the toooos
drop in pitch. (Still with me??)
The snare is the same except that you don't want to detune one of the pins. Consistency of pitch
at all the tuning points is essential on both top and bottom skins.
Damping: I try and avoid dampening toms. The old system of a piece of Gaffa Tape all over the
toms doesn't produce a very good sound. If you need to dampen the toms or the snare I suggest
a piece of dacron (polyester wool) with a piece of gaffa stretched over from rim to rim but with
the gaffa not touching the skin, only the wool.
This way the skin is not choked too much and you can apply small amounts of dampening. The
toms are going to ring - but so what. You can always gate them out later or better still
automate them out of the mix with fader/mute automation. If you dampen them down so they
don't ring they will sound like cardboard boxes like Ringo used to play. There are now available
small squares of some kind of synthetic rubber that feels like a jelly baby which you place on the
skin. They work very well for dampening because like the dacron they don't choke the skin.
You can also dampen Cymbals by sticking a small strip of gaffa tape to the underside of the
cymbal. This is only necessary when the cymbal (usually the ride) is too ringy and lacks
definition.
KICK DRUMS
Kick drums are another story. There are three ways a kick drum can be setup depending on
whether the drummer uses one or two skins or has two skins with a hole in the front skin. Some
drummers actually line the inside of their kick drum with a layer of foam that acts as a
permanent dampener.
Both skins on
This is the traditional kick drum setup. Having tuned the drum using the previously suggested
method you must next determine whether it needs dampening. If you feel it does (typically) you
can use a pillow or a blanket pressed up against the front skin and held in place with a sand bag,
brick, mike stand base, or anything with weight lying around the studio.
Both skins but with a hole in the front skin
In this situation you have access through the hole to place dampening inside the drum. Once
again a blanket or pillow is placed on the base of the drum and held in place with a sand bag or
weight.
Beater skin on only.

Here the front skin has been removed allowing dampening to be placed in the drum as in the
previous example. This is the most typical system yet I notice nowadays that there are a lot of
drummers opting for the more traditional sound of using both skins and going for a more
"natural" kick sound as opposed to the clicky percussive sound used throughout the last few
decades.
Additional Option
Now that the kick drum has been dampened to your likening may I suggest you can dampen the
whole drum by placing a blanket (or better still a sleeping bag) over the whole kick drum. This
helps gain isolation of the microphone in the drum from the rest of the kit. ( a sort of acoustic
gate) Additionally a weight (sand bag) can be placed on the kick drum to make it rigid and
dampen the shell vibrations.
MICROPHONES
With the plethora of microphones around you'd be surprised at how engineers all over the world
seem to use the same mikes. Go surfing to all the studios and you'll find the same mikes in their
mike list. I've got to state here that I'm not pushing any particular brand or type - I am not
sponsored - so I'm only stating what I've observed over the years.
So how do they work? Basically all microphones have a diaphragm that vibrates when hit by
sound waves. The vibration of the diaphragm is translated into an electrical signal that
corresponds to the variation in the sound wave. That is why it is necessary to clean the
diaphragms in your mikes on a regular basis as a build-up of dust, spit etc. will impede the
vibration of the diaphragm and thus distort or colour the sound.
The Dynamic
In a Dynamic microphone, also referred to as a moving coil microphone, the capsule is rather
like a speaker in reverse. The cone is the diaphragm and it has a coil attached that is suspended
in a magnetic field. When the diaphragm vibrates the coil creates an electrical current. This is an
entirely passive circuit as the magnet can be a permanent one so no external power is
required.
The Condensor
On the other hand the Condensor microphone has two plates, one fixed and one moveable,
that are each charged with a polarising voltage that creates a capacitor. The vibration of the
plates creates a change in the distance between them which changes the capacitance and thus
the sound wave is converted into an electrical current. In this case external power is required
as there is an electrical circuit required to produce the polarising voltage. Because the current
obtained is so small an amplifier circuit is also included.
Thus when using Condensor mikes an external power supply is required. This can be either a
stand alone power supply for one or more mikes or it can be fed to the microphone from the
console down the microphone cable and is commonly referred to as Phantom Power and is now
standard at 48 Volts and all new consoles have that facility and usually consists of an on and off
switch on the rear of the console or is an on/off option on each module. Incidentally, don't worry
about sending phantom power to a dynamic microphone, it won't blow it up as the circuit is
inactive in a dynamic mic situation.
The Electret
An Electret Microphone is also a Condensor microphone except that the charge on the plates is
created by a permanent electrostatic charge. Therefore an external polarising voltage is not
required but once again the voltage obtained is small so an amplifier is usually built in and
powered by an internal battery. Electrets are often thought of as the cheap cousin to the
condensor mike because the material required to hold the charge on the diaphragm is heavier
but good electrets can sound fine.
The Pressure Zone

The PZM or Pressure Zone Microphone is also an Electret microphone except that it is
mounted in a special housing near the pressure zone on the surface of a plate. This plate can be
mounted on a flat surface like on the wall, floor or the lid of a piano. I have found that PZM
mikes are not prone to popping and appear to have no proximity effect. They are typically used
for pianos in concert situations where the lid can be closed to reduce spill and are also ideal as
floor mikes in stage show productions.
The Ribbon
The Ribbon Microphone consists of a thin metal ribbon that is placed in a magnetic field. The
vibration of the ribbon within the magnetic field induces a current that is proportional to the
variation in the sound wave. This is also a passive circuit as the magnetic field can be created by
a permanent magnet.
The Valve Microphones

Finally I must say something here about Valve Microphones. As mentioned before, the signal
from the diaphragm in a Condensor microphone is small and must be amplified before it reaches
the console where again it is amplified further. It is within this area that signal deterioration can
easily occur and therefore the quality of the microphone must also be judged by the quality of
the first stages of amplification. In a valve microphone the Condensor stage is a standard
condensor system but the amplifier section uses a valve circuit to amplify the current as opposed
to a transistor circuit used in later models. When I first started as an engineer in 1966 all the
Condensor microphones were valve and each had its own power supply. The introduction of the
transistor microphone eliminated the need for power supplies because phantom power was
invented for the purpose.
The other major factor in those days was signal to noise. The average tape recorder had a signal
to noise ratio of around 58db as opposed to the 70+ with today's analogue recorders.(Mainly due
to the improvement in the surfacing of tape.) With such a low signal to noise ratio we were
always careful about the high end of our recordings because if you had to add it later you
sacrificed your noise and increased hiss. So when the transistor microphone came out we all
remarked "Far out!" (it was the 60's) listen to that top end!!" and immediately used them instead
because records were getting brighter then. What we were hearing was the difference in
distortion between a valve and a transistor. A valve distorts in the 2nd harmonic first whilst a
transistor distorts in the 3rd harmonic. The 2nd harmonic distortion is smooth, we can handle it
but 3rd harmonic distortion is hard and harsh to our ear hence the difference between the two.
The valve appears warmer like a valve Marshall does compared with a transistor version.
Today, on the other hand, the top end and noise is not a problem as modern analogue tape
recorders have good signal to noise ratios and our mike preamps are also quiet yet from another
aspect it is. The top end of digital is extremely bright compared with analogue tape due to the
inherent distortion of frequencies above 7kHz created by the slow sampling frequency of 44.1kHz
which in reality produces close to a square wave above 10kHz I find it produces what I call digital
fatigue. Rupert Neve was recently reported as saying that we will need to sample at 24
bit/192kHz to equal analogue. (We will eventually) Meanwhile the warmth of the valve acts with
the harshness of digital and produces a great compromise, hence one of the reasons for the
popularity of valve mikes today.
Alternatively engineers today will put a mike through a valve preamp which is the second stage
of amplifying a mic signal. Once again it is the soft clipping of the high end that produces that
warm sound. What a lot of manufactures do today is the put a valve within a transistor circuit
thus obtaining the soft clipping of the valve with the improved signal to noise of the transistor
circuits. I've even seen an ad for a CD player that has a valve circuit in it!!
Polar Patterns
A dynamic microphone has a set sensitivity pattern called Cardiod Pattern or "heart shaped" or
"Kidney shaped" pattern and the response looks something like this.
Please note that this is not the response curve of a SM57, a SM57 is might tighter than this, it is
only a demonstration. The line through the centre of the mike is called the Axis and when
standing directly in front you are said the be On Axis as opposed to being Off Axis at the side
and rear. In this example at 0 degrees there is full sensitivity, at 90 degrees the signal is
reduced - 5db, at 120 degrees by 10, at 150 degrees by 20 db etc.
Condensor microphones have the added advantage of being able to alter their pattern from the
standard cardiod and produce either a Figure 8 pattern or an All-round pattern.
.........
FIGURE 8 and ALL-ROUND PATTERN

When using a Fig 8 mike you can place an instrument or singer on either side of the mike. With
the all-round pattern you can place anyone anywhere as the pattern picks up through 360
degrees. Incidentally the all-round pattern does not exhibit proximity effect.
Hypercardiod
Finally there is the Hypercardiod pattern. This is like a cardiod pattern but tighter.
MS Stereo
MS stereo is short for Mid Side miking. It is recognised as being the truest form of stereo
miking because it is not subject to centre lift in mono. When you join a stereo signal into mono
the instruments panned to the centre (i.e. equal left/right) lift in the balance and is referred to as
Centre Lift. MS stereo recordings don't have that tendency. You can buy MS Stereo
microphones but if you've got a cardiod quality mike and condensor that will produce a Figure 8
pattern you are in business. Set them up like this.
The signal from the Figure 8 mike will need a mike splitter that splits into two mike inputs. This
is the tricky part, To have a figure 8 mike it must be a condensor with phantom powering and if
you split it and phase reverse, it will cut off the phantom power. You can purchase a transformer
box like this:
The other way is to bring the Fig 8 mike up into a console and then take a feed from the direct
out of that channel and bring it back in via a line input on another channel and phase reverse it.
Bring up the cardiod mike and pan it centre, now take the two splits of the figure 8 mike and pan
one left and one right. Now reverse the phase of one of the splits. If you now have the cardiod
mike pointing at what you are recording and you slowly add the fig 8 mikes you will hear the
sound change from mono to wider and wider stereo as you add more of the fig 8 mike. The
Cardiod mike is called the mid mike and the fig 8 is called the side mike.
When you mono this signal the left and right signal cancel each other and you are left with the
mono centre signal which is a true mono. You can use MS Stereo for all sorts of things like
overheads on a drum kit, ambience room mikes, stereo ACC guitars and pianos etc. You can
always buy a MS Stereo mike but they are very expensive.
Proximity Effect
Anyone using microphones must understand proximity effect. When you get close to a
microphone there is a rise in the low frequencies called the proximity effect. This low end boost
can be 20+db boost at 50Hz!! A vocal mike like the Shure SM58 has a built in roll off to
compensate for this because live performers like to sing close to the mike, but if you stand back
from the mike it will start to sound thin, in other words if you want the SM58 to sound flat you
must be close to it. Most mikes will have proximity effect so a low cut filter option is often
supplied to compensate for it.
MICROPHONE PHASE RELATIONSHIPS
Before you record anything it is imperative that you check all your microphones for phase.
Two diaphragms in phase
Here the two diaphragms are moving in the

same direction so they are in phase. Imagine
them as two overhead mikes and they will
both receive the signal from the drums in the
same phase.
Two diaphragms out of phase
Here the two microphones

are pointing opposite to
each other yet their
diaphragms are receiving
the same signal. When the
left mike's diaphragm
moves in the other mike's
diaphragm moves out.
As a result the two mikes

are said to be out of phase
and a phase reversal must
be inserted or the two
microphones will cancel
each other. Officially they
should cancel totally but
they don't entirely in
practice because each has
a slightly different signal
because of it's different
position in the sound field.
It will be most noticeable
in the low frequencies so if
you top and bottom mike
a snare and don't use a
phase reversal the sound
will be thin and lack low
frequencies.
Similarly, miking toms top and bottom the bottom mike will require a phase reversal. If your
console doesn't have a phase reversal switch on it (funnily a lot don't) you should build some
phase reversal plugs of your own. This can be done by simply making a male to female mike lead
with pins 2 and 3 reversed. It's a good idea to paint them red or something so you know that
they are phase reversal cables. You can also purchase pre-made phase reversal plugs from some
retailers. Some people simply connect a male and a female cannon plug together with the leads
reversed, paint them red and insert them into the mike lead before the mike patch point.
Checking your phase

A small note here - before you start recording it is a good idea to check the phase of all your
microphones and cables. You can purchase small phase check boxes where you plug each end of
the cable into it and if all three lights light up the cable is OK. At some stage it is worthwhile
setting up all your mikes, select one mike as a reference, and getting someone in the studio to
speak into your reference mike and each mike in turn to check that each mike is in the same
phase and that all your cables are correct. You will notice immediately if one of your mikes is out
of phase.
THE MOST COMMON MICROPHONES

The famous D112 from AKG - a standard
kick drum microphone.
The classic AKG 414 EB This is a great

overhead - hihat mike (it's also a great kick
mike if you're prepared to put such an
expensive mike on the kick).
The AKG 451 is a beautiful all purpose

quality Condensor microphone. It comes
with various alternative diaphragm capsules
with different pickup patterns.
The classic Sennheiser MD421 tom

microphone. (John Laws has a gold one!! for
you OZreaders)
Sennheiser MD441 is another great snare

mike and can be used on toms.
The fantastic range of microphones from
Neumann Germany. Unfortunately they are
now so expensive that the average home
studio owner can't afford them. You can
probably buy 10 SM57/58s for one budget
Neumann!! But they do sound extremely
good and are one of the best!! If you can
afford at least one, preferably a pair, you'll
never regret it.
The classic Shure SM57. Probably the best

value microphone available. You can use it
on drums, guitars, vocals, whatever.
You can check all the mikes out at the websites of the manufacturers:
NEUMANN

SENNHEISER

AKG ACOUSTICS

SHURE

BEYER
COMPRESSOR/EXPANDERS, LIMITERS and GATES
Dynamic Range
Before we look at compressors and limiters we must understand the term Dynamic Range. The Dynamic Range of a sound is
the range between the quietest section and it's loudest section or in the case of a recorder the range between the noise floor
and the point of distortion. You know how loud a Symphony Orchestra can get yet you also know how quiet it can get. An
Orchestra has a wide Dynamic Range.
The meters above show a dynamic range of 72db. On a home cassette recorder the quiet section in this track would be below
the noise of the tape recorder and all you would hear through the quiet passage would be tape hiss. The distance from the
loudest section to the point of distortion is called the Headroom. If distortion is reached at +6db then we currently have a 4db
headroom. To reduce the dynamic range you could ride the whole track with a fader and turn it up when it's too low and pull it
back when too high or your can use a compressor.
Compressors
A Compressor can change the input signal to output signal ratio.
In the diagram above unity gain means that what you put in you get out. In the 2:1 ratio example When the signal is above the
threshold the signal output is reduced in a ratio of 2db in will give 1db out when the compression ratio is set at 2:1, so you
have saved 1 db off the top of your dynamic range and you can turn it all up by 1 db without effecting the headroom. In a more
severe case like the 20:1, which is more commonly called limiting, for a 20 db rise in signal only 1 db comes out. The
compressor and limiter can be used together in one unit where the compressor works in the 2 - 15:1 range whilst the limiter
stops the extreme transient peaks in the signal in the 15 - 20:1 ratios which is why it is often called a Peak Limiter.
In the above graph the threshold of the limiter has been raised so that the main program material will be compressed above
the threshold of compression at 2:1 and above the limiting threshold it will be 20:1. A compressor is a gain reduction device,
therefore all compressors have a make up gain control so that if you are using 3db of gain reduction you can turn the output
by that amount and still retain the same headroom.
In the diagram above the transition from unity gain to compression at the threshold of compression/limiting is gradual instead
of a straight line. This is called a Soft Knee threshold and is much smoother.
The Meter on a compressor can usually be switched to read either the input level, output level or the amount of gain reduction.
It is advisable to check that the input level is correct before you start adjusting the threshold and setting the compression ratio
etc.
The attack time determines how quickly the the compressor reacts to signals above the threshold. Signals have short sharp
peaks called Transients that can easily trigger a compressor to act. The attack time determines how long the peak should be
above the threshold before compression takes place. These short transients are important in the clarity of a sound but don't
effect the loudness of the sound. The aim of compression is to make the instrument sound louder, to squeeze the dynamic
range, therefore you may wish to lengthen the attack time and let the transients through (to be dealt with by a limiter if
necessary) and the compressor will then be working on sustained levels above the threshold.
The release time determines how quickly the compressor lets go, or restores normal gain. If the release is too fast for the
amount of gain reduction applied then the return to normal gain over and over as the signal moves above and below the
threshold can cause what is known as pumping because the gain structure is changing rapidly. It is advisable to ask the player
to play sustained notes and set the release so the change of gain is smooth. Instruments that have long sustaining notes like
bass guitars should tend to use a slower release times than sharp percussive instruments like percussion. Most of the new
generation compressors now have an Auto button that leaves it to the compressor to work it out, and they usually do it fine.
Take a look at a typical compressor and its controls:
The left section is the Noise Gate section. It has controls for the threshold at which the gate opens, the release time variable
and a switchable fast/slow attack control. The centre section is the compression section with the standard controls over
threshold, ratio, attack and release. The Peak/RMS switch determines how the compressor will track the signal i.e. its peak
content or its RMS content. The Auto button is often an option where the compressor works out the attack and release times
itself by analysing the program material. The hard/soft switch determines the Knee setting. The meter can read input or
output levels plus it can read the amount of gain reduction. Finally there is the makeup gain control (Often just labelled
output level) The link button is there if there are two compressors in the unit . Stereo Compressors have a link facility that
makes one of the two compressors a master. (Usually the left compressor). All the controls on the master effect the slave
compressor, so they both operate together. If the compressors weren't linked any strong signal on the right would be gain
reduced and the stereo image would move because centre panned instruments would vary in their left to right balance so when
compressing a stereo signal make sure the compressors are linked.
Similarly take a look at this image of a Computer program Compressor by Waves. All the controls are there.
The Electro and Warm options are computer additions not found in a stand alone analogue version. As you can see the
threshold is below the peak signal so gain reduction is taking place as indicated in the attenuation meter. The ratio is set at
2.90:1 and there has been no make up gain applied.
The attack time is set to 3.66ms and the release is at 214ms and the control on them is manual (not auto).
Expanders
The expander is a compressor in reverse. There are two types of expander. In some, signals above the threshold remain at
unity gain whereas signals below the threshold are reduced in gain, whereas in others the signal above the threshold also has
the gain increased. Therefore you can use an expander as a noise reduction unit. Set the threshold to be just below the level of
the player when playing. When the player stops the signal will fall below this threshold and the signal is reduced in gain thus
reducing the noise or spill.
The drawing above shows the different actions of compressors and expanders. The expander in the drawing is increasing gain
above the threshold and reducing gain below the threshold.
Most recording in popular music today has had heavy compression. Recording are loud and in your face! As well as most of the
components of a track being individually compressed the whole mix overall has been compressed and limited before going to
CD. I don't think that's a bad thing.
Limiters
A limiter is just a severe compressor where the compression ratios are high. On some units like the DBX 160 and the Aleisis
compressors an additional Peak limiter control with a LED that flashes is supplied, but units like the Aphex Dominator are pure
limiters and are very sophisticated in how they attack and control peaks and you can get some pretty hot "brick wall" mixes
through them.
De-Esser
A De-esser is a frequency selective compressor/limiter that compresses only at a predetermined frequency. If set to the
frequencies around the sibilance area of a vocal (4kHz - 8Khz, it varies between men and women,) the vocal will be
compressed only at those frequencies which will reduce the sibilance. Sibilance is the peaks of high frequencies created by 'S',
'T', 'C's etc.
The new generation

The new generation compressors, expanders gates etc. in the new computer programs are worth a mention here. These
compressors have one outstanding advantage over the stand alone compressor. They can read the signal ahead of time by
extracting the signal from the hard disk ahead of time, analysing it and then outputting it in real-time. They know what is going
to happen next which gives them a distinct advantage in maintaining smooth control over the signal.
Noise Gates
Noise gates are units that let a signal pass if it's above a certain level and shut it down if it's below that threshold.
The diagram above shows how a gate works on level. When the signal falls below the threshold the gate reduces the level to
the specified reduction level. The attack time here determines how quickly the gate will open and the release time determines
how fast it will close. Some gates have a Hold function that allows you to tell a gate to hold open for a set time once it is open
and then the release time can take over and close the gate. This facility can stop the gate opening and closing quickly due to
peaks. It can also be used as an effect, especially if it is put over the return from a reverb unit. If you have some reverb on say
a snare, and you put a gate over the reverb return signal, you can get the hold function to hold the reverb open for a period set
by the hold function and then to quickly close it by using a fast release. This effect is called Gated Reverb and is now a
standard program in most reverb units.
A gate can also be set to be triggered by something else via a side chain. For example, if you put a gate over a room
ambience mike you could use the snare mike to trigger it to open when the snare was hit and to close when the snare stopped.
This is called Gated Ambience. Another effect is to put a hihat feel into the side chain and modulate the gate to open and
close on a synth sound. The effect is a modulating synth with the attack and release times controlling the modulation.
Gates can also so linked so that one controls the other and when one opens the other opens as well. (Like the compressor)
This is used in stereo gate situations like over stereo toms.
TRACKING & BALANCE
Before we start on this subject let me tell you a story: I was working at a studio called the Music
Farm and there was an English engineer working there called David Tickle. He went on to
produce Tony Childs . One evening a muso asked him "How do you become a good engineer?" -
to which he replied:
"You go into the studio and ask the drummer to hit the kick drum and you grimace. You put a
mike on it and go into the control room and ask the drummer to hit it again. Again you grimace.
If you can make the kick drum sound like the one you hear in your head - you are a good
engineer! It doesn't matter how you do it but a good engineer can"
I can't really tell you how to do it either but this page will give you a starting point.
METERS
Before we do anything we must talk about meters and what they read.
Take a look at this image of a drum sample: Click image to hear sample.
You will notice from the shapes that the waveform rises sharply at the start and then tapers off
as it fades. This initial rise to the highest point at the front is called a transient and the highest
point is called the peak of the waveform.
There are two ways to meter this waveform:
1. The VU or Volume Unit meter

2. The Peak Meter.
The VU Meter
These meters read volume units , hence VU, and do not give an idea of the peak material. They
indicate the RMS (root mean square) value. Some VUs might have a peak LED in the upper
corner. NOTE: If you peaked this meter to Zero on the drum waveform the actual peak transient
would in fact be 10 - 20db or so higher than the meter was reading and can give you a wrong
impression of the actual peak content. If you peaked this meter to about what it's reading now
the peaks would be in the red section. So if you are working with this type of meter don't peak
transient sounding instruments like kick, snare, percussion, toms etc. to zero because you will
distort all the transients. In the late seventies the console manufacturers started adding peak
meters. So if you are into retro gear with VU meters watch your transient levels!!
The technique of pushing magnetic tape by saturating the tape with high level was really just a
method of compressing the transients by using the tape as a compressor thus the new trendy
term Tape Saturation. Actually the tape played back the transients but only for a few days -
after a week or so the transients were well and truly eliminated and engineers complained that it
sounded great when we recorded it but somehow the tape has lost it!!! ( the tape also suffered
from extreme print through which is a condition where the magnetic flux on a tape is so high that
it printed through onto the tape layer adjacent to it when it was rewound and was left sitting on
a shelf.)
The Peak Meter
These are the ones used today. They often incorporate a peak hold function that lets you see
what your peaks are doing and are therefore easier to use and are a truer indication of what is
actually happening. Don't try saturating your tape with one of these meters as digital doesn't
saturate - it just distorts severely!!
TRACKING
Before we can balance the mikes we must bring them up into the console and set their levels to
tape. How the mikes are tracked to tape is really dependent on how many tracks you can afford
for the drums. I usually allow 7 or 8 tracks for drums. This might seem extravagant to some and
not enough to others. I'm conditioned to the old 24 tracks of recording and therefore think in
those terms but nowadays with multitrack digital recorders and multitrack computers anything
goes but with 8 tracks you can assign your drums as:
1.
Kick
2.
Snare
3.
Hats
4.
Tom 1
5.
Tom 2
6.
Tom 3
7.
Overhead left
8.
Overhead Right
Alternatively you can premix your toms and put the snare under mike on it's own track or you
can mix your toms and put the bass on the 8th track. (Or in my case the 4th track because I
think in pairs!!) so it goes:
1.
Kick
2.
Snare
3.
Hihat
4.
Bass
5.
Toms L
6.
Toms R
7.
O/Head L
8.
O/Head R
This setup will leave you 16 tracks for the rest of the instruments and vocals etc. I'm not going
to go into how to assign your tracks, I assume you will know this.
I would make a special note here:
You must always be aware of the gain structure surrounding a console. Look at the gain
structure and how it works.
The microphone signal is amplified by 40 - 50 db by the mike preamp to bring it up to the

operating level of the rest of the console. Any EQ is additional gain or reduction. Lets say you
open the channel fader 1/4 the way up and then increase the gain by increasing the mic preamp
gain. It is very likely that the higher output from the mic preamp will distort when it enters the
input of the next stage, the EQ. You will then EQ it adding more gain and more distortion.
Similarly with the group output fader if you have one. If you run this fader low all the previous
stages will be driven to overload.
Always start with your channel and group faders at Zero

and adjust your gain at the mic preamp
You'd be amazed at the number of times I glance at a PA console and see the group faders at
1/4 and the channel faders at 1/4 and the mic preamps turned up!! and a worried engineer
wondering why the PA sounds awful.
EQUALISATION
EQ can enhance a sound but it can't fix it. Too many engineers try to EQ their way out of trouble
and believe me - it doesn't work. Adding highs might make it sound better but if you play it
through a speaker that doesn't have the great highs your studio speakers have it will still sound
awful so you really must aim to get the sound correct its source.
OK so we open up the kick mike and ask the drummer to hit the kick. Bring up the preamp level
so that the kick peaks to just short of Zero in your meter. (I guarantee that the drummer will hit
it louder during the track) Now if you are going to EQ it you can do it now or later. I prefer to do
a basic EQ now because it makes the kit sound better while you're tracking the other instruments
etc. but don't make it extreme.
The traditional EQ for a kick drum is to add some mids (say +4db) at around 3.5K to bring out
the attack, and pull out some low mids ( -4db) around 300Hz. (The fundamental frequency of a
22" Kick drum!) I wouldn't advise adding any low end at this stage but you might be tempted to
add some 80Hz, say +3db.
If it still sounds awful you must fix it by either:
1. Using a different mike.

2. Changing the dampening.
3. Moving the mike
4. Retuning the Drum.
If none of these work you've got problems.
Moving on: now let's open the snare mike and as with the kick bring your fader to zero and
increase the level of the mic gain. Now you have the option of adding the under mike if you've
put one up. (Remember the phase reversal) By balancing these together you can vary the
amount of the metal snarey sound in the mix.
The standard EQ for a snare drum is basically add some tops around 7Khz. Once again if it
sounds awful you must change it at the source.
Now the overheads:
Ask the drummer to hit the snare drum and then open the overhead mikes and balance them so
that the snare drum sounds centre. Then ask the drummer to play the whole kit with the toms
and the cymbals and listen to the stereo spread listening for the balance of the cymbals. If say
the left crash is too low go into the studio and lower the overhead over that crash. Repeat the
process starting with the snare again. What you're doing here is making sure that the overhead
sound is a true stereo image of the kit. You will hear the toms spread left to right, the kick will be
centre. as will the snare. The Hihat will be off to 1/2 right. If you wish to EQ it maybe add some
10Khz tops to it but if your mikes are good you shouldn't need anything else.
Now you go through the toms one by one. Basically toms need a bit of tops and they often need
some upper low mids(400Hz) taken out. You may also wish to add some lows like +4db at 80Hz.
I suggest you balance the floor tom higher than the first rack tom. Drummers invariably loose
energy as they go around the toms - sorry guys. Now that you've got all the toms peaking to
zero in the meters open up the overheads. Suddenly they will start to come to life as the
overheads will give them space and transient attack.
Finally the HiHats. Remember these are extremely transient so watch the level carefully. For EQ I
usually roll off a lot of the bottom end quite severely as well as adding highs at around 10Khz to
give them some sizzle.
Gates: Gates can be used when you record drums and the most common method is to gate the
kick and the snare and if you have enough gates to gate the toms individually. Personally I don't
gate at the record stage mainly because if you haven't set them up correctly you can't redo it
later because you've already lost the signal. I recommend that you gate at the mix stage where
you have more time and freedom to get the gate tuned exactly for each track. Quite often you
will find that the drum sound for one track works great with gates yet another track sounds
better ungated. For toms I prefer to automate the tom tracks with mutes instead of using gates.
For more on the operation of gates refer to the Effect Pages.
Reverberation. I don't advise adding reverb to your drum tracks at the record stage however
you can add reverb to your monitoring by adding reverb to the monitoring controls of your drum
tracks, This allows you to have reverb on your monitors but the reverb is not on tape. N.B. you
can't undo reverb!!
RECORDERS
The recorders are data storage systems and fall into three categories:
Analogue Tape

Digital Tape

Digital Hard Disk.
Analogue Recorders
Despite the onslaught of the digital recorders over the past decade the analogue recorders are
still hanging in there primarily because engineers still believe that the analogue sound has not
been surpassed by the digital medium, and quite rightly so. The top of the line 2" analogue
recorders are still being used and sold but primarily by the perfectionists and dedicated
audiophiles who will probably have a digital system with the analogue being used for bass,
electric guitars and drums. Vocals and the rest being covered by digital. Some engineers still
insist on mixing down to 1/2" stereo analogue masters and some mastering studios actually
transfer their digital masters via an analogue recorder to soften the harshness of the digital top
end etc.
Meanwhile the home recording enthusiast is very likely to have a 1/2" sixteen track or a 1"
twenty four track etc. The important aspect of the analogue recorders is their need for regular
maintenance and servicing. If you are an owner of a analogue recorder you must have the
alignment tape required for correct alignment of the transport and electronics. This is a tape
with pre-recorded signals that determine:
Head alignment

Transport alignment

Frequency response and level structure
So let's look at the Head alignment factors:

The above are the variables in head alignment in an 8 track analogue recorder. The head
alignment is executed in three dimensions through the variables:
Azimuth
Zenith
Wrap
Skew
All variables are really self-explanatory from the diagram and the head block on which the heads
sit has the appropriate adjustment screws to line the head up. What is required is the correct
alignment tape for the tape format and speed. These tapes can be purchased from their
manufacturers.
Before you let your precious alignment tape near your recorder you must degauss it:
Degaussing
As the magnetised tape travels through the tape path the magnetic flux on the tape
progressively is transferred to the metal parts such as the tape guides and heads. This build-up
if left unattended will induce magnetism onto these parts and they will progressively erase the
high frequencies on the tape as it passes. Therefore regular degaussing of these parts is a
necessary maintenance procedure. A degaussing tool is required for this operation.
It is common practice to wrap a layer of insulation tape around the head of the degausser so
that the metal doesn't damage your heads if you accidentally touch them. The technique here is
to first turn off the tape recorder. If you don't you'll blow out all your circuits!! Then switch on
the degausser with the head 1m(3ft) away from the recorder. Slowly bring the degausser head
up to the play head and move the degausser slowly down and back up the head then slowly
draw it away around a foot, then do the same to the next head and the next if there is one.
Then using the same technique do the tape guides, rollers etc. Finally draw the degausser away
from the machine around 1m(1ft) and switch the degausser off. Your machine is now
degaussed.
Cleanliness.
The heads on a recorder acquire a build up of tape oxide after constant use so it is necessary to
clean the heads and the tape guides regularly, like twice a day. Cotton buds and Isopropyl
Alcohol (available from most chemists and drug stores) is the most common method. The main
thing is to regularly check your heads to make sure that there is not too much build up. If there
is you should recheck your head alignment or transport alignment. Too much tension on the
tape can cause oxide build-up because of the increased pressure on the head. Another cause is
worn heads so if you suspect worn heads get your machine checked by an expert. This is an
exaggerated version of what happens:
Alignment Tapes
Alignment tapes come in three types according to their calibration. They can be either CCIR,
NAB or IEC alignment. CCIR is recognised as being the British or European standard whereas
the US is NAB. IEC only applies to the speed of 30ips. When you purchase an alignment tape
you must know what your machine is calibrated to. Secondly you must know what speed your
machine uses and if more that one speed is available you will need to purchase an alignment
tape for each speed. These standards apply to the pre equalisation curves that a recorders have.
Basically they boost the high frequency onto the tape and then reduce it back again on playback.
This improves the signal to noise ratio. Each of the types, CCIR, NAB or IEC, have different EQ
curves.
Secondly alignment tapes are recorded at a specific flux level measured in nanowebers. The
first ampex alignment tapes were at 185 nw which meant that if you aligned your playback head
to zero and then your record head to zero you would be recording a flux density of 185
nanowebers. Tapes then came out at 250 and then 300. You must check what level your
alignment tape is at and what the capability of your machine is. The new tapes are capable of
recording a higher flux than the old and the level has been going up an up over the years as
tape manufacturing improves.
Bias
Simply the bias is a high frequency (typically between 150kHz and 280kHz depending on
recorder type and model) signal that is added to the record signal to compensate for the
irregularities in the ability of tape to hold flux uniformly. It is an adjustment in the record side
calibration.
Alignment
You will need an oscillator capable of producing frequencies from 50Hz to 16kHZ for this
procedure. Some consoles have a tone generator built in. Your machine will also have 1,2 or 3
cards for the record, play and bias adjustment.
OK so the machine has been cleaned and degaussed, now you can put your precious alignment
tape on the machine.
The first thing to do is to check the tape path and confirm that the tape moves freely within the
guides etc. Now you are ready to check the playback alignment. You must first check the
azimuth. Take a playback signal from your two outer tracks, 1 - 8, 1 - 16, 1 - 24, and bring
them up on your console at equal level panned centre. Now playback the 10 or 16kHz section on
the tape and adjust the azimuth on the head so you get the highest reading. This assures that
the head azimuth is correct and that the phase relationship between your outer tracks is the
same.
Now play back the 1kHz tone and adjust the playback level to zero. If you wish to add more flux
to tape because you have the latest tape you may wish to put 3db more level onto tape. In this
case you should line up the playback reference to -3db. Now play back 10kHz and adjust the
playback highEQ on your record card to read as close to zero as you can get or -3db if adding
higher flux. Now playback the low frequency tones like 100Hz and line these up to zero (or -3db)
or as close as you can get. Now recheck your 1kHz tone and do all the other tracks the same.
You are now ready to align the record head. Remove the alignment tape and store it in a safe
place away from anything magnetic like speakers etc.
Now put on a new roll of tape which you have labelled Record Test Tape. Keep this tape with
your alignment tape for further alignment sessions. Put the tape on and put all the tracks into
record. Roll tape and hit record and select playback on all of the tracks. You must now adjust the
record level so that you get around 0db level played back. Now you are ready to check the Bias
level. The adjustment for this will be on your bias card. Record a 10kHz tone and switch the
machine to playback and adjust the bias level control. You will notice that as the signal rises it
reaches a peak and starts to drop again. If not you must find this area. Line it up to the peak
and then keep increasing it until it drops by 3db. This is called overbiasing by 3db. Do this for
each track.
Now you must check the azimuth of the record head which is done by recording 10kHz onto your
two outer tracks and playing them back through the console like we did before adjusting the
azimuth so that you get the highest reading.
Now that the bias and azimuth are calibrated you can start the frequency response calibration
starting once again with 1kHz followed by the high and low frequencies and adjusting the record
card controls. A good machine should give you a flat response + or - 3db from 50Hz to 15Khz.
Digital Recorders
If you have moved to the new digital recorders and have just read the previous rave you must
be breathing a sigh of relief as the digital recorders don't have frequency response or bias and
azimuth adjustment. All they do is record 0s and 1s, albeit really fast at 48kHz/sec.
So lets start with some basic knowledge of digital sound. What does it mean when they say that
a track is recorded in 16bit digital at 44.1K.
Bit Rate
Digital sound is made up of words of 0's and 1's and 00, 11, 01, 10 are the four possibilities in
a two bit word. A three bit word can be made up with 000, 111, 001, 010, 100, 101, 011, 110,
which means there are eight possibilities. You see - 2 bit gives 4, 3bit gives 8, 4 bit gives 16, 5
bit gives 32, and so on. Now if we were to use the bit words to express volume with a four bit
word we could give 16 different values for volume. So the higher the bit rate the more accurate
the resolution be it volume, digital pictures or digital sound. So 24 bit digital sound has more
resolution and accuracy than 16bit digital sound.
Sampling Rate
Digital sound is produced by sampling a sound (or should I say the electrical version of it) in real
time and expressing it in bit words. Once you start sampling or recording digital sound a clock
starts and progressive samples of what the sound is are taken. The rate at which the samples
are taken is called the sampling rate.
The drawing above shows a wave of a sound being sampled. If the time in the drawing is 1
second, then there are 6 samples (the last one is the first in the next second) of the sound in
one second or a sampling rate of 7. So obviously the higher the sample rate the more accurate
the resolution. So when we say that the sound is 16bit, 44.1Khz it means that the sound is
being sampled at 44.1 thousand times a second and it is being measures with 16 bit accuracy.
In the above waveform the sampling volume levels given would be 0,2,2,0,-2,-2, Not a very
accurate version of a simple waveform. But 44.1kHz, now that's fast, or is it? Lets look at sound
in seconds.
In this chart you can see the relationship between the sampling rate and the waveforms it's
sampling. 1kHz will have 44.1 samples taken of each of it's waveform as its oscillating at 10,000
waveforms a second. 100Hz will have 441 samples taken of each of its waveforms. But 10kHz
will have 4.41 samples taken of each of it's waveforms. Now look at the first waveform we drew.
In that drawing we took 6 samples of the waveform and got an amplitude reading saying
0,2,2,0,2,2. imagine how inaccurate 4.41 samples are of a complex waveform. That is why
digital high frequencies sound harsh!! The industry has constantly denied this factor and even
gone to the extent of saying the hear can't distinguish between a square wave and a sine wave
above 7kHz. Pigs Bum.
At a sampling rate of 96kHz you get 9.6 samples of a 10kHz wave and believe me, you can hear
it.
In an article by Rupert Neve, I read recently, he said that we should aim for 24bit resolution and
192kHz sampling rate if we want to equal the quality of high quality analogue recording. We will
get there. DVD is already up to 24 bit 96kHz sampling so we are on the way. But if your 16bit,
44.1kHz CD sounds bright, consider what makes it bright and you will see that it's a false bright
created by the high frequencies sounding like square waves!!
Why 44.1kHz Sampling Rate?

Why not 44, or a nice round number like 50. When the first engineers were inventing digital
sound they had worked out the on/off, 0/1, idea and needed a way to represent it. The idea
came to use white dots on a TV screen where a white dot was on and a black dot was off. Neat.
So you record it like a video picture on a video recorder. That was fine, but the engineers had
been caught out before. What about PAL (the European video standard) and NTSC? (the
American and Japanese standard.) They weren't going to get caught up in that again, no way, so
they configured a number that was compatible between the 528 line NTSC and 625line PAL and
the number was 44.1kHz. Just a piece of useless info you might want one day!
What you can see from the above is how the digital recorders were developed. They were Beta
Video Recorders with an external processor and digital audio had arrived. The beta video
became the DAT and the DAT became the ADat and the D88 and they are all basically video
recorder decks. The ADat used a SuperVHS deck while the D88 used a High8 deck. The basic
SuperVHS deck was pretty awful and the ADat of today is a completely rebuilt deck. It's a shame
the world chose to make the VHS deck the standard because the Beta Decks were far superior.
Market forces don't always give the best outcome. Did you know that when an ADat or D88
records on a new track it plays the bit stream off the tape , mixes in the new track, and records
it again. Now that's worth thinking about.
Hard Disk Recorders

Now we are entering a new era in recording with the advent of the computer. Now hard disk
drives can store gigabytes of information and retrieve it at high speed. The Pentium 11, 350mHz
computer I am writing this on will play back around 13 - 16 tracks of stereo digital audio in real-
time at 16 bit 44.1kHz. It will also process it in real-time so I can add EQ, Compression, Reverb
and Effects, I can cut and paste it, I can timestretch and pitch change it, I can even put it in
tune. It cost under $US-2,000 and will do my Internet, keep my tax records and play games as
well. The hard drive can be backed up onto CD with my CD Burner. At the time of writing the
new 1gig processors are being released and my local dealer is offering 27gig hard drives for
around $300. As the telephone lines get better and the Internet gets faster and faster I will soon
be able to play with a guitarist in London , a Bass player in the US, a drummer in Africa in real-
time wow! where is it going to go?? Anyway back to reality.
The first units were stand alone units that had hard drives built in and all the companies were
there with some kind of model. Meanwhile the programmers were busily rewriting their
computer sequencer programs (Cubase and Notator) for the Macintosh platform and Cakewalk
was writing theirs for Windows. Dididesign was developing ProTools and computer based hard
disk recording was born. Now they all have computer based hard disk recording programs and
due to market forces again they were all rewritten for Windows.
I won't go into the pros and cons of all the current programs, there are lots of qualified people
able to discuss them all and I suggest you read the reviews suffice it to say that this is where
the next generation of studios will be heading. In fact I would suggest that most of you reading
this will have some kind of computerised hard disk recording already and are as excited about
the future of this technology as I am. They are already offering 24bit 96kHz sampling and the
new generation effects units are getting better an better. For the price of an ADat you can get a
whole studio!!!!!
I say no more
CONSOLES
Recording Consoles are simply a set of component parts all put into one box which can be configured into two modes
Record/Overdub
Mixdown
RECORD/OVERDUB MODE
The components in the Record Mode are configured thus:
The recorder and the speaker and power amp are not part of the console but are part of the chain. So let's go
through each component and look at what it contributes to the chain.
The Microphone Preamp

The microphone preamp is possibly one of the most important component in the whole recording chain apart from the
microphone itself. Between them, these two units bring the minuscule voltage coming from the diaphragm up to the
1+volts that we work with. From that point on everything is operating at normal levels and the signal can be
recorded, EQ'd etc. without problems with noise and distortion. The typical mike preamp can have the following
controls.
I said "can have" because not all do. First there is the mike input level knob. This is the gain control for the
preamp.
A special note here - Before you start to set the mike input level you must set the console up for unity gain. This
involves first setting the console output faders to Zero, then the channel fader to zero. If you are going out a group
put that fader to zero. This first step is vitally important because a console is capable of increased noise and
distortion if not setup with correct gain structures. If you have a little Mackie or something which doesn't have a
separate control over monitoring turn your amp and speakers down. Basically if you run the output faders low you
have to get the gain from somewhere so you turn up the mike preamp which is capable of adding noise and
distortion.
If you find that you are fully counterclockwise and still have too much signal you must insert the Pad. The pad drops
the level by 10 - 20db (depending in the console) and stops the preamp from overloading. Some consoles will include
a phase reversal button which is a very handy option to have. There is also a button for Phantom Power which
will supply power to your mikes if required. (See microphones) It often comes as a single on/off switch on the
rear of the console. There is often an optional line input knob with an associated mike/line switch. This allows you to
trim the level of the line inputs. Finally you will probably find a mike/line button that allows you to adjust the level of
the line input individually. The flip button is only on certain consoles. It swaps the two main faders over (line and
monitor), More about this option later when we look at monitoring..
The Equaliser
The equaliser is the next stage within a console. This are the controls on a quality console equaliser.
This equaliser is a true 4 band, parametric equaliser. It incorporates four separate bands with the low-mid and
high-mid bands fully frequency sweepable with variable Q. Go to the full page on equalisation for more about EQ.
Insert Send/Return
The insert send/return appears as an output and return on the rear of the console typically as a stereo jack with the
tip being the send and the ring the return and the sleeve as a common ground. This facility allows you to insert a
compressor or effect unit into the signal chain.
The Auxiliary Sends
The auxiliary sends are split feeds of the track signal that can be sent to other units such as reverb and effects
units. It can also send a separate mix to the headphones for the musicians. The send knobs are self explanatory,
the pre/post switch determines whether the signal it sends comes from before or after (pre/post) the fader. If it is
set to post fader as you turn the fader up and down the signal going out the aux send follows and also goes up and
down. The pre fader position means that changes you make to the fader won't effect the aux send output and its
level will remain constant. This is important if you are sending a mix to the musicians which is normally done with pre
fader sends so that your mucking around with the faders doesn't change their balance. The monitor button will switch
the input to Aux 1 and Aux 2 from the channel signal to the monitor signal on that module.
The Channel Fader

The channel fader gives you control over the level of the channel. As mentioned earlier In recording it should be used
as near to 0 as possible. It also becomes your mix fader.
The Routing Switcher/Output Groups

The output group is the amplifier that joins all the separate channels together and routes them to their destination.
The Routing Selector allows you to send the signal from a channel to any of your subgroup outputs. I say sub
groups because you will also have a stereo main output. The Pan control allows a stereo output between two
selected channels typically 1 & 2 , 3 & 4 etc. The pan control will have a pan additional pan on/off switch. The
direct out button allows you to send the channel directly out of the console via a plug on the back of the console
without going through a group amplifier. The stereo output selector sends the signal directly to the master output
and is used when in mix mode.
The Recorder
The signal from the group outputs will now appear at the recorder inputs and when monitoring through the recorder
or playing back, it's output will appear back at the console at either the line input or more typically the tape return
input depending on console type.
Monitoring
You must be wondering how you are going to hear what you are recording. The key line here is "what you are
recording". You must follow the sound through the above outputs etc. so that when you monitor the sound it is "what
you are recording". Now we have all the channels going to the recorder and back. You must now listen to the signal
through the Monitor control. This is usually a knob (or fader on the more expensive consoles) below the auxiliary
send section and has a pan control associated with it so that each monitoring return can be panned left or right in
the monitor speakers as the output of the monitoring mixer goes directly to the monitoring output of the console
and to the speakers. You can change the balance, switch instruments on and off in the monitoring section without
affecting the signal you are recording. The monitoring section also has auxiliary sends or it has access to them via the
monitor switch in the auxiliary send section as mentioned earlier. This is where you should send your
headphone sends for the musicians. You must also send them prefader mixes so your changes doesn't effect their
balance. I have emphasised this because it's very important:

If you have added effects to an instrument and inserted it between the console and the recorder the musician
will not hear it if you have sent his headphone send directly from the input channel aux send.

When you have recorded a take you can immediately play it back with the same balance you had when you
recorded it and the balance in the headphones doesn't change for the musicians.

Also if you sent their feed from the channel they will hear nothing in the headphones when you play the track
back.
The Monitoring Selector Section

The monitoring output goes to the power amplifier and thus to the speakers. Some better consoles have a monitoring
selector section where you can control what signal goes to the speakers.
There are two outputs from the monitoring switcher, one goes to the studio speakers (if you have them) and the
other to your control room speakers. The selection buttons allow you to select the source for your speakers The
control Mix selection puts the . The mono button allows you to check the signal in mono and the A/B button allows
you to have two sets of monitor speakers (main and nearfield) and switch between them.
Auxiliary Returns
The aux returns (or effects returns) are the return channels for all your effect units like reverbs etc.
The aux returns are where you bring the returns from your effect units back into the console. They are usually
supplied with a routing selection output just like the group outputs so they can be sent to the group outputs as well
as the master outputs. If sent to the group outputs the effects can be sent to the recorder with the original signal
from the channel and recorded.
Overdubbing
In the overdubbing situation nothing need change as the console is already in that mode. Tracks from the recorder
are played back through the monitoring mixer and new instruments processed through the channels and group
outputs to the recorder. Please make sure you send the musician a good headphone balance, it makes it so much
easier for them to play well.
MIXDOWN MODE
To mixdown your creation the console needs to be reconfigured thus.
You will note that what was the mike channel has now become the tape return channel and follows the same route
through the equaliser, inserts, fader and aux sends to the routing switcher. Here it can either go direct to the master
output or several channels can be grouped together via the output group and then on to the master output. The aux
returns are as before. The signal now goes through the DAT and back to the monitor output and to the speakers.
Once again we have a "what you are recording" situation where you are listening to what is going on tape.
Automation
I have included an extra in the mixdown stage called automation. Console automation makes use of a unit called a
voltage controlled amplifier or VCA. An amplifier is normally in full gain mode and you change the gain of a signal
by putting more or less into it. A VCA on the other hand is an amplifier where you can adjust its gain via a change in
an external voltage. In an automated console the amplifier at the fader stage is a VCA and the fader adjusts the
voltage of the amplifier thus its gain. When you turn the fader down you decrease the voltage and visa versa when
you increase the gain. There are three important advantages here:
The fader doesn't carry the audio signal only the voltage to the VCA therefore if its cheap or dirty the audio
signal is not effected.

The variation in this voltage can be read and recorded externally and played back.

When you turn the voltage off you in fact mute the VCA so you can also automate mutes.
The variations in the voltage during a mix are recorded as data in the automation module and when the mix is played
back the VCA changes in time with the track and follows your changes. These fader changes (and mute changes) can
be altered progressively until you are happy with the mix. I Love Automation!!! For the automation to stay in sync
with the track it will need a timebase reference which is usually Time Code from the recorder.
Some companies like Mackie produce an automation package that you can plug into any console so long as you have
insert send/return plugs on the back. Here the VCA becomes like an external effect and is plugged into the channel
via the Inserts. You place all your faders to Zero, turn on all your mutes and proceed to set your fader moves using
the supplied remote control. All the changes in the VCAs are recorded within the unit and can be played back. I've
done heaps of mixes using this system and I highly recommend it as a simple, cheap, and reliable automation
system. (Hey I'm not sponsored by anyone so I can say what I think!)
Summary
We are now entering an era where the console is coming back but in a different form. Companies like DigiDesign with
ProTools and Yamaha with the O2R are producing consoles designed to be a user interface to a computer based hard
disk recording system where there is a friendly 'hands on' interface to a sophisticated hard disk recorder setup. I
dream of the Virtual Reality studio where the home recording artist can put on a set of Virtual Reality glasses and be
in any control room they like with touch sensitive virtual consoles and effects. Your friends can be included with
selectable identities and even though you are only in your 8' x 10' bedroom in virtual reality you are in the control
room of a major studio. Any smart software writers out there?
EQUALISATION
Whilst compression effects dynamic range, equalisation (EQ) controls frequency range. The frequency range of
sound is shown in the chart below and is divided up into four bands.
The scale along the bottom of the chart shows the frequencies from 16Hz to above 16kHz. When engineers talk about
the high mids they are referring to the frequency range from 1kHz to 8Khz, roughly.
The following drawing shows a typical EQ Peak Curve based around a Centre Frequency.
The centre frequency is around 750Hz and the Gain Increase (boost or cut) is around 18db. The Q Factor is the
width of the frequencies effected by the boost and is measured in octaves. A high Q is narrow and a low Q is wide.
The following is is a drawing of a Shelf Curve where the frequencies above or below the centre frequency are all
boosted or cut.
There are two kinds of equalisers, Parametric and Graphic and each can control a number of bands.
The Graphic Equaliser

Here is a drawing of a typical 10 band graphic equaliser.
You will note that there are slider controls for each frequency and the scale along the base shows which frequency.
The scale along the top states how many db change has been made at each frequency and it can be positive or
negative (boost or cut). A typical graphic equaliser does not have any controls over the Q factor of each boost, it is
normally pre-set.
The Parametric Equaliser

For an equaliser to be called a parametric equaliser it must have a variable Q factor and a variable centre frequency.
Below is an example of a parametric equaliser:
The left unit is a typical high end console analogue equaliser whereas the right one is a new generation computer
program digital ones. The left one has a switchable peak/shelf High frequency control. It has two sweepable mid bands
with variable Q and a peak/shelf low frequency control. The computer version has 4 Bands each with it's own centre
frequency, Q width and gain. The resultant EQ curve is displayed as well. (It's a digital EQ) The mid bands of the
analogue version are usually divided into two sweepable bands the the low - mid covering 100Hz - 4Khz with the other
covering 600Hz - 15Khz (typically - it varies from console to console) You will note that the digital unit is sweepable
from 20Hz to 20KHz in all bands.
MIXING
Without doubt the hardest part of recording is mixing, yet it is also the most enjoyable as this is
when everything starts to come together and all the hard work justifies itself. A good mixer
paints a picture in sound that attracts the listener and conveys the song clearly and simply. I
could sum up a good recording as a series of priorities which are:
The song
The singer
The "feel" or "groove"
The "fiddly bits"
The song is set from the start and a good producer will have chosen a song that has 'something
to say' and a good mixer will convey that something to the listener.
The singer is the next most important aspect and a good mixer will allow the singer to be heard
and the lyrics to be conveyed clearly but with style. There is nothing more annoying than
hearing a track and not being able to distinguish the lyric amongst a babble of instrumentation.
Fortunately you don't hear recordings like that on commercial radio as they just don't get a look
in. Engineers are often guilty of cluttering up tracks with all sorts of tricks and garbage that
distract from the song and the singer because they know the song so well after days in the
studio that they think everyone hears it like they do. If the track is to have a chance of
commercial success it must be understandable from the first hearing. Always underestimate
the ability of the listener as they are not professional listeners like you.
The "feel" or "groove" is what catches the listeners attention initially and sets up the mood
and emotion of the track. This is created by careful balancing of the rhythmic aspects of the
track be it drums, percussion or a great guitar groove.
Finally there is the "fiddly bits" as I call them; they are the musical phrases linking lyrics,
joining verses to choruses and filling solo sections etc. that are created by the guitar licks, the
piano fills, the answering vocal phases etc.
So where to start?
Monitoring Speakers
Monitoring speakers come in two types. Nearfield and Main. I like to use both. I work
primarily on the nearfield to establish my balances etc. and then every now and then I will
switch it up to the big speakers as they give a better idea of the low frequency balance, plus it
sounds good eh! (I was a Yamaha NS10 freak for years but now I'm totally sold on the Event
20/20. Well done Event!) To me a big speaker system is like a magnifying glass, it blows the
sound up and you can hear more but for a big system to be really good you have to flush mount
them and have good speakers and a good amplifier system. Can I say here that I don't like
equalised speaker systems. If they don't sound good flat, get another speaker!!
Level Structure
The first important procedure is to setup your console for mixing. The first requirement is to
setup your levels to and from your master recorder, usually a DAT. If your console has an
oscillator send tone to the DAT and balance left and right channels. Then check that the return
to your console, which is what you'll monitor, is balanced correctly left and right. At this stage it
is also recommended that you insert your master compressor either in the master stereo output
inserts or inline between the console and the DAT and line up correct left/right balance. This
procedure is very important as it effects your level structure from then on and if you don't do it
now you can end up with your levels all over the shop later.
Aux Sends and Returns

Next you must establish your auxiliary sends and returns.
One of the best ways to get perspective and separation within your mix is to what I refer to as
"putting everyone in their own space". You can achieve this through the use of reverb and
effects. I like to have one reverb unit dedicated to the drums. No other instruments are sent to
this effect, only the drums which will put them in their space. The choice of reverb for drums
depends entirely on the track but I start by putting reverb on the snare and going through the
presets to find the one that works best for the track. I find it usually ends up with a bright
reverb of shortish length around 1 - 1.2sec reverb time.
Note: A very fine producer in OZ was once quoted as saying "Give me a studio with 10
Midiverbs over a studio with one Lexicon 224XL" We all know what fantastic units the Lexicons
are but if it's all you've got you are limited to only one perspective.
Next I'll dedicate a reverb unit to act as my overall reverb effect. I look for the best (not
necessarily most expensive) unit in the studio for this will be my master reverb for vocals etc.
In the example above there are 6 sends with 5 & 6 being an option over 3 & 4. I therefore like
to use 1 for my drums and 2 for my master verb. Then I can assign the others for effects. I do
this so that I can always add master reverb as well as effects if necessary and if I had used say
3, I couldn't put master verb on channels where the effect was assigned to 5. Should I use a
stereo or mono send to the effects?? To be perfectly honest I don't think it matters. Most of
the stereo input reverb units I find have a mock stereo input, not a true stereo. If you use two
sends it really doesn't make a difference unless you are working with the more expensive units
like the aforementioned Lexicon, and even then I question the validity of two inputs especially if
you are limited in the number of sends.
I then assign the sends 3 - 6 to additional effects like delay, pitch change etc. to act as
perspective enhancers. When establishing delays I set them to the track tempo. See Tempo
Chart. The idea is to add these perspective effects so you only just hear them when in solo
and they appear to disappear when mixed into the track. Bob Clearmountain - the world famous
mixer - always had two delays going, one on eighths and the other on 16ths. It puts an air
around instruments and if mixed in correctly you won't actually hear them, just sense them.
Pitch change is another effect to consider with say the left channel set to -.008 cents and the
right to +.008 cents. This effect is great on harmony vocals and it puts them in a different space
form the lead vocal. Finally a soft flange or chorus is another effect I'll have as an option for
guitars etc. See Effects pages for settings.
Make sure that all your effects are returned through the effect returns and assigned to the
master stereo output. If you are fortunate enough to have spare channels on your desk you can
return your delay and chorus type effects back through a console channel as this gives you the
option of adding master reverb to them and using the channel EQ. Delays can soften if master
reverb is added to their returns plus you can attain your feedback from the console instead of
using the control on the effect unit. Say you are using send 3 to a delay unit you can feed back
to the delay by sending the send 3 on the return back into the unit. N.B. Incidentally, make sure
that the dry/wet or mix controls on your effect units are set to wet as you are only wanting
the effect from the units and you won't need any dry sound. (If you are using the Alesis
Quadraverb check this as all the default settings have 50% dry and 50% wet.) The returns from
effects are usually panned full stereo L/R, but you may wish to bring the drum reverb back half
L/R to separate the two.
Your console should now be setup like this

Mixing
Some mixers start with the drums, others start with the vocal. I must admit I start with the
drums as they convey the dynamic of a song. Hopefully you will have automation on your
console, if not, you must now start setting up a series of moves and remember where and how
they occur because, let's face it, the balance within a mix is not static, it varies continuously
throughout a song. For example lets say the drummer plays a rimshot snare through the verses
and full snare in the chorus. The EQ required on the rimshot snare sound is probably different
from the chorus snare sound so I often split the snare return from the recorder into two console
channels so I can EQ and effect each separately and automate the switch between the two. For
example, the snare in the chorus will probably require more reverb than the rimshot so having a
separate channel allows for that. Automation also allows for the tom mikes to be muted when
not needed thus reducing the spill of the rest of the kit and cutting out the constant ringing of
the toms which occurs with undampened toms. The overhead mikes also will need to be ridden
throughout the track, I tend to lower the overhead mikes when the rimshot is playing to achieve
a tighter sound, then I lift them in the chorus when the full snare comes in. Reverb on the
overheads gives reverb on the cymbals but it also adds reverb to the snare in the chorus and
lifts the whole ambient sound of the kit. This has the effect of changing the perspective of the
drums in a mix. You can also change the perspective by putting master reverb on the overheads
which blends with the drum reverb.
Once we have achieved a reasonable balance of the kit and the dynamics are set in place we
can add the bass. The bass and the kick drum will determine the bottom end of the track so the
balance between the kick and bass is critical. The kick will give the bass punch and attack when
they hit together.
Note: I must say a few words here about bottom end. The big mistake in mixing is to make the
bottom end sound too big by adding lots of bottom end EQ to the kick and the bass. You must
bear in mind how the track will be played back by the listener. Nowadays everyone has a stereo
system with bass boost as an option either as a loudness switch or as a sub bass control.
Everyone who has this option has it switched on!! If you get out a few of your favourite
recordings and listen to them on your mixing speakers you will find that they are relatively shy
in the bottom end and yet when played through your average boom box sound tight and fat.
You have to start to understand what a flat response really means and learn to mix that way.
If you put a bass on a VU meter you will notice how much energy there is in the bottom end. A
bass peaking to zero will have the same apparent loudness as a highhat peaking to -30db.
That's because a hithat has no real bottom end compared with a bass so be careful with your
low end EQ on basses and kick drums. I like to solo the two together and EQ them so that they
are tight but not boomy.
Add the vocal

OK, so the bass and drums are now at their first mix level so next I will add the vocal and mix it
sitting just above the bass and drums. This might mean an EQ change so they all sit tightly
together. The vocal might need to be ridden with the automation and I'll probably compress it
again to keep the dynamic range within the boundaries of the whole track. I often find that the
reverb on the vocal will need to be ridden so that the screaming high notes need more reverb
that the quiet intimate sections in the verse. Here I take a feed from the direct out of the
vocal channel and bring it up on another channel on the console. I then deselect this channel
from the stereo mix output so it goes nowhere but the aux sends are still working. By adding
reverb to this channel I can use this channel to ride the reverb on the vocal as an automated
send.
Adding the rest
Now we can start to add the fiddly bits like the rhythm guitar and keyboard pads etc. adjusting
their balance to fit tightly but not overpowering the vocal. (Please understand I am not
defaming guitars etc. by calling them fiddley bits, they are just as important as every other
part) The track should now be starting to take shape. If the dynamics of the drums and vocal
have been set correctly the placement of the additional instruments will fall into place easily.
The vocal harmonies, and solo instruments can now be mixed into the track and we are nearing
the completion of the first mixdown.
Note: It is important to keep checking your mix in mono. Unfortunately stereo and mono are
not compatible. When you switch to mono, instruments that are panned centre are 3db higher
than in stereo so your vocal, kick and snare, for example, will come up in the mix. Some
engineers actually make two mixes of a track: One that is full wide stereo with full dynamic
range for home listening and one where all the hard left and right signals are panned to the
centre or half centre and compressed for radio. It's really hard because if you make a mix sound
great on a good home hi-fi it won't have the tightness and punch a mix made for commercial
radio will have where the dynamic range is low. It's common practice to make separate mixes of
the singles from an album for radio whereas the remaining tracks are mixed totally for home hi-
fi. I think you will find that most commercial records are mixed to sound great on FM Radio.
Rest and Recreation

It is important that you constantly give your ears a break during the mixing process as your
ears have little compressors in them that will progressively shut your ears down. Have you
noticed that when you've been in a loud club with a loud band when you go outside you can't
hear as well. It's part of your ears protection system and a cup of coffee in another room
watching TV or something will allow them to start opening back up. I like to "mix from the
kitchen" as I call it. This means playing the automated mix and listening to it from an adjacent
room with the control room door open, you'd be surprised how clearly you can hear the balance
between instruments when you get away from the direct sound from your speakers. The
relationship between the bass and kick, the balance within the harmonies, the clarity of the
vocals etc. all become clearer when you relax and listen from another room.
Monitoring Level
Unfortunately the human ear is not flat at all levels. Some guys called Fletcher and Munson
worked out what the response curve of the ear was and found that at low levels the ear missed
out on the low frequencies and the high frequencies, whereas at loud levels it was the opposite.
From the above chart you can see that around 80 - 90db the ear is the flattest. The fact that we
don't hear low frequencies and high frequencies at low levels created the Loudness switch on
stereo systems which boosts the low and high frequencies to compensate for the ear.
Unfortunately, Joe Public doesn't know this but knows that when it is switched in things sound
fatter and brighter so they leave it in all the time. It is generally recognised that a level of 85db
is where the ear is at it's flattest so don't mix too loud if you want a flat response.
The important thing about mixing is apparent loudness, or relative loudness. If I whisper into
a mike and then I shout into a mike the shout will appear louder because I know that shouting
is loud. It's the same with mixing. You create an illusion of loudness, everything is relative. You
can't get bigger if you are already at your maximum. If I mix a soft acoustic guitar and vocal
and peak to zero then bring in a full kit and grunge guitar also peaking to zero it will apparently
get louder because I know that drums and guitar are loud. Mixing is the art of making signals
that all peak to zero sound as if there is a dynamic range. Nowadays with the excellent
compression systems we have most recordings are heavily compressed. I was told of a producer
who hired a mixing engineer to mix an album. The guy turned up with racks and racks of
compressors and set about compressing every track. He had one compressor for this and
another for that etc. In the end the whole mix was pumping away and almost mixed itself. That
album went on to sell millions of copies world wide. Those of you who have played with Waves
Ultramaximiser will know what compression can do for a mix. If you watch most modern pop
recordings on a VU meter the needle is almost static varying only a few db yet the tracks go
from quiet intros to full on chorus and solo sections yet still there is only a small variation in
level. So setting compression (and limiting) levels is important. I will always have a compressor
across the output of my mixes as it helps control the peaks and brings up the loudness of the
track but I may use individual compressors on separate channels.
Finally - do take the time to get a good mix. If you don't you have not given justice to all the
effort you put into recording it in the first place. It may take a few remixes, so what - it's the
final product that counts.
SPEAKERS
As discussed in the opening page, the car and open air are good listening environments because
they have either no reverberation or a controlled reverberation field. One of the key points was
the flush mounting of the car speakers. By building your speaker boxes into the wall you can
tighten the bottom end of the whole system because you eliminate the reflections that emanate
from the back of the speaker, hit the wall and then come back to you in and out of phase.
Here the sound from each speaker comes back off the walls and creates a general mayhem of
frequencies in the middle where the engineer would probably be sitting.
In the open air the sound waves are passing you and never come back.
Flush Mounted Speakers
Look what a difference the flush mounting makes to the speaker wave fronts. The difference to
the ear is even more dramatic.( As with most techniques in the modern studio in the past 3
decades Tom Hidley was the acoustician who started it.) The angles give what is referred to
as a 60 degree monitoring system. Some studios played with 90 degree monitoring which gave
a wider image and is an option you may experiment with. I used it at Music Farm Studios and I
really liked it as the image was really wide yet the centre was still tight.
As far as the construction is concerned it has to be solidly built. As with all studio construction
glue and screw as you go. Usually a frame is made and then a box 2mm bigger than the
speaker box is built so the speaker slips in tightly so you are in fact pushing hard against the air
pressure when you place the speakers in it. Some people line it with rubber pads so that the
speakers are suspended and are mechanically isolated from the whole frame. You could build
them in concrete which would be ideal but expensive.
The angle of the speakers is set so that they create a 60 degree angle at the focus. If you want
to mount them high you must angle them down so that they point at the engineer. I find it
annoying when I go to a studio and sit at the console and the speakers are pointing over my
head. Whilst that effect looks good it is expensive and complicated to build so don't attempt it
unless you've got a good carpenter as the resultant angles are complicated especially if you
have a window between the speakers.
The area underneath the speakers makes a good bass trap and the front face should be
absorptive so that reflections from the front of the console are eliminated.
Flush Mounted Speakers Elevation

The port is using the area under the speaker as a bass trap which is a good idea because there
is a lot of bass frequencies generated in the front of the control room but you may use the area
for
A rack for your power amps. This option is popular because it is generally recognised
that the shorter your speaker leads the better and the cavity if lined with insulation
absorbs the fan noise. You may also consider putting your computer stack there for the
same reason.
Have a tape recorder there. This is a good option because you can always see the
machine and observe its operation and meters.
Have a window to the studio. This is an option I've seen in a few studios. I personally
have a problem with having glass in this area because of the reflections off the back of
the console can cause all sorts of problems when it interacts with the glass and if you
have a window to the studio between the speakers you can land up with too much
interference in the front area.
Personally I reckon you should have a flat absorptive surface and a trap behind! It
completes the flat surround of the speaker.
REAR REFLECTIONS
If a signal is placed equally in each speaker (panned centre) you will hear it as if it were coming
from a centre speaker, (sometimes referred to as the Phantom Centre). It is good practice to
establish the phantom speaker in your room. If your system is correctly setup you will hear a
third (phantom) speaker coming from the centre between your speakers. There was a record
label on New York during the early 60's that even promoted their recordings because they had a
phantom speaker in their recordings. They called their recordings Dimension D and their main
artist was Enoch Light and the Light Brigade. They were spectacular recordings for their time
though!.
If you have a recording setup try this experiment. Bring the same signal up into the console
twice with one of the signals sent through a delay unit. Now pan one left and one right with no
delay. You should hear the signal coming from the centre. Now slowly add delay to the delayed
signal. You will find that something occurs around 18 -20 milliseconds. Suddenly you will start
to perceive the signals as two separate signals one from the left and one from the right yet
when the delay was below 18ms the ear couldn't tell the difference between the delayed and
the direct so they both created a centre image. (Good way to create a doubletracked guitar
effect) In other words your ears can't distinguish delay below 18ms or so. (It is different if that
delay is changing as in a flanger or phaser)
Now if you have a room with a reflective rear wall the signal from the speaker will pass you and
then be reflected back. Sound travels at approx. 330mm (1 ft) per millisecond so if you are 3m
(9ft) from your back wall you will hear the sound as a 18ms delay and your ears will be
confused and think that there is another speaker behind you. Most home studios have a back
wall closer than that so it shouldn't be a problem but I really recommend you don't try that
system unless you are fully aware of the technical problems involved. I have seen lots of LEDE
pulled out and rebuilt because the designers didn't fully understand the geometry involved and
the room sounded weird.
You could try this test. Sit in front of your speaker system and cup your hand behind your ears
to block the sound coming from behind you. If you notice the sound tighten significantly you
have a rear wall problem.
TEMPO DELAY TIMES
1/4 1/8 1/16 1/4 * 1/8 * 1/16 1/4 1/8 1/16

TEMPO Note Note Note Note Note *Note Triplet Triplet Triplet
60 1 .5 .25 1.5 .75 .375 .666 .333 .166
61 .983 .491 .245 1.475 .737 .369 .655 .327 .164
62 .967 .483 .241 1.451 .725 .363 .645 .322 .161
63 .952 .476 .238 1.428 .714 .357 .635 .317 .158
64 .937 .468 .234 1.406 .703 .351 .625 .312 .156
65 .923 .461 .230 1.384 .692 .346 .615 .307 .153
66 .909 .454 .227 1.363 .681 .341 .606 .302 .151
67 .895 .447 .223 1.343 .671 .336 .597 .298 .149
68 .882 .441 .220 1.323 .661 .331 .588 .294 .147
69 .869 .434 .217 1.304 .652 .326 .579 .289 .145
70 .857 .428 .214 1.285 .642 .321 .571 .285 .142
71 .845 .422 .211 1.267 .633 .317 .563 .281 .140
72 .833 .416 .208 1.250 .625 .312 .555 .277 .139
73 .821 .410 .205 1.232 .616 .308 .548 .273 .137
74 .810 .405 .202 1.216 .608 .304 .540 .270 .135
75 .800 .400 .200 1.200 .600 .300 .533 .266 .133
76 .789 .394 .197 1.184 .592 .296 .526 .263 .131
78 .769 .384 .192 1.153 .576 .288 .512 .256 .128
79 .759 .379 .189 1.139 .569 .264 .506 .253 .126
80 .750 .375 .187 1.125 .562 .281 .500 .250 .125

81 .740 .370 .185 1.111 .555 .277 .493 .246 .123
82 .731 .365 .182 1.097 .548 .274 .487 .243 .122
83 .722 .361 .180 1.084 .542 .271 .482 .240 .120
84 .714 .357 .178 1.071 .535 .268 .476 .238 .119
85 .750 .352 .176 1.058 .529 .264 .470 .235 .117
86 .697 .348 .174 1.046 .523 .261 .465 .232 .116
87 .689 .344 .172 1.034 .517 .258 .459 .229 .115
88 .681 .340 .170 1.022 .511 .255 .454 .227 .113
89 .674 .337 .168 1.011 .505 .252 .449 .224 .112
90 .666 .333 .166 1.000 .500 .250 .444 .222 .111
91 .659 .329 .164 .989 .494 .247 .439 .219 .109
92 .652 .326 .163 .978 .489 .244 .434 .217 .108
93 .645 .322 .161 .967 .483 .242 .430 .214 .107
94 .638 .319 .159 .957 .478 .239 .425 .212 .106
95 .631 .315 .157 .947 .473 .236 .421 .210 .105
96 .625 .312 .156 .937 .468 .234 .416 .208 .104
97 .618 .309 .154 .927 .463 .232 .412 .206 .103
98 .612 .306 .153 .918 .459 .229 .408 .204 .102
99 .606 .303 .151 .909 .454 .227 .404 .201 .101
100 .600 .300 .150 .900 .450 .225 .400 .199 .100
101 .594 .297 .148 .891 .445 .222 .396 .197 .099
102 .588 .294 .147 .882 .441 .220 .392 .196 .098
103 .582 .291 .145 .873 .436 .218 .388 .194 .097
104 .576 .288 .144 .865 .432 .216 .384 .192 .096
105 .571 .285 .142 .857 .428 .214 .381 .190 .090
106 .566 .283 .141 .849 .424 .212 .377 .188 .094
107 .560 .280 .140 .841 .420 .210 .373 .186 .093
108 .555 .277 .138 .833 .416 .208 .370 .185 .092
109 .550 .275 .137 .825 .412 .206 .367 .183 .091
110 .545 .272 .136 .818 .409 .204 .363 .181 .090
111 .540 .270 .135 .810 .405 .202 .360 .180 .090
112 .535 .267 .133 .803 .401 .201 .357 .178 .089
113 .530 .265 .132 .796 .398 .199 .354 .176 .088
114 .526 .263 .131 .789 .394 .197 .350 .175 .087
115 .521 .260 .130 .782 .391 .195 .347 .173 .087
116 .517 .258 .129 .775 .387 .194 .344 .172 .086
117 .512 .256 .128 .769 .384 .192 .341 .170 .085
118 .508 .252 .127 .762 .381 .190 .339 .169 .084
119 .504 .252 .126 .756 .378 .189 .336 .168 .084
120 .500 .250 .125 .750 .375 .187 .333 .166 .083
121 .495 .247 .123 .743 .371 .186 .333 .165 .082
122 .491 .245 .122 .737 .368 .184 .327 .163 .082
123 .487 .243 .121 .731 .365 .183 .325 .162 .081
124 .483 .241 .120 .725 .362 .181 .322 .161 .080
125 .480 .240 .120 .720 .360 .180 .320 .159 .080
126 .476 .238 .119 .714 .357 .178 .317 .158 .079
127 .472 .236 .118 .708 .354 .177 .315 .157 .078
128 .468 .234 .117 .703 .351 .175 .312 .156 .078
129 .465 .232 .116 .697 .348 .174 .310 .154 .077
130 .461 .230 .115 .692 .346 .173 .307 .153 .076
131 .458 .229 .114 .687 .343 .171 .305 .152 .076
132 .454 .227 .113 .681 .340 .170 .303 .151 .075
133 .451 .225 .112 .676 .338 .169 .300 .150 .075
134 .447 .223 .111 .671 .335 .168 .298 .149 .074
135 .444 .222 .111 .666 .333 .166 .296 .148 .074
136 .441 .222 .110 .661 .330 .165 .294 .147 .073
137 .437 .218 .109 .656 .328 .164 .292 .145 .073
138 .437 .218 .109 .656 .328 .164 .292 .145 .073
139 .431 .215 .107 .647 .323 .161 .287 .143 .072
140 .428 .214 .107 .642 .321 .160 .285 .142 .071
141 .425 .212 .106 .638 .319 .159 .283 .141 .070
142 .422 .211 .105 .633 .316 .158 .281 .140 .070
143 .419 .209 .104 .629 .314 .157 .279 .139 .069
144 .416 .208 .104 .625 .312 .156 .277 .138 .069
145 .413 .206 .103 .620 .310 .155 .275 .137 .069
146 .410 .205 .102 .616 .308 .154 .274 .136 .068
147 .408 .204 .102 .612 .306 .153 .272 .136 .068
148 .405 .202 .101 .608 .304 .152 .270 .135 .067
149 .402 .201 .100 .604 .302 .151 .268 .134 .067
150 .400 .200 .100 .600 .300 .150 .266 .133 .066
151 .397 .198 .099 .596 .298 .149 .264 .132 .066
152 .394 .197 .098 .592 .296 .148 .263 .131 .065
153 .392 .196 .098 .588 .294 .147 .261 .130 .065
154 .389 .194 .097 .584 .292 .146 .259 .129 .064
155 .387 .193 .096 .580 .290 .145 .258 .128 .064
156 .384 .192 .096 .576 .288 .144 .256 .128 .064
157 .382 .191 .095 .573 .286 .143 .254 .127 .063
158 .379 .189 .094 .569 .284 .142 .253 .126 .063
159 .377 .188 .094 .566 .283 .141 .251 .125 .062
160 .375 .187 .093 .562 .281 .140 .150 .124 .062
RECORDING WITH EFFECTS
Without doubt the biggest influence on recording styles over the past 25 years has to be the
introduction of digital effect units. When I started there were only four effects available - tape
delay, reverb chamber, reverb plate and spring reverb.
I remember when the track Itchycoo Park by The Small Faces (I think!) came out and had tape
phasing!! Wow. I remember desperately trying to figure out how they did it. It wasn't till I joined
Armstrong's Studios in Melbourne under the great Roger Savage that I learnt how to achieve it
and I used it on the now classic Australian track "The Real Thing" by Russell Morris. Now you can
dial it up as an option on almost any effect unit and you can even get it in a stomp pedal!! but
it's not the same.
Tape delay was limited to how fast the tape machine would go so it was usually limited to
quarter and eighth note delays around the two speeds of 7-1/2 (which gave 16th delays around
120bpm) and 3-3/4 ips speeds (which gave 8th delays) which were standard speedsn on the
tape machines then. It wasn't until the introduction of the digital delay that suddenly the whole
gamut of sound effects that work in milliseconds appeared and brought us phasing, flanging,
chorus.
Then came the first of the digital reverbs and a whole new world opened up. Suddenly you could
change the reverberant field around any instrument quickly and easily. To explain how all the
effects work and how to use them I will look at it from the perspective of how these effects came
about because a lot of the terms relate back to those days. For example the term flanging came
from the technique of slowing down a tape machine to create phasing effects by holding onto the
flange of the tape reel. Silly isn't it!
Select the different pages from the index list above.

REVERBERATION
Imagine someone singing in a large room with painted concrete floor, walls and ceiling. Where is the sound
going and what is the microphone hearing?
Every sound that leaves the singer reflects off the walls, floor and ceiling. Initially the sound from the singer will
reach the microphone first - followed by the first reflections. In this instance the first reflection would be from
the floor followed by the ceiling as they are the closest, then the walls on either side followed finally by the
reflections from the walls in front and back of the singer. These reflections wouldn't stop there, they would go
on and on. Then would come the longer reflections where the sound has bounced off the ceiling, hit a wall then
the floor and back to the mike. The time taken for the first reflections to arrive back at the microphone is
proportionate to the size of the room. Sound travels at 30cm(1ft) per millisecond so if the singer was equidistant
from the side walls and they were 20ft apart the first reflection from those walls would be delayed by 20ms. If
the singer was 20ft from the end wall those reflections would arrive at the mike 40ms later. Then the late
reflections would start arriving but by them the reflections would have built up and up until a reverberant field
was established where none of the reflections were distinguishable and true reverberation has occurred.
Because the room is made of painted concrete there would be a good reverberation of around 2.21 seconds
(According to the reverb calculator). The walls of this room are flat and reflective so there is nothing on
their surface (or the floor or ceiling) that would scatter the sound around like if they would if they were made of
river rocks, or were covered in triangles and boxes etc. So the reflections within the reverb are slowly bouncing
around and decaying. If the walls were made of rocks the reflections would be going all over the place and there
would be a mass of differing reflections. The reflections would be more dense or diffuse and we would say that
there was more diffusion.
So what does the microphone hear?

A Singer followed by Reverb created in a Room (not a hall) of a particular Size and made up of a First
Reflection followed by the Early Reflections and the Later Reflections and finally by the Reverberant Field
that arrives after it's Pre Delay and has low Diffusion but creates a Reverberation Time of around 2.21
seconds
But there's more. From the reverb calculator you can work out the reverb time at different frequencies. The
room above comes out like this:
The 2.21 reverberation time noted before was at 1000Hz whereas at 250Hz it's 3.09 seconds. In other words
the decay at 250Hz is longer than the decay at 1000Hz. Some of the reverberation units and programs give you
individual control over the reverb time of the high and the low end. Others allow you to EQ the reverb to roll off
or boost the highs or lows.
Anything more and you are in the hands of the programmers who write the reverb, hall, bathroom
programs, or are you ?
At 120 bmp one bar lasts 2 seconds. 32nds are 62.5ms, 16ths are 125ms, 8's are 250ms and quarterbeats are
500ms. Sound like good predelay and early reflection times to me. Set the reverb at 1 second and it'll last half a
bar. At least you can get it in time.
Before there were plates, halls, rooms etc. there were reverberation chambers. These were rooms specially build
for their reverb. They were build under the studios like Abbey Road in London and Capitol Studios in LA. They
were (or is it are?) large rooms designed to create a uniform reverberant field. A speaker (or two) were placed
in them and a microphone (or two) was setup to pick up the sound. You used it like you would a reverb unit
today. You sent a feed to the speaker and mixed the return with your track. But any area can act as a reverb
chamber, a stairwell is pretty good, a garage, the local hall, a concrete water tank is a beautiful reverb chamber,
a large concrete pipe with a speaker at one end and a mike at the other. If you want to experiment there's lots
of ways of creating reverb and now that all the recording gear is so portable, why not take the drummer out to
the local hall one day. The reverb you get will at least be distinctive and with good ambience mikes you can
control most of it.
REVERBERATION
In theory, it is easy to determine the reverberation time of a room. It depends on the volume of
the room and the rate at which the sound energy is absorbed by the wall surfaces and the objects
in the room. In a bare room, the reverberation time is thus proportional to the ratio of volume to
surface. It is customary to define the reverberation time as the time required for the sound level
to decrease by 60 dB (hence the abbreviation RT60). In 1922 a pioneer in the study of room
acoustics, Wallace Sabine came up with the formula which is used here by this calculator:
RT60 = k(V/Sa)
k is a constant that equals 0.161 when the units of measurement are expressed in meters
and 0.049 when units are expressed in feet.
Sa is the total surface absorption of a room expressed in sabins. It is a sum of all the
surface areas in the room multiplied by their respective absorption coefficients. The
absorption coefficients express the absorption factor of materials at given frequencies.
V is the volume of the room.
I got this page and the following calculator from a page offered by the New York University
education page. The author was Piotr Filipowski (Just trying to give credit where credit is due) Its
Java script which I haven't learnt as yet. I have his permission to use this great chart.
RT60 Calculator
Enter the measurements of your room. Make sure you specify the units.
Width Length Height
feet meters
Windows,Doors and other Surfaces

Walls Material
Material Size How Many
Front Gypsum board Glass-windows 0 x 0 0
Back Gypsum board Glass-windows 0 x 0 0
Left Gypsum board Glass-windows 0 x 0 0
Right Gypsum board Glass-windows 0 x 0 0
Ceiling Concrete-painted Glass-windows 0 x 0 0
Floor Wood floor Carpet on concrete 0 x 0 0
125 Hz 250 Hz 500 Hz 1000 Hz 2000 Hz 4000 Hz
Estimated RT60 of your room is 0 seconds
CALCULATE Reset Room Info
So what can we work out with this calaculator?

Let's start with a room with the following dimensions.
Length 6 metres
Width 5 metres
Height 2.4 metres.
and with all the walls and ceiling in painted concrete and the floor in unpainted concrete. (i.e. we
are going to treat the garage) We are then going to add the typical treatment (you know, get
out the egg cartons, heavy drapes and curtains etc)
1. Carpet on the floor

2. Heavy drapes on the rear wall
3. Heavy drapes on the front wall
4. Light drapes (egg cartons) on the side walls
5. Light drapes (egg cartons) on the ceiling.
The original room has a relatively flat response (Bright Blue), albeit a bit long all over and
especially in the top end, and would sound very bright and live with reverb times around .8 sec.
When the typical treatment is applied we land up with a room that has a long reverb time at
125Hz and short at 4kHz. Because the high end reverb is shorter people will say the room is dead
but in fact it's not really, it's only dead in the top end and too dead at that. The frequencies below
500Hz are the real concern.
This is the mistake everyone seems to make. All the treatment added only effects the high
frequencies. You must consider all the frequencies when you treat a room. The shorter
reverberation time in the high end is reasonable at 0.3 sec (around 0.4 -0.5sec is desirable) but
you must take down the low end as well. The reverb time at 125Hz is around 2 sec, at 250 it's
0.92 sec,at 500 it's down to 0.49 sec and it reaches 0.3 sec at 1000Hz and is right down to 0.21
sec at 4kHz. The low mid and lows need correct treatment. See the low-mid and low frequency
absorber pages.
RECORDING WITH DELAY
Tape Delay
The first delay was created using a tape machine. The following is a drawing of the tape path in a
typical tape recorder.
The tape first passes the erase head that wipes the tape. Then it passes the record head where the
signal is put onto the tape. Finally it passes the play head where it is played back. The time taken
for the tape to pass from the record head to the play head determines the delay. If the tape
recorder is going fast the delay will be short, if slow the delay will be long. The speed of the tape is
determined by the speed the capstan motor is turning and in the early seventies the tape machine
manufacturers started to add variable speed motors with a control called varispeed. Using this you
could set the delay (by ear) to the appropriate speed. If you took the output from the play head and
fed it back into the input of the recorder you would get repeat after repeat as it cycled around. As
only a small amount of the first delay was fed back it would continue round the loop dropping in
level each time and the classic delay was created. This control was called Feedback.
If you were using a stereo tape recorder you could have stereo delay. If you put in a mono signal
you got out a mono delay because each side of the delay was the same. Thus:
Which sounded like this visually!
Click on the Image to hear the sound
Because both sides have the same signal the delays sound mono in the centre.
Enter the Digital Delay Unit.
Digital Delay
The digital delay unit changed everything. Firstly you could dial up the delay you wanted but more
importantly you could vary the delay of the left and right sides. So if you put in a mono signal and
set the left side to 500ms and the right side to 250ms, applied some feedback and you ended up
with a delay in which the left side was different to the right side and true stereo delay was starting
to be a reality.
Example 1 Click on the Image to hear the sound
I hope you are following me here - what I'm trying to do is to get you to picture the delays as they
would be heard. On the left side we have a 500ms delay while the right is at 250ms. Every 500ms
the left and right delays are the same - therefore the sound comes from the middle. What you think
is stereo really isn't, what you are wanting is something like this:

Or this
or this!

This is where the understanding of what makes things stereo and what is mono is extremely
important. The thing that makes a sound come from the left can be more than just it being a mono
signal coming from the left. It can be a stereo signal, where the left is different from the right, yet
appears to come from the left. If we were to set the delay so that the left delay is 510ms and the
right is 490ms the delays would be 20ms apart, right? If two sounds are 20ms apart or greater they
sound like two different signals left and right - so if two delays are 20ms apart then they should
sound as though they come from left and right. Thus:
Sure the left delay will be 20ms later every delay but in 4 delays that's only 80ms out of 510ms and
remember relative to the beat one delay starts 10ms ahead whilst the other is only 10ms behind.
Try it with a track as you'll see what I mean.
Another way to make each side dissimilar is to change the pitch of one side or both sides. I created
Example 3 by doing just that - I changed the pitch of one of the sides so even though they are in
time they appear as stereo because they are dissimilar.
I created these delay sounds using a Multi-Tap delay. Instead of using feedback to create the
repeats as with tape and straight digital delay, in a Multi-Tap Delay you can control each delay. If
you imagine that each delay is a Tap, you can set what each delay will be at each tap - even where
it is panned. This is much more extensive a control of the delays than using straight feed back
where each delay is just a repeat of itself.
You can see how I've played around with aural visualisation and that's what the guys and gals who
make all those incredible delay programs do all day. Next time you get one out try looking at the
sound it makes and picture what's going on instead of just listening to it.
sound has depth,height,breadth
Setting the Delay time
Setting the delay time depends on the Tempo of the track you're recording. If the tempo is 120
beats per minute there are 120 beats in 60 seconds or 120 beats per 60,000 milli seconds which is
one beat every 500 milli seconds. So with 4 beats in a bar, quarter beats are 500ms, eighths are
250ms, sixteenths are 125ms etc. So how do you find out what the delays are if you know the
tempo? There are some computer programs like Beat Calc that will automatically work it out for
you, some of the new delay units and programs have a tap function that allows you to tap in a
tempo and the device will work it out and you can get charts with it printed out. They will not only
tell you what the 1/2, 1/4, and 1/8th beats etc. are but will also tell you the dotted note delays and
the triplet delays. I've created a chart for you called the Tempo Chart
Quick Delay Calc

There is a quick way you can work out the delays of a track using a stopwatch that reads 100th of a
second.
Play the track and start counting the quarter beats. Then start the stop watch on the beat and count
ten quarter beats and stop the clock on the eleventh beat. You will get a reading like:
Thus a quarter note beat will be 460ms, an eighth will be 230ms and sixteenth will be 115ms etc.
This is a handy technique if you are a PA mixer and you want to put a delay on the vocal and you
quickly need to work out the tempo the band is playing at.
ENTER THE MODULATORS
The Phaser/Flanger
When a jet flies over, you hear it coming and its sound is going up in pitch - then when it goes away
its pitch drops. This effect is called the Doppler Effect. If you had a very tight delay of 10ms but
could change it using a modulator so it varied from 0ms through to 10ms and back to 0ms etc.
sweeping forward and back relative to the original signal. When the delay is increasing the phase
shift is increasing and the doppler effect will cause the sound to lower in pitch like the jet flying
away and when its modulating back and shortening the delay the phase shift will cause the pitch will
rise like the jet coming towards you.
That is the classic phaser sound. The effect was originally created on short wave radios where a
receiver was picking up a signal that had come around the world one way as well as another longer
way and when the two were added together at the receiver they added and subtracted from each
other causing the phase shift or Comb Filter Effect that creates the sweeping pitch effect we now
associate with phasing, and which is also why they are called phasers. Phasers shift the phase
relationships within the sound using phase shift circuits. Flangers are the same sort of thing,
except that flangers use time shift circuits to obtain the effect. The modulator has controls like
Delay which is how much delay do you want,

Depth which is how much do you want the modulator to control the delay,

Rate which is how fast or slow do you want the modulator to oscillate,

Feedback which is like the tape feedback, sending the signal back into itself and

Shape which is how do you want to drive the modulator, i.e. with a sine wave shape or a
triangle wave shape etc.
The Chorus Unit

If you increase the delays from the 0 - 10 ms area and go out to the 60ms - 80ms delays but still
modulate the delays the effect changes to the Chorus effect.
The guitar in this track has the classic chorus guitar sound most of which was created using the
classic Roland Dimension D, in fact there were two of them. The modulation rates are usually faster
than phasing but the depth is a lot less so there is only subtle change going on. If you look at the
controls on a chorus unit you will find the same controls as in phaser units, Delay - Depth - Rate -
Feedback - Shape. You can set the delay on a chorus to be in time with the track so if the tempo is
120bpm the 16ths are at 125ms and the 32nds are at 62.5ms - or you could try 62.5ms and
31.25ms as well. It really makes a difference.
AUDIO WIRING
The main thing about audio wiring is understanding how the earthing works. Lets take the
connection of a 24 track analogue recorder as and example. You send a balanced lead from the
balanced output of the console to a balanced input on the recorder. You then return to the
console with another balanced lead. Now remember, in the balanced system the audio runs
through the +ve and -ve leads. The earth is just a shield established to drain unwanted
interference off to ground. But if you connect the shield at both ends of each lead you are
establishing the potential for an earth loop. It is in fact joined to itself in a loop. In fact it's two
loops because each machine is connected to an earth as well. If on the other hand you
disconnect the shield going to the recorder at the recorder end any interference generated in
the shield must go to the console earth. Now if you disconnect the shield in the return lead but
disconnect it at the console end any interference will be drained to the recorder's earth. No
loop! Because the recorder and the console both have mains power in their circuits there must
be a link to earth for safety so don't de-earth to get rid of the hum when there is a safe way
such as this.
BALANCED
Now the same problem but with unbalanced leads.
Remember that the earth is now the negative as well as the ground
UNBALANCED
As you can see the recorder has a connection with the positive but also a connection to the
negative but via the earth (ground) and once again there is no loop and both machines are
earthed safely.
In the modern studio there are lots of simple external power supplies that just feed a single unit
like a reverb unit (wall warts we call them). Have you ever noticed that they are not earthed to
ground. Their mains connection has only two pins. These units allow the circuit to float above
ground so the shield must be connected for the unit to receive the negative feed.
Some may say that by not earthing the wall warts it is dangerous but as they don't feed high
voltage (typically 9 - 12 volts) to the units it's not necessary. If you did disconnect the shield in
this circuit, because there is no negative, the sound would become what they call one legged
and the sound would be thin and low in level.
So when you start wiring up your studio think of what is earthed and what is not and then you
can establish when it is safe to de-earth a unit to minimise ground loops.
Additional things to consider
Maintaining Phase: It is essential that your wiring maintains constant phase. With
unbalanced leads it's pretty obvious - the centre wire is positive and the shield is
negative/earth. But with balanced leads you can run into problems. Unfortunately the
world has two standards. On your standard microphone plug (often called by the brand
name Cannon) Europe uses pin 3 as the positive and pin two as the negative and pin 1 is
earth. On the other hand the US has pin 2 as the positive and pin 3 the negative. The
same applies to inputs and outputs on equipment. It is essential that you check each
piece of gear and work out which pin is positive and which is negative and wire
accordingly. Pin 1 is always earth and is usually a little longer so it connects first. In ring,
tip and sleeve plugs the tip is always positive, the ring negative and the sleeve earth. A
handy little piece of gear is a microphone line phase checker that has three lights that
check the three lines in the lead for continuity and phase.
Guitar Lines: It is advisable to incorporate guitar leads between rooms. This allows you
to plug a guitar into a jack in the control room and pick it up in the studio and plug it into
an amplifier. Guitarists often like to play in the control room, especially if you are using
effects, so a cable between rooms saves having to run leads through doorways. I have
seen ads for a product that has a battery powered amplifier in the cable that
compensates for the high frequency loss experienced when running long unbalanced
guitar leads.
Speaker Leads: There is a lot of discussion on this topic and proprietary speaker cable
can be purchased, but it is expensive. If you can't afford it use the standard power cable
as used by your electrician. When running your speaker leads run at least two sets per
side. This allows for a replacement if one cable gets damaged also it allows you to go BI-
amp later should you want to without having to climb behind the speakers to add the
extra cable.
Transfer Lines: It's a good idea to have some standard line level lines between the
control room patchbay and the other rooms. You can feed line level instruments like
keyboards down them and plug them straight into the line inputs on your console.
Telephone Lines: It is a good idea to incorporate a telephone line into your control
room, especially now that computers are common and you may need to hook up to the
web for software update downloads. We are not far off having the ability to record in real-
time down these lines and can now transfer Wave and MP3 files.
Direct Boxes: Direct boxes are designed to match impedance between your guitar and
the microphone input. Without going into the full electronic detail here basically what
happens is that a guitar is designed to plug into a high impedance input whereas a
microphone input is a low impedance input. Plugging a high impedance magnetic pickup
into low impedance results in a loss of highs. Direct boxes can match the impedance by
either using a transformer (passive) or using a circuit (active). Active direct boxes
are identified by the fact that they have power - either as a battery or powered by the
phantom power system. DI boxes also have a pad switch to reduce the level of a line
level instrument down to the lower microphone input level.
Patchbays
Patchbays can save a hell of a lot of trouble when interfacing your recording equipment. Even if
you only have a bedroom studio it is a lot easier if all your gear appears on a patchbay and you
can easily patch one thing into another. Patchbays can be cheap or expensive depending on the
style and construction. They can also come in balanced (Ring tip and sleeve) or unbalanced (tip
and sleeve). It really depends on the gear you have and your requirements but don't overlook
the advantage of having a patchbay.
Lets look at the standard layout of a patchbay:

The inputs and outputs usually go like this:
The main idea here is that each row is normaled to the next. i.e. microphone line 1 is directly
connected into preamp in 1. Insert send 1 is directly connected to Insert return 1 - group 1 is
directly connected to recorder in 1 and recorder out 1 is directly connected to line input (tape
return) 1.
In other words, with no patch leads the circuit is complete and you only use a patch lead if you
wish to change from the normal - that's why it's called normalling
Here we have a standard stereo plug and socket. When the plug isn't inserted the +ve and -ve
pins are shorted to the two normalling pins. The normalling pins then connect to the through
connections.
As you can see the insertion of the plug breaks the normalling and allows the new connection.
Prebuilt patchbays often have the normalling as an option. Tascam have some excellent
unbalanced ones but fully balanced normalling patch rows are expensive. A simple check is to
count the pins - 3 pins are standard and a normalling patch bay has 5 pins. (Earth is common)
Other normalling areas to consider are your console outputs being normalled to your master
compressor input and its output is normalled to the input of your DAT recorder. Then the output
of your DAT recorder is normalled to your External Monitor input. That allows you to start
mixing without having to setup a huge patch . If you need to access your master compressor
you just patch into it and break the normalling.
Another area is your Aux Sends. It is advisable to normal your regular setup - such as 1 &2 to
your headphone amp, Aux 3 & 4 to your stereo reverb, Aux 5 to your effect unit 1 and Aux 6 to
your effect unit 2. You can go further by normalling the returns of your three effects units into
tape/line returns 23 - 28. With such a setup you can start a mix without having to patch a
thing!!
If you are the only user of your studio it is probably not really necessary to label the patch bay
fully but if you have outside clients it must be labelled clearly.
BI and Tri - Amping

In a standard speaker the various components,(woofer, midrange and tweeter) and divided
from each other with what is referred to as a crossover unit. What actually happens is the
crossover divides the frequency response into 2 or 3 bands. The lows drive the woofer, the mids
drive the midrange speaker (often a horn) and the highs drive the tweeter.
Here the signal from the console goes to the amplifier and then to the speaker. Within the
speaker the crossover circuit splits the frequencies into to three and feed to each speaker.
Alternatively here the output of the console goes to the electronic crossover unit that then feeds
to each amplifier that drives a speaker independently. Although you need three amplifiers the
amps don't need to be as big. Big PA systems run on this system and are described as being 2
way, 3 way and 4 way - bi - amped, tri - amped, and quad- amped. The additional crossover in
the 4 way system feed the low mids.
You can now buy small near field monitors that have the electronic crossover and the amps built
in - all you need to do is connect the output of the console into the rear and you are away. The
multi crossover multi amping system is extremely efficient and you don't need huge 500 watt
amplifiers etc.
POWER & LIGHTS
I'll deal with power first because the audio wiring comes after the power installation and
because they both interact we'll sort the power out first. The important thing about power is not
how much is coming in or

do I have enough grunt
but " Is it earthed correctly??"
Balanced and unbalanced electrical circuits

We use balanced and unbalanced leads all the time in the studio but what does it mean? The
following diagram illustrates the difference.
Here we have the two standards. The mic lead and the guitar lead. Balanced and unbalanced.
Three wire system and two wire system. Note that in the mic lead the positive and the negative
don't contact the earth whereas in the guitar lead the negative and the earth are one in the
same thing. The earth - (ground) is exactly that. The green earth wire goes to a copper stake in
the ground so that any short circuit between the positive and earth will send the current to
ground. But because the positive and the negative don't contact the earth it is said to be
floating above ground. The shield acts as a protection from interference by sending any
extraneous electrical interference like hum, to ground. Unfortunately in the unbalanced circuit
negative is ground!
So you would expect that your standard electrical feed from your power supplier would be
balanced. Well unfortunately here in Australia it's not. Sure we get a red positive and a black
negative from the power companies transformer but by Australian regulations the electrician
must link the negative to the earth so we become unbalanced. I understand that is not the
system in the US which is why Marshall amps hum in OZ but don't in the US. I would be
interested in any information I could receive on this matter from anyone from the US.
The system designed to get around this is called the Star Earthing System where you ask your
electrician to earth each power outlet individually like this:
STAR EARTHING OF MAINS POWER

In this setup each power point sees the same ground directly and a unit earthed to outlet 1 and
connected with a patch lead to something earthed to outlet 2 won't see outlet 2 as it's earth
because it has it's own more direct route to ground.
The earth, as stated before, is connected to a copper stake in the ground. It is definitely
advisable to increase this factor by getting your electrician to put two or more stakes in the
ground and connecting them together to increase your ground connection. I've seen systems
where designers have put a whole web of copper stakes under the concrete slab before it is
poured to ensure a good ground connection. In this country where it gets very hot the ground
around the stake can dry out and the connection gets weaker and weaker. It can be solved to a
certain extent by pouring salty water around the stake but two or more stakes is a better
solution.
Lighting: It is advisable to have your light circuit separate from your power circuit. This
decreases the chance of lighting interference in your power circuits. (See lighting further
down this page)
Three Phase Power: Ideally you should have three phase power into your studio. Obviously
the home studio owner won't have it but if you are looking at a professional facility it is a
beneficial addition. The advantage of three phase power is that you can spread your electrical
circuits over the three phases:
Phase 1: Equipment and studio power.

Phase 2: Lighting.
Phase 3: Air-conditioning
Transformer Isolation/Power Conditioning: It is now becoming common to install a power

conditioner in a studio. The advantage here is that you have a transformer between you and the
supplier so that spikes are smoothed and with additional circuitry you can have a voltage
stabiliser that keeps your power voltage stable no matter what the supplier is giving you. You
can also have an added feature that adds battery backup in the case of power failure. This is
great when you have computers as it allows you to save your current work. All these features
are advantageous but can be very expensive! In a three phase setup you can put the
conditioner over your equipment and studio power phase only. One of the common annoying
items is the fridge. Fridges are prone to sticking spikes in the power so watch out for that one.
It gets really tricky setting up your earthing requirements but if you start with your mains
power installed correctly you've got a better chance when it comes to your audio wiring.
LIGHTING
Good lighting is essential in a studio and ideally a separate circuit should be allowed for it.
Downlights over the console and effects area are advisable plus additional downlights for the
client etc. Lighting dimmers can also make for a comfortable environment but be careful here.
You will probably find that the standard light dimmer will cause a buzz interference in your
electrical circuits. I suggest you discuss this with your electrician. There are light dimmers
available (zero crossing) that don't interfere with your electrics but they can be expensive!!
It's not a bad idea to test a few different dimmers before you purchase the full set. In a studio
situation you often require full lighting if the musicians are reading charts through to low mood
lighting when the vocalist is performing a soft ballad. I believe dimmers are the only way to
go.
It's not a bad idea to have control over your studio lights from the control room with a lighting
panel mounted somewhere in your control room. That way you can control your lighting from
one place.
A recent addition to the lighting system is the 12 volt lighting system. This is a good idea in a
studio as these lights are already transformer isolated through the power supply which delivers
the 12 volts.
RECORDING ELECTRIC GUITARS
AND BASS GUITARS
Electric guitars lend themselves to multiple recording techniques. You can put a mike on and amp and leave it at that or you
can try all sorts of things. The following options are available:
1.
Direct feed from the guitar.
2.
Close mike on the amp
3.
Ambience mike on the amp.
4.
Direct feed from the effects units
5.
Room Ambience mike.
6.
Second Guitar amp.
Direct Feed.
If you are fortunate enough to have a real-time analyser you will find it interesting to plug your guitar straight into it and look
at the frequency response a guitar puts out. The standard Fender Strat peaks at around 7kHz and rolls steeply off from there
up whereas the old classic Les Paul peaks at around 4kHz and falls off quickly from there. It's worth noting that factor when
listening to the direct sound from a guitar. If you are going to plug the guitar directly into the console you will need a direct
box.
This is a box that matches the impedance of the console and the guitar. A guitar is designed to plug into an amplifier that has
a high impedance input whereas a console mike input is designed for low impedance inputs thus the direct box. The
impedance matching circuit can be either a transformer - passive - or a circuit - active. If your unit is an active one it will
require power which can be supplied either by an internal battery or by Phantom Power fed from the console mike input. Once
plugged into the console have a listen to the sound. You will find immediately that the sound is dull and has no real bite in the
top end like we are used to in a guitar so quite a large amount of high end EQ is required to brighten up the sound. You can
put the direct feed through some effects units and compress it and it will probably sound better but it won't sound like an
electric guitar as we know it . On the other hand a small amount of the equalised/compressed direct signal added to the amp
sound can add a soft presence to the sound that is nice in certain circumstances like a soft chorus guitar playing chords etc.
To get the full grunt of a guitar you will need an amplifier.
Miking an amplifier.
The thing about guitar amplifiers is that they have a huge amount of upper-mid and high end equalisation in the first stage,
which is called the pre-amp, to compensate for the lack of high end in the original signal. Guitar amps also have addition
equalisation on the front panel as an option. This equalised signal is then fed to the power amp and the speakers. Some
amplifiers allow you access to the signal after the preamp and before the power amp. It is then possible to take a split of the
signal after the preamp , with all the additional EQ, and feed it into a direct box and then straight to the console.
The standard mike technique for recording an amp is to place a mike 10cm(4") from the speaker at an angle.
You will note the mike at the rear of the cabinet. This mike has a boxier sound than the front mike and is 180 degrees out of
phase to the front mike so a phase reveral is required. Remember when setting the sound of an amplifier to put your head
where the microphone is. The front of a standard amp is directional and if you stand above the amp you won't get the true
sound coming from the speaker. The microphone used must be capable of handling high sound pressure levels.
Adding an ambience mike.
An ambience mike will add another dimension to the sound. It can be another cardiod mike or you can us a U87 in a figure 8
pattern. (Very popular) This puts the direct sound off axis to the ambience mike and it also picks up the room ambience. This
extra mike can be mixed with the other mike onto one track or it can be tracked to another track allowing you to adjust the
balance at the mixing stage. It can also be panned differently than the close mike which gives the guitar sound a stereo sound
with more breadth and makes the guitar sound bigger. Alternatively you can use a MS Stereo setup.
Using effect boxes.

Most guitarists these days have a bank of effect units setup between the guitar and the amp. You can intercept them by
plugging them into the direct box before the amp or you can use your own effects. You must remember that the sound coming
out of the DI box will not be the same as the one coming out of the amp because the amp adds all its EQ etc. but a feed from
the units can contribute to the sound. Some of the effect units such as the multipedal floor units also operate in stereo and
can give you a stereo feed of the signal with stereo effects.
But what about my own effects I hear you say - why should I use that cheap $150 delay stomp box when I've got a $2000
delay unit. This question is a matter of choice - the guitarist might like the cheap effect unit , is used to it and has created a
sound around it - on the other hand you may be able to produce a much more diverse delay effect. Remember the guitarist's
effects are going through the amp whereas yours aren't. This is where you and the guitarist must play around and try different
things. If both of you are into getting the best sound you will get it but if you are both into maintaining your respective egos
all hell could break loose.
For more info regarding using effects units go to the pages on Using Effect Units.
Adding a room ambience mike.

You can go one step further than the close ambience mike and add a room mike (or two). This can give that large grunge
guitar an extra beef and for extra effect can be gated so it stops short when the guitar stops. It can be a mike like a U87 with
a figure 8 setting or you can use a shotgun mike and aim it at the amp. Considering that sound travels at around 30cm(1ft)
per millisecond a mike at 15ft is going to be delayed by 15ms. This could be great or it could be awful - experiment!!
Adding a second amplifier.

You can also add another amp and split the guitar feed into each. If you have a stereo effect system you can split it left and
right, mike each amp and put a stereo ambience mike between both amps. You can set each amp up differently, or use two
different amps. If miked separately you can achieve a perfect double track as each amp will sound different but have the same
signal.
Playing in the Control Room.

Most guitarists like to play in the control room even though their amp is in the studio. This allows them to hear the guitar as it
would in the track on your speakers and with any effects that you've added. To enable this you must run a long guitar lead
through to the amp. It is worth considering having a plug in the wall that they can plug into that can be picked up on the
other side of the wall and plugged into the amp. Alternatively you can run a long lead via the doors - unfortunately guitar
leads don't like being long as they loose high frequencies when travelling long distances. One way to stop the loss is to use
two passive transformer based direct boxes. You plug the guitarist into one in the control room and then take the low
impedance feed out and run that into the studio. In the studio you plug in the other DI box and come out of the guitar input
and plug it into the amp. What we are doing here is using low impedance to travel the distance and bring it back up to high
impedance to plug into the amp.
You will need a sex change plug from male to female XLR to get back into the second DI.
Additional Factors.
There are a few additional factors that must be considered here. I'm sorry but a great engineer can't make a bad guitarist
sound great!! There are a few things that can seriously effect the sound a guitarist makes. Firstly, is the guitar setup
correctly? Apart from the pickups, model etc. which are set, the variables are - correct alignment of the neck so that the
strings are not too low. If they are too low you will experience string distortion caused by the string hitting the adjacent fret,
which tends to muddy the sound as the string is not free to vibrate evenly. Secondly the strings used can be too light. A guitar
strung with light gauge strings will not sound fat and grungey. A very good guitarist friend of mine says that most people can't
play his guitar because it is strung so high and the strings are heavy gauge, but believe me his sound is great. From a musical
point of view the guitar might not have the harmonics in tune so that when the guitarist plays up high on the frets the guitar
is flat or sharp. All these factors affect a guitar sound but you can't beat the truism that if you want a great guitar sound get a
good guitarist.
There are many ways of approaching recording the electric guitar. The main thing I believe is to give yourself as many options
as you can. Experimentation is the call here, as with the acoustic guitars take the time to put up all the mikes and experiment
with the different combinations. I can remember when I was recording an OZ band called Mondo Rock and we wanted a close
sounding power chord in a song called "Come said the Boy". The sound we wanted was a Marshall wound up to 11 but
recorded close. We tried every mike we had but they all distorted when put so close to the amp, even an SM57 fell over. That
day a rep from Neuman came to the studio to try to flog us the new TLM mike. I was reading the specs and it said that it
would handle up to 139spl so we asked him if he could leave the mike with us and we'd assess it. When he'd gone we quickly
stuck it on the amp and bingo! it worked. The song went on to sit at number two on the charts for about eleven weeks
constantly stopped from going number 1 by John Lennon's Imagine. Them the breaks!!
BASS GUITAR
The electric bass guitar differs from the electric guitar in that the direct signal from the instrument does not need special EQ
so direct feed via direct box is the normal way of recording a bass guitar. Typically most bass amps offer an extensive EQ
section and some offer a valve preamp but the bass amplifier is just a dirty big power amp which is required to move the
cones of the large heavy speakers. Often a bass amp setup will have two boxes, one with a set of 10" speakers and another
with a heavier 12" or 15" speaker. In this setup you can mike each box individually
The bass guitar also lends itself to bi-amping where a crossover circuit divides the signal into two or three frequency bands
and uses a separate amplifier and speaker for each band.
..............
The split from each frequency band is sometimes available as a console feed from the rear of the amp so you can take
two/three direct feeds into your console. This allows you to compress and EQ each band separately and assign them to
different recording tracks for full control later in the mix. The crossover frequency is selectable in most amps with the
crossover frequency usually at around 100 - 150Hz with the 10" speakers handling the high section and the larger 12"/15"
speakers handling the powerful lows.
I often feed the bass straight into a DI box and have the player in the control room which helps separation. The bass guitar
lends itself to compression. The low frequencies it produces contain a lot of energy and containment with compression is
recommended.
FITTINGS
The Control Room
Fitting out the control room is an important part of the construction process. The correct
ergonomic layout of equipment and machines can make the room a pleasant place to work.
The position of the console is set because of the position of the speakers but how the
effects units recorders etc. are arranged is a variable. It is important that effects units can
be accessed from the rear so that cables etc. can be accessed. I have found that the area
above the meter bridge on most consoles is a good place to have at least one rack unit
space for effects:
This system is great for the commonly used effects that you need to see operating such as
compressors, gates, reverbs etc. It also tends to bring the nearfield monitors up to ear
height with consoles like the Mackie 8 buss, Tascam 3500, Yamaha O2R etc. The cable duct
at the rear also cleans up all those stray leads that normally hang around the rear of the
console. The leads can be bought up into the duct through one port at the end through a
simple timber duct. The back plate can be hinged so access to the rear of the console is
available. Most effects units will fit into this system and these days they seem to be getting
smaller and smaller. Your normal Mackie 8 Buss will allow 6 - 8 single rack space effects
units to be mounted this way.
Further effects units can be mounted either in side wings or in a full effects rack behind
you.
THE REAR EFFECTS UNIT

EFFECT RACK WINGS
The effects wing rack construction is the same as for the rear effect unit. With this system
the cables can all run within the racks and access to the rear of the equipment can be
gained through removable panels at the rear. These wings are also good for computers or
keyboards or any additional effects added for the session. Visually they can be glanced at
quickly which is more friendly than the rear units.
It is a good idea to put air grills in the back plate to allow air circulation to keep the units
cool. (Never underestimate the heat generated by effects units. 10 x 50 watt units equals a
500 watt heater!!) For large powered units, especially valve units it is advisable to space
the units in the rack to allow for total air circulation. Single rack spacers with grills can be
purchased for this from your local equipment supplier.
Down lights mounted above the effects racks are a simple but effective addition - why do
the manufacturers make their units black?? Power outlets can also be built into the rear of
the unit which should be installed as per the directions on star earthing. The cable duct
in the flooring can be bought up within the unit so cables are hidden.
The rear effects unit system is good because it provides a work bench for keyboards or
additional equipment. It is therefore advisable to have some tie-lines to the console
patchbay so that the keyboards/effects can be accessed at the console. Some people
mount these on the rear of the unit while others prefer to have plugs mounted in a single
rack unit strip on the front with the other effects.
The height of these units is a matter of comfort but I find that 720mm (2 '41/2") is a
good starting point as it's not too low for the tall and not too high for the short.
The mounting of effects units is a matter of budget. You can screw them into timber if you
don't intend shifting them around or proprietary rack mounting strips can be purchased.
There are two different types of rack mounting strips. One system has square cut-outs that
a nut clips into and then you screw into it. The other type has holes with a screw thread
welded into it. I prefer the latter.
The square hole system is designed to give latitude for the sizes but these days the
manufacturers build rack units to a set standard and I always seem to loose those little
sprung square nut fittings!
CONSTRUCTION
Isolation
Before you can consider your construction you must consider your isolation requirements. Pages
could be written on this subject but you must consider how much isolation you really want. The
idea of perfect isolation from external noise started in the days when loose miking techniques
were used. One microphone suspended over a string section meant the mic was wound up fully
and was extremely sensitive to ambient noise. Nowadays a mic 6" from a marshal amp is a
totally different story. At Big Toe Studios I often have a window open and the artist will say -"
Hey I can hear the birds, should I close the window?" To which I reply, "No, the only person
who will hear it is some stoned out freak with headphones on who will remark excitedly - wow
man I can hear birds on this track!" But if you have problem neighbours who don't like drums
pounding all day I suggest you apply a certain amount of sound isolation.
The acoustic term here is Transmission Loss. When sound hits a wall there is a certain
proportion of the sound reflected back into the room, some is lost in the absorption of the wall
and the rest travels through the wall and is called the transmission loss.
TRANSMISSION LOSS
The measure of the amount of sound that is transmitted through the wall is called the:
Sound Transmission Class- STC
The transmission loss obviously varies relative to frequency - the STC is a specially weighted
reading across all frequencies and is centred around 500Hz.. Every different wall construction
has a different transmission class.
When sound hits a wall the energy is transferred through the plasterboard to the other side via
the connection to the stud. This problem can be reduced via two ways:
Staggered Studs. Here you use two studs for each side of the wall. The plasterboard on
one side is attached to one stud and the plaster on the other side is attached to the other
stud. The two studs are connected to a common base and top plate.
Flexible Channel. Here a metal channel is attached to the stud and the plasterboard
attached to the metal channel thus reducing the connection to the stud. The channels are
mounted horizontally at 600mm (2 feet) centres. This system is extremely effective -
check out the figures in the STC Chart.
Studs:
Except for staggered stud systems, substituting timber studs for steel studs
generally results in a significant decrease in sound isolation.
Increasing the thickness of steel studs from 0.55BMT to 0.75BMT or 1.15BMT will
decrease sound isolation
Decreasing the stud spacing will decrease the sound isolation.
It is also interesting to note here that the higher the transmission loss the less reflected sound.
In other words in a tent there is a high transmission loss but also a low amount of reflected
sound so a tent makes a good recording room!! So people in the country who can afford a high
transmission loss because there are no close neighbours can allow their sound to get away thus
reducing the amount of treatment required to handle the reflected sound.
The standard gypsum wall in a house has a high transmission coefficient at 100Hz as well as a
high absorption figure because the gypsum panel's resonate frequency is around that figure.
Therefore the reverb in the room is low around 100Hz but higher around 300Hz where the
transmission and absorption are lower. That is why most rooms in a house have a reverb peak
around 300Hz. (You know the one you keep taking out of kick drums and toms.) Check it out on
the reverberation calculator.
Perfect isolation can cost heaps because there is only one thing that will stop sound and that is
MASS. The following solutions apply:
Floating the rooms. A typical construction consists of creating new rooms within your
existing rooms. This means building a floor on top of the existing floor with neoprene
isolation pads and Rockwool on the underside and then building walls and a ceiling using
the new floor as a base. This is the ideal system for total isolation because the new room
is not mechanically connected to the main room but it is also the most expensive system.
The main advantage of floating rooms is the low frequency isolation it gives. If you are
building in a block of flats it would be essential as it is the only real way of achieving total
sound isolation but if you are building a garage studio it really would depend on how
much isolation you require. If you are an acoustic folk band - forget it - if you are a
heavy metal power band it is essential because it is the only way you will stop the bass
and kick drum from annoying your neighbours.
Double Walls. This basically means building two timber/gypsum sheet walls between
each room with Rockwool in the cavity. If you wish to go further you can double the layer
of gypsum and even further by sandwiching a layer of fibre board between the two sheets
of gypsum. (This is extremely effective because sound doesn't like going through
changing medium densities). There are some companies who make sound isolation
gypsum which is thicker and heavier than normal gypsum sheet. In Australia it's called
Soundcheck and is 16mm(5/8") thick. The hollow concrete block -(Besser Block) is an
excellent wall construction as it attenuates sound efficiently and cheaply.
Sealed Environment. There is one important factor that must be understood about
sound isolation. If you build a beautifully sealed wall between two rooms you will get
good isolation BUT if you put one nail hole in the wall you will loose a lot of the isolation!!
Sound is really not the waves we keep on describing but air pressure difference so if you
allow the two air masses to join at any point the pressure gradient will transfer, so make
sure that all walls are sealed tightly. Also make sure all joints around doors, windows
and air-conditioning ducts are sealed.
Double Wall with Floating Floor
In the drawing above I've shown a single gypsum (pale blue) layer but adding to the layer can
dramatically increase the transmission loss. The options are
Adding another layer of gypsum which is glued (not nailed) to the first sheet and should
be a different thickness than the first sheet. i.e. 16mm (5/8") and 12mm (1/2")
sandwiching a layer of fibreboard between the two sheets.
This really works well, a double wall with a triple layer as described above on a floating floor
will create a room that will allow you to set up a band and not hear it outside the room! Just
remember that all the sound is now trapped inside the room and heavy acoustic treatment is
required to control it all.
STC CHART
In the following charts all plasterboard joins and edges are

sealed with an appropriate acoustic sealant. The figures
given below are based on all surfaces being sealed to
airtight.
TIMBER STUD STC RATING
Standard stud wall with one

layer of 16mm (5/8")
No Insulation 28
Plasterboard
Typical wall construction No Insulation 35

with 1 layer of 16mm(5/8")
Plasterboard on each side of
95x35mm (4 x 11/2") timber
studs.
With Insulation 38
Typical wall construction No Insulation 41

with 2 layers of 16mm(5/8")
Plasterboard on each side of
95x35mm (4 x 11/2") timber
studs.
With Insulation 45
The Flexible Channel
Single 95x35mm
(4"x2")Stud wall with 1 layer No Insulation 40
of 16mm (5/8") Plaster
board on one side and 1
layer of 16mm on the other
side on a horizontal flexible
With Insulation 47
channel at 600mm (2ft)
centres screwed to the stud.
Staggered stud wall
construction with 1 layer of
No Insulation 42
16mm(5/8") Plasterboard on
studs of 95x35mm (4 x
11/2") on a 120mm (4 With Insulation 48
3/4")common base.
Staggered stud wall

construction with 2 layers of
No Insulation 50
3/4")common base.
Staggered stud wall

construction with 3 layers of
No Insulation 56
3/4")common base.
BRICK VENEER
The typical Brick Veneer
construction with a layer of
10mm (3/8") plasterboard.
This is the standard
No Insulation 53
Australian house
construction.
With Insulation 55
No Insulation 54
Brick Veneer construction
with a layer of 16mm (5/8")
plasterboard. With Insulation 56
STEEL STUD
No Insulation With Insulation
1 layer of 16mm (5/8")
plasterboard each side of a
steel stud. 37 42
Stud sizes
38 42
51mm
64mm 39 43
76mm
92mm - 46
150mm
42 46

2 layers of 16mm (5/8")
steel stud. 46 51
Stud sizes
46 52
51mm
64mm 48 53
76mm
92mm 48 53
150mm.
51 55
steel stud. 50 54
Stud sizes
51 55
51mm
64mm 52 56
76mm
92mm 53 57
150mm.
55 58

1 layer of 16mm (5/8")
staggered steel stud. 41 47

staggered steel stud.
52 58

plasterboard each side of a 55 60
staggered steel stud.
Double steel studs opposite

each other with 2 layers of
No Insulation 55
16mm (5/8") plaster board
on each side.(Insulation is
125mm (5") glass wool With Insulation 59
25kg/m2)
Single 150mm (6") Steel
stud with 3 layers of 13mm
(1/2") plasterboard on one No Insulation 55
side and on the other side
another 3 layers on a
flexible channel screwed
horizontally onto the steel
With Insulation 64
stud. (Insulation is 125mm
(5") glass wool 25kg/m2)
BRICK and CONCRETE BLOCK
Standard house brick

unrendered. 39
House brick rendered both

sides (13mm (1/2")
1:1:6,Cement:Lime:Sand)
45
8" Concrete Block, hollow

core, no wall finish 45
8" Concrete Block , hollow

core, painted 2 coats, 2
sides
48
CEILINGS
1 x Layer of 16mm (5/8")
Plasterboard fixed to a
flexible channel fixed at
400mm (16") centres.
(a) 45
(b) 49
(c) 45
2 x Layers of 16mm (5/8")
(a) 56
(b) 58
(c) 55
3 x Layers of 16mm (5/8")
(a) 56
(b) 59
(c) 55
I am indebted to the people at Boral for their excellent web page and
the TecSYS CDROM they sent me free for the figures presented above
for plasterboard installations.
FLOORS
Adding a floating floor can dramatically increase your sound isolation because it disconnects the
whole room structure from the rest of the building. This is especially necessary when isolation of
low frequencies such as bass drums and bass amps is required.
There are two ways of floating a floor.
Floating Timber Floor

Floating Concrete Floor
The floating timber floor is the more typical for a home studio whereas commercial studios
(usually built in commercial buildings) usually opt for the floating concrete floor.
Floating timber floor

This consists of laying a double layer of 16mm (5/8") plywood or particle board flooring on 100
x 50 (4" x 2") joists that have neoprene pads placed at the points where the original flooring
joists are. Rockwool is placed in the cavity between the joists to dampen any resonance.
FLOATING TIMBER FLOOR
Floating Concrete Floor

Floating a concrete floor also gives excellent isolation. You can suspend the floor:
on proprietary spring loaded suspension pads that are commercially available

or you can float it on plastic which sits on a double layer of fibreglass with a layer of
fibreboard sandwiched between the sheets.
FLOATING CONCRETE FLOOR
As you can imagine the concrete compresses the fibreboard between the sheets of fibreglass
and creates a multimedium isolation barrier. The concrete has a reinforcing steel mesh laid in it.
Cable Ducts
Don't forget to place your cable ducts in whatever floor structure you choose. There is nothing
worse than cables all over the floor in a studio because they forgot to lay proper cable ducts.
The ducts can be standard poly drain pipe or you can create a wide shallow duct in your
formwork. It is advisable to run your power down one duct and your audio down another or split
your one duct into two as shown.
CABLE DUCT
When your cables go through walls make sure that you seal around them or all the work you
put into creating a sealed room will be lost.
SHAPE AND SIZE
So what actually happens to speakers setup in the open air? Firstly, the direct sound from the
speakers is all you hear. There are no reflections coming from walls, ceiling, floor etc. Secondly
no sound comes back to you a second time from a rear wall. In open air there is no
reverberation time yet every room has a reverberation time. You can work out the
reverberation time of your room by using the Reverberation Calculator Page.
Speakers Outside
Speakers Inside
As you can see the walls really muck up the sound waves because they reflect back off the wall
behind the speakers and the reflected waves arrive at the listener in and out of phase. As a
result they add and subtract from each other creating confusion in the bottom end. Also the
highs and mids are going to reflect off the walls and the room rear wall.
STANDING WAVES
Another problem created inside is parallel walls and standing waves. Standing waves are when a
sound reflects off walls that are opposite each other and a wave equal to the distance is formed.
As you move around a room with standing waves you can hear as you walk in and out of a
standing wave. In one spot the bass is booming yet in another there is hardly any bass. Makes
it hard to figure out how much bass you have? It works like this:
Typical Standing Waves
What happens here is that if you stand at the high point of an in phase standing wave you
hear double the volume of the frequency yet when you stand at the same point in an out of
phase standing wave the waves cancel each other and you hear nothing. It's pretty hard to
figure out your sound frequency balance when this happens throughout your control room. It
happens at all the octaves of the frequencies as well so if the frequency is 440Hz it also happens
at 880Hz , 1760Hz ,3520Hz,etc. This is what creates coloration in the room. As you move
around the room the frequency response keeps changing causing room coloration. It's also a
problem in the studio if your mic is sitting directly in an out of phase node!! So the first thing
you must do is eradicate all the parallel walls in a studio design. I believe that a wall must be at
least 12 degrees off parallel to stop parallel wall standing wave interference. That's either one
wall at 12 degrees or two walls at 6 degrees each. If you can afford to make the angle
bigger, do so. Also you can create angled walls within a rectangular room by adding
acoustic treatment. (I'll demonstrate that in the pages on wall treatment). (Note: having non
parallel walls doesn't entirely stop standing waves - they still form within a room but along
different lines of repetition). The main reason for having angled walls in a control room is
because of reflection control of the high frequencies for true imaging from your speakers.
This effect also happens between the floor and the ceiling so to stop the effect you must angle
one! well it's not going to be the floor is it. I must state here that putting angles into the ceiling
is expensive so I would recommend that for the home studio you use the acoustic treatment to
break up the ceiling. (See wall and ceiling treatment pages)
ROOM MODES
The formula for determining the fundamental frequency of a standing wave for a particular room
dimension is:
f = V / 2d
f = Fundamental frequency of the standing wave
V = Velocity of sound (343m/sec (1130 ft/sec)
d = Room dimension being considered in feet (length, width, or height)
Other standing waves occur at harmonics of the fundamental frequency - that is 2, 3, and 4
times the fundamental.
Fundamental Frequency Calculation
Enter the value of one of your room dimensions and you measurement system.
If you are working in:
Metric insert the speed of sound as 343 m/sec.

Feet and inches insert the speed of sound as 1130 ft/sec.
Then click on any other field and all fields will be calculated
Please Note: You might have trouble if you are using Netscape!!
Speed Of 343 Metres/Feet/ per

Sound sec
Room Length Meters/Feet
Fundamental Hz
1st Harmonic Hz
2nd Harmonic Hz
3rd Harmonic Hz
RESET
Thus a room with an 6 metre dimension has standing waves forming at
27.5Hz (the fundamental frequency )

55Hz (the first harmonic),
82.5Hz (the second harmonic)
110Hz (the third harmonic)
yet a room dimension of 3 metres gives
55Hz (the fundamental frequency )

110Hz (the first harmonic),
165Hz (the second harmonic)
220Hz (the third harmonic)
In other words rooms with dimensions that are multiples of each other create similar room
modes - so avoid room shapes with dimensions that are multiples of each other.
The Kick Drum: It is also interesting to play with the calculator in other ways. Try entering the
measurement of a 24" - (2 ft) (0.600M) kick drum. A kick drum being a circle is a continuous
parallel wall and it will have a fundamental frequency. You will find that the fundamental
frequency of 20" - 24" bass drums is around 300Hz - wow ! do you recognise that frequency
when equalising kick drums??
So what size should a control room be? These days people are building bigger and bigger
control rooms because so much happens in the control room now. Often the bass player and
keyboard player actually sit in the control room. I often have the vocalist singing in the control
room while the bed tracks go down, so a good size control room is a good idea.
So many people think that you need a big studio and end up putting a poky little room at the
end and call it a control room. I suggest that a control room ideally should be at least 6m x 5m
with a minimum ceiling height of 2.4m. This size room means a good sized working area with
space for the musos, friends and hangers on. Also because you don't want rear reflections to
interfere it is better to start with a longer front to back dimension than the side to side
dimension. i.e. 6m x 5m.
The studio, on the other hand, requires a different set of considerations. The first to consider is
do you want just one room or more? The trend nowadays is for more than one studio so the you
can get isolation between players and different acoustics for each room. The acoustics you want
for a live drum sound is totally different than you want for a vocal for example. I recommend at
least 2 rooms. You can put drums in one, guitars in the other, keys , vocals and bass (DI) in the
control room or maybe the band in one and the vocalist in the other. Getting separation
between guitars and drums is usually OK in a good sized room but keeping them out of the
vocal track is hard if they are all in the same room. Just a note here - the kitchen is an
important room to consider. Gallons of coffee and tea are drunk in studios. An engineer was
once asked - how do you normally have your coffee? "Cold with a fly in it !! "- was the reply.
Also a kitchen allows for a fridge for the beer and a microwave for the pizza, vital ingredients in
every album recording session. Toilets (bathrooms for you in the US) are also a vital service a
studio should offer. The coffee and beer has to go somewhere.
RECORDING PIANOS & ORGANS
The Grand Piano

The piano is really just a guitar (or more accurately a harp) lying on its side. It has strings stretched
between two bridges, a striking area where it is hit with a soft hammer, and a sound board below.
The drawing above shows the main areas of concern when recording a piano. The mikes can be placed in
any of the position A - D as well as underneath. The sound holes give you access to the sound board
below the strings. So lets look at each position.
Position A
Position A is the most typical mike position used in studio music recording. It's a stereo pair that is about
150cm(6") apart, placed over the hammers with one pointing to the lower strings and the other directed
toward the high strings. They should be about 150cm(6") above the strings. They are placed just behind
the music stand. If there is no music involved the music stand can be removed giving a cleaner access to
the strings. If these two mikes are placed correctly you can achieve a really good stereo image where the
low strings appear from the left and the notes follow to the high notes on the left. Because the mikes are
over the hammers the notes are bright and have a nice attack.
Position B
Position B utilises the hardness of the bridge and can be used to emphasise the low strings. I often add a
small amount of it to the left of the image to accentuate the bass strings. Great if you have a 7 or 9 foot
grand!!
Position D
Position D is the traditional Classical way of recording a grand piano and is still used today when
recording grand pianos with an orchestra. It can also be used to add body and warm to a position A
setup.
Position C
Position C is a position that accesses the sound board. The level coming off the sound board is quite high
so it is a good position if you are caught having to record a grand piano in a studio with other instruments
and you want separation from the other instruments. Two mikes places in the sound hole s allow you to
lower the piano lid to the lower stand and with a couple of blankets or sleeping bags thrown over the lot
you will get good separation yet a clean sound that will sound even better with a bit of high shelving
added. Another way of accessing the sound board is to place a mike under the piano pointing straight up.
This is often used in TV where they don't want the mikes to show.
PZM Mikes
There is one more way of recording a grand and that is to use 2 x PZM mikes fixed to the lid and then the
lid closed. This produces a beautiful clean sound and is also great if you have spill problems, hence there
use on stage shows. Give me two good Neumans or AKGs and I'll go with them anyday though. The AKG
451 is my favourite.
The Upright Piano

Unfortunately most home studio owners don't have a grand piano but lots of you have an upright. So
what's the best here - well - treat it like a grand. It has all the same spots.
Here we have a typical upright piano with the typical three positions. To access some of these positions
you may have to pull the piano apart. The front panel above the keys can easily be removed as can the
panel below the keys. The easiest and simplest is to simply drop two mikes on boom arms through the
top and set them up as a stereo pair as in the grand piano over the hammers and about 15cm(6") apart
and pointing left right. This placement is easier if you can remove the front panel.
Position A
This is the standard position as described above and can be supplemented with either a rear soundboard
mike (out of phase) or a lower mike in a sound hole under the keys. In this case the lower front panel
must be removed. Try and avoid getting the lower mike too close to the pedals as their sound will become
annoying.
Position B
Position B is the one used on the old TV shows where they didn't want you to see the mike but it also has
a lot of body and warm in the sound so when incorporated with position A it can be helpful.
Position B
Position C is a variation of position B except that it can also incorporate the harder bridge sound.
Personally I would go for position A every time and would only use the other positions to supplement the
sound or because I can't get into the piano and can't take off the front panel.
PZM Mikes
Once again two PZM mikes strapped to the front panel at the height of the hammers will work very nicely
indeed.
The Organ
The Hammond Organ is another beast altogether and although it's not a stringed instrument I'll include
it here. The Leslie box consists of a divided cabinet. In the top section is a rotating horn covering the high
frequencies from around 800Hz up while below is a woofer cabinet covering the lows. The woofer also has
a wooden horn shape that rotates. This is how I like to mike a Hammond Leslie Box.
The rear of the Leslie cabinet will need to be removed, its only a few screws. The microphones are
basically two stereo pairs which you pan L/R. If you wish to be really mad you can carefully put one of the
high frequency mikes into the cabinet like this:
This really gives a great stereo effect and the Leslie rotates in your head with headphones. The top mikes
are totally 180 degrees out of phase but who cares, the effect is great.
Incidentally, if you want that incredible Emerson Lake and Palmer growl from the Leslie, remove one of
the large output valves that I've drawn in the picture above. Some people have modified their Leslie
cabinets so you can plug a guitar directly into the valve amp so you can get a real Leslie effect on a
guitar. If you also remove the output valve you'll get the wildest guitar grunge!!
RECORDING VOCALS AND HARMONIES
I suppose the recording of vocals is the one area of recording where the nerves start to
automatically step in. Deep down everyone knows that the recording will sink or swim on the
ability of the vocal to sell the song. Sure, the other musicians have had their own stresses and
strains throughout the recording process, but everyone knows that the public listen firstly to the
song and the vocal, but before we get into that we must decide what vocal mike to use.
The Vocal Mike

The choice of vocal mike is really dependent on what mikes you have in your mike cupboard. I
believe it's a good practice to setup your best mikes in a row and have the singer sing into each
one and then compare them all. What worked yesterday for one singer mightn't work with
another. So make sure you start off with the best mike for the situation. Let's face it, any of your
best mikes will do the trick. I've heard great vocals recorded on a SM58.
Wind Shields
The use of wind shields or pop filters as they are sometimes called is a good idea. If you want the
singer to be in your head they have to be close to a mike, and I mean close, like 2.5 - 5cm(1" -
2") away. Therefore a wind shield is a good bet. P's, B's, C's, all produce a jet of air and it's that
shot of air that causes the diaphragm to bottom out as it were and produce a pop sound.
Wouldn't you be pissed off if you got the great performance but there was a big POP in the
middle. (Sure you can whip it into soundforge and EQ it out but that's another story) The problem
with windshields is that they effect the top end response but if you use an appropriate material it
shouldn't effect the sound too much. You can purchase windshields but they are easy to make.
Make a wire hoop out of an old coathanger and tape a strip of nylon stocking over it and gaffa it
to a mike stand. There you are - a great wind shield! Incidentally I have found that PZM mikes
don't pop even when extremely close and they don't have proximity effect. Check out this
factor if you haven't already.
Presence
Presence on a vocal is important. A big factor is the room the singer is in. If it's live there will be
a lot of room sound in the sound but if they are in a dead room it will be lower. I prefer a dead
room for vocals.
Headphones
The singer can use either headphones or speakers to sing too but the usual way is to use
headphones. The most important factor here is to make sure that the sound and balance is right.
Please make sure that the headphones are in stereo!!!! It's a good idea to spend the time to get a
good stereo mix of the track with reverb and effects in the singers cans. Secondly make sure the
singer can hear themselves clearly above the bandtrack and give them some nice reverb, it
makes so much difference. Some singers like to sing with one can off (tucked up against their
head) so that they can hear themselves properly. If your headphone balance is correct this should
not be necessary.
Speakers
Alternatively, you can get the singer to sing to a pair of speakers, either in the studio or in the
control room. In this case their will be bandtrack spill into the vocal track but if you position the
mike between the singer and the speakers with the speakers off axis (180 degrees) to the mike it
shouldn't be too bad. Personally I prefer headphones.
Compression
You should consider using a compressor when recording vocals. I like to, not heavily but enough
to keep the dynamic range under control. It's a bit like the windshield, it's a protection. Say
around 3db on peaks at 6-1 to 10 - 1 ratio. Check out the compression page.
Multi-tracking Vocals
Multi-tracking vocal takes is a good idea. Get the singer to record four or five takes on different
tracks. Then you can go through the each take and select the best performance of each take.
That's cheating!!! I hear you say. Well, if it is, every major singing artist in the world is a cheat
because they all do it. Bring up all the vocal takes on your console and set them all at the same
level and assigned to the track you want your final vocal to land up on. Now go write out all the
lyrics on a sheet and mark each take of each line that's OK like this:
Now, playing the track back mute and unmute each take that's OK and bounce them down to the
final track like so: You will note how some get a double tick - I do that when I really like a
delivery.
This could take some time as working out each line, line by line, is tedious but believe me the
singer will love you for it (providing their ego is in tact) because you will have captured the best
they can do and that is what recording is about!! Incidentally, I do all this on a computer now
because it's so much easier, then I archive all the out takes in case I need to revert back to them
later.
The Harmony Vocals

Recording harmony vocals really depends on how many harmony parts there are. You can put
two or three singers on one mike, or two singers on one mike in a figure 8 position with one one
each side or you can give each singer their own mike. It's up to you really, so long as the
outcome is properly balanced parts. Headphone balance with a three part vocal group can be a
problem so I recommend that if one can't hear themselves tell them to remove one can.
Multi-tracking Harmony Vocals

Tracking harmony vocals can be a problem - do I put each part on a separate track or mix them
together etc. It really depends on how many tracks you can afford and if you are going to double
track them. I usually double track harmony vocals as it creates a blend of parts. So I'll mix the
parts straight away onto one track and then double it again onto another. If we then decide that
there is another harmony to add and I don't have lots of tracks free I'll mix the first track with the
new harmony and record it onto another track. Then I'll do the same with the double track and
end up with three parts on one track and three parts on another. Then I'll wipe the first two
tracks to free them up for other things.
Alternatively you can use one singer and put all the different parts onto different tracks and mix it
all down later. Using one singer gives the harmony vocals a character of their own. The blend one
singer has with themselves is great.
So who sings what

Let's say you've got three singers. You can either:
Give each singer a part each and record three part harmony straight out.
Get all three singers to sing one part, then all three sing the second etc.
It really depends on the ability of the singers. The first is the hardest because the singers must be
good to hold their own part, the second way is really good when you have a band where the
backup singers aren't that good plus it's easier to balance.
HIGH FREQUENCY ABSORBERS
Well just about anything absorbs high frequencies so be careful in this area. If you suck out all
the high frequencies in a room your room will become lifeless. The main consideration in room
acoustics is to aim to make the reverberation time near to equal in all frequencies. So if you
put walls and walls of high frequency absorbers you will not have enough wall space to lower
the low frequencies proportionately.
If you look at the coefficient of absorption figures for the various products you will note
that whilst some attenuate the highs some also attenuate the low mids as well. 100mm (4")
fibreglass for example not only absorbs high frequencies but it also works down into the low
mids depending on how thick it is.
The other main factor is what are the highs in your room doing? Consider the fact that your high
frequencies are coming from your speakers which have a directivity factor. In a standard multi -
speaker system the highs are coming from the tweeters or horns. Both these units have a fan
shaped dispersion of around 30 degrees. And create what is referred to as the on axis off axis
effect. Stand in front of a speaker and you hear all the highs but go 30 degrees off axis and the
highs start to reduce to the point that if you are 90 degrees off axis the highs are eliminated
completely (apart fro highs that reach you my reflection from some other surface.)
Take a look at this plan of a control room:

Control Room Plan
The dotted lines indicate the axis of the high frequency projection. Note that the engineer is
sitting on axis to the speakers yet someone sitting to the right of the console is off axis to the
right speaker but still on axis to the left speaker. The high frequencies are reflected by the
opposing walls (in this case glass doors). The idea of this control room design is make sure (by
angling the walls) that the high frequencies from the right speaker are not reflected back
into your left ear.
Once the sound passes the engineer the rear of the control room absorbs the sound and it
doesn't come back to the engineer.
RECORDING PERCUSSION
The main concern with percussion instruments is the extreme transients they produce.
Tambourines and shakers all have extreme transients so beware of getting tambourines too
close to a mike and make sure you meter the peak content in the signal.
Congas
Congas are usually miked with one mike between each conga thus:
The under mikes are an option that can add depth and body to the sound of congas. Like
undermiking toms the mikes must be phase reversed relative to the overhead mikes.
Bongos
Bongos are similar to the congas:
Here once again watch the transients. You can put one mike per drum if you want to spread the
stereo sound for effect and you can also mike them from underneath of you want separation in a
studio situation.
Tambourines, Shakers, Maracas , Bell Tree etc.

Tambourines, shakers etc. produce extreme transients so use a mike capable of handling high
Sound Pressure Levels if you want to close mike them. For shakers and tambourines I like to
have the player stand 3 - 4 feet from the mike so some of the room ambience creates a space
around them.
Vibes and Marimba

Vibes and Marimbas can be either stereo miked or single miked. A single mike need to be higher
than stereo miking to capture the full range of the notes.
The Main thing with percussion is WATCH THE TRANSIENTS
WALLS
In the page on isolation I describe the standard double wall construction on a floating floor with a resultant wall
thickness of 300mm.(1ft). That is 100mm (4")for each wall and a 100mm (4")air space between. The finished
room then has a wall surface of plasterboard that requires further acoustic treatment to handle the reflected
sound within the room. If you then put 100mm of treatment on each wall you end up with a wall 200mm(8")
thick and a total wall thickness including the air space of 300mm (24"). In the home studio space is precious so
a small saving of wall thickness can really help. So I have developed a simple and cost effective solution.
MY ROOM ............................................NORMAL
This system halves the thickness of the wall (100mm(4") instead of 200mm(8")) and makes the 95mm(4")
cavity in the wall available for acoustic treatment. (Slat resonator depicted in drawing) Construction requires
building the wall panels on the floor, applying the outer cladding and gluing the insulation to it then standing
the wall panels up into place. You end up with a wall with the cladding on the outside and a 100mm (4") cavity
for internal acoustic treatment. Obviously the panels must be sealed when connected to each other and if you
are using a double or triple layer you can seal it like this.
Offsetting Sheets for a good seal
CEILINGS
A ceiling can be built using a similar system. Here again any savings in height can benefit the home studio
builder. Support beams are placed on the inner walls and ceiling panels are attached to the beams thus:
CEILING ELEVATION
CEILING PLAN
This is similar to the wall construction where the gypsum is on the outside thus freeing the 95mm(4") cavity for
acoustic treatment.
The beauty of this construction is that it can be made modular. The wall and ceiling panels can be made on the
floor then placed in position and joined together with a silicon seal between panels and screwed together and to
the floor. They could also be made in a joinery factory and then screwed together at the site. This means it can
be disassembled and moved. If you are renting, or shift house, you can take it with you!
The individual wall and ceiling panels can be treated acoustically using different treatments as described in the
acoustic treatment pages.
Sealing an Existing Ceiling
Plans and STC ratings for sealing an existing ceiling can be found on the STC Chart page.
The above drawing shows a typical system for sealing an existing ceiling. The plasterboard is attached to a
flexible channel thus mechanically isolating it from the existing ceiling. For even more isolation you can add an
extra lining under the existing floor thus:
This system gives an STC rating of around 60 which should stop you from annoying the rest of the house and
your neighbours.
ABSORPTION COEFFICIENTS
For an instrument to sound good you must hear a balance of its high and low
frequencies. If your room is too dead in the high frequencies the instrument sounds
lifeless and tubby, whereas if it's too live in the high frequencies the instrument
sounds distant and has no presence. To balance a room you must have a true balance
of the high frequencies and the low frequencies with the right amount of deadness to
liveness.
The coefficient of absorption is the basis of acoustic treatment and is the amount of
sound energy a surface absorbs and reflects and is measured at different frequencies.
If we say that a surface material has an absorption coefficient of 0.25 we are saying
that the surface will absorb 25% of the incident acoustic energy, while reflecting back
75% of the total acoustic energy at the specified frequency. It can be displayed as a
chart like this:
Absorption Coefficient Chart for typical surfaces
If you look at the brick figures you will see that a brick wall will reflect almost all the
incident sound energy whilst a wall covered with 25mm(1") Rockwool will absorb fro
80 - 90% of the high frequencies but only 35 - 60% of the low frequencies. Whilst
most people think that heavy curtains are a good deadener according to the chart they
work only reasonably (45%) at high frequencies and not much at all at low
frequencies.
This is the major problem experienced by studios who treat their rooms by putting
drapes, carpet or egg cartons on the walls. Basically they are lowering the reflections
(reverberation time) of the room BUT ONLY MARGINALLY IN THE HIGH FREQUENCY
RANGE! The lows are still humming around the room and the resultant reverberation
field creates what we call a muddy - woofy sound. The solution is always to add highs
to the recorded signal to return the top end yet really what needs to happen is the low
end (100 - 1000Hz) reverberation time needs to be lowered in the recording room and
probably in the control room for you to even hear the difference.
Alternatively, look at the absorption figures for a product called Insulco Semi Rigid,
50mm (2") thick. This is a product made from compressed fibreglass and only takes
up 50mm (2") of wall space.
Insulco Semi Rigid 50mm
As you can see this stuff really works. It actually absorbs the same amount at 125Hz
that curtains absorb at high frequencies. (These figures actually state that it absorbs
over 100% of the sound, I think the scale was made before products like Insulco were
invented!!) I'm sure overseas visitors to this site can find your local equivalent if you
ask around.
When purchasing insulation ask for the absorption coefficient figures
To see a list of typical coefficient figures click here
These absorption coefficients are measured with the material mounted flat onto a
surface. Another aspect to consider is how to mount the material on the wall. The
following diagram shows what happens if you lift the acoustic material off the wall.
Insulation relative to the wall
As you can see there is an added advantage if you actually lift the insulation off the
wall because the point of maximum energy in a wave is at the highest point in the
wave which is at the 1/4 wavelength point. So if you, for example, create a box frame
100mm(4") deep and mount the 50mm (2") Insulation flush with the front of the box
it will be 50mm(2") off the wall. This will lower the effective frequency of maximum
absorption.
DIFFUSION
Up to this point we have discussed reflected sound off flat surfaces. There are three things that
can happen when sound hits a wall. It can be reflected, absorbed, or diffused. If you have a
multifaceted surface such as a rock wall the sound reflected will be diffused. Diffusion spreads
the reverberant sound evenly throughout a room, which not only prevents standing waves but
also eliminates dead spots, i.e. places where components of the sound are missing through
phase cancellation as discussed in standing waves. Flat surfaces can be broken up by placing
diffusers on studio walls.
Diffusion in the Control Room

I must say at this point that I have a problem with diffusion in a control room. Sure the diffuser
does disperse the sound evenly within the room and it sounds impressive but I've found that
when working in a diffuse control room you get a distortion of the amount of "life" in a sound.
The diffusion makes everything sound airy and open but what's on tape might not have that
factor.
A control room is a working environment, not a listening

room
In a control room you are wanting to hear exactly what is on tape and you want to be able to
analyse it completely so that you can add the necessary components such as EQ,reverberation,
compression etc. Direct sound from the speaker is the aim in a control room and I feel diffusion
clouds that image. I realise I'm about to be criticised for such a view soI leave it up to you.
The shapes chosen for diffusers are really a matter of taste and cost. Avoid concave curves,
which focus sound instead of dispersing it, but otherwise pyramids, lattices, or computer
designed random surfaces all work well. The depth of a diffuser determines the lowest
frequency that will be affected. A diffuser one foot deep will scatter sound down to 160 Hz.
Diffusers can be built by the home studio owner quite simply by creating a multi surface plane.
The typical one is lots of blocks of wood of various sizes glued onto a backing sheet. Go to a
house construction site and ask for all the 100mm x 100mm (4" x 4") offcuts. Glue the blocks of
irregular lengths onto a backing sheet of plywood, spray it with paint and stick it on the wall.
You will now be the owner of a new trendy studio with a diffuser on the back wall of your control
room.
( Sense a bit of cynicism??)
Diffusion In the studio

Diffusion in the studio is a great idea and one of the best way to add it is to have stone walls in
your studio. Not flat stone but round and irregular stones that create a rigid random diffuse
surface. The reverberation created in such a room will be rich and diffuse which is what you
want in a good reverb unit. (Note here that some effects units have a control over diffusion in
their reverb programs)The greater the "depth" of the diffuser the lower the frequencies
affected. I recommend such a wall in a drum room if you want live drum sounds. Otherwise try
the wooden block system, it works really well also
VARIOUS TYPES OF DIFFUSERS

I have called the angled - curved - pyramid shapes absorbers/diffusers because they can be
built as low frequency absorbers yet will also act as diffusers. See the section on low frequency
absorbers. Today various companies manufacture pre - built diffusers that can be purchased
and installed in your studio.
I must say here that the last three diffuser types are pure diffusers and perform no other
function whereas the the first three types can act a low frequency absorbers. In the home
studio your room sizes are usually small and low and low-mid frequency coloration is your main
problem so to waste treatable wall space with just a diffuser to me is a waste of wall space. Slat
resonators also act as diffusers because the slats with the gaps break up the surface and I
would advise you to use them instead. The best place to use diffusers is in a live room if you
have the space to dedicate a room specifically for that.
WINDOWS AND DOORS
Windows and doors require special construction because no matter how much you seal your
walls if the windows and doors aren't built correctly your isolation will be ruined. The main thing
with windows is that they must have the following features:
Different Glass Thickness. It is essential that the two sheets of glass be different in
thickness. I recommend that you put the thicker of the two panes on the control room
side. The thicker the glass obviously the better the sound isolation plus the thicker glass
has a lower resonate frequency. Unfortunately thick glass is expensive. I would suggest
you try 8mm and 10mm glass. (5/16", 3/8"). Any thinner and you are going to start
getting resonate frequencies from the glass and inadequate sound isolation.
Angles. The two sheets of glass must be at an angle to each other else the two sheets
will interact in a resonate sympathy and the sound reduction properties will be reduced.
You can angle the glass as in the following drawing but don't forget that the glass can
also be angled in the horizontal plane as well as the vertical plane.
Silica Beads. Because the windows are sealed the cavity created is a different
temperature and humidity than your rooms which are probably air conditioned. It is
therefore possible for the glass to steam up as in your car but not quite as dramatically.
It is therefore recommended that you purchase some silica beads ( like you get in a little
sachet when you purchase a quality camera or the like) and put them in the cavity
between the glass.
Insulation. The cavity between the glass is like any space and will have a reverberant
field so you must line around the cavity with insulation. The easiest way to do this is to
cut sheets of fibreboard to the shapes and then glue thin fibreglass to it. Then you can
wrap cloth around it for aesthetics and glue it into place. It is also a good idea to drill
25mm - 50mm (1" - 2") holes in the fibreboard in which you can put the silica beads.
The following drawing shows how to construct your windows.

WINDOW CONSTRUCTION
WINDOW CONSTRUCTION DETAIL
DOORS
You can use two types of doors in a studio. Solid core doors or glass doors. Obviously if you
wish to use glass doors the glass, like in the windows above, must be of a reasonable thickness
to stop resonance. I'd suggest a minimum thickness of 8mm (5/16") yet obviously the thicker
the better. Glass doors are good because they increase the communication factor which is
important in a studio but if you are to use a two door sound lock you must have the doors at an
angle to each other or you will get standing waves between them that will reduce isolation.
Hinged Doors
Seals. As with windows once again correct sealing of doors is the main determinant that
effects the sound isolation. Doors must be sealed all round and it is advisable to purchase
proper commercially made door seals. There are a number of different manufacturers of
door seals and I suggest you contact your local supplier. The most important seal is the
one at the bottom of the door as it is the hardest seal to make. Some commercial
manufacturers make a seal that has a spring loading so that when the door is closed a
lever is compressed that causes a rubber seal to be forced downwards on to the door
jam. When the door is opened the seal is lifted again.
Thickness. It is recommended that you purchase solid core doors. If you wish to isolate
you can clad the room side with extra timber that gives a nice finish and increases the
effective sound isolation.
Insulation. Like the window the two doors create a resonate cavity when closed so it is
advisable to line the cavity and the doors with some insulation and cover with cloth.
SEALING DOORS
You can purchase proprietary door seals that fit into the base of the door. The unit has a sprung
button that when the door is closed forces a rubber seal down onto the door jam. When the
door is opened the spring releases the seal.
Sliding Doors
I personally like sliding glass doors in studios because of the visual communication they afford.
Like windows they can't be parallel so I always put them at an angle in the horizontal plane.
(The vertical plane creates unbelievable problems with runners and seals.) Sliding doors can be
made of either timber or aluminium.
Seals. Naturally a glass sliding door will not have the sound isolation of a hinged door
purely because of the construction complexity but if you use a quality door and discuss
the seal problems with your local manufacturer you can come up with a pretty good seal.
Thickness. I recommend you use at least 8mm (5/16") glass but here again the thicker
the better but too thick makes the door extremely heavy to slide.
Insulation. Once again the cavity between the doors must be lined with insulation to
stop the reverberation within the cavity. The same method as in the window (i.e. cloth
over fibreglass over fibreboard is the simplest system.)
RECORDING DRUMS
Have you ever noticed how engineers all seem to start with the drums when recording a band or
mixing a PA. This is probably because the drums are among the hardest instruments to record,
yet the drum sound can dictate the whole sound of the band or the PA Mix. If you are learning,
recording drums can be a real challenge as they are the most complicated instrument - or I
should say - group of instruments.
Every engineer has his or her system of recording drums and they are all probably as similar as
they are different. What I intend to do here is to establish a method for recording drums
because without a starting method you can spend hours mucking around not really getting
anywhere. I started recording drums in the mid sixties here in Australia and I really had no
teachers (as opposed to you in the US who had a rich tradition of recording engineers). I think I
was one of the first to record drums in stereo onto 2 tracks of an eight track with the bass mixed
in. Therefore I had to develop a technique of my own, I used to spend hours in the studio late at
night with drummer friends and play around trying every mic placement possible until I
developed a system that we could apply in all circumstances. At the time we were recording
commercials and television show bands (Brass, strings, saxes etc all in) continually day in day
out.- I hope that this method assists your recording techniques.

RECORDING VOCALS AND HARMONIES
I suppose the recording of vocals is the one area of recording where the nerves start to
automatically step in. Deep down everyone knows that the recording will sink or swim on the
ability of the vocal to sell the song. Sure, the other musicians have had their own stresses and
strains throughout the recording process, but everyone knows that the public listen firstly to the
song and the vocal, but before we get into that we must decide what vocal mike to use.
The Vocal Mike

The choice of vocal mike is really dependent on what mikes you have in your mike cupboard. I
believe it's a good practice to setup your best mikes in a row and have the singer sing into each
one and then compare them all. What worked yesterday for one singer mightn't work with
another. So make sure you start off with the best mike for the situation. Let's face it, any of your
best mikes will do the trick. I've heard great vocals recorded on a SM58.
Wind Shields
The use of wind shields or pop filters as they are sometimes called is a good idea. If you want the
singer to be in your head they have to be close to a mike, and I mean close, like 2.5 - 5cm(1" -
2") away. Therefore a wind shield is a good bet. P's, B's, C's, all produce a jet of air and it's that
shot of air that causes the diaphragm to bottom out as it were and produce a pop sound.
Wouldn't you be pissed off if you got the great performance but there was a big POP in the
middle. (Sure you can whip it into soundforge and EQ it out but that's another story) The problem
with windshields is that they effect the top end response but if you use an appropriate material it
shouldn't effect the sound too much. You can purchase windshields but they are easy to make.
Make a wire hoop out of an old coathanger and tape a strip of nylon stocking over it and gaffa it
to a mike stand. There you are - a great wind shield! Incidentally I have found that PZM mikes
don't pop even when extremely close and they don't have proximity effect. Check out this
factor if you haven't already.
Presence
Presence on a vocal is important. A big factor is the room the singer is in. If it's live there will be
a lot of room sound in the sound but if they are in a dead room it will be lower. I prefer a dead
room for vocals.
Headphones
The singer can use either headphones or speakers to sing too but the usual way is to use
headphones. The most important factor here is to make sure that the sound and balance is right.
Please make sure that the headphones are in stereo!!!! It's a good idea to spend the time to get a
good stereo mix of the track with reverb and effects in the singers cans. Secondly make sure the
singer can hear themselves clearly above the bandtrack and give them some nice reverb, it
makes so much difference. Some singers like to sing with one can off (tucked up against their
head) so that they can hear themselves properly. If your headphone balance is correct this should
not be necessary.
Speakers
Alternatively, you can get the singer to sing to a pair of speakers, either in the studio or in the
control room. In this case their will be bandtrack spill into the vocal track but if you position the
mike between the singer and the speakers with the speakers off axis (180 degrees) to the mike it
shouldn't be too bad. Personally I prefer headphones.
Compression
You should consider using a compressor when recording vocals. I like to, not heavily but enough
to keep the dynamic range under control. It's a bit like the windshield, it's a protection. Say
around 3db on peaks at 6-1 to 10 - 1 ratio. Check out the compression page.
Multi-tracking Vocals
Multi-tracking vocal takes is a good idea. Get the singer to record four or five takes on different
tracks. Then you can go through the each take and select the best performance of each take.
That's cheating!!! I hear you say. Well, if it is, every major singing artist in the world is a cheat
because they all do it. Bring up all the vocal takes on your console and set them all at the same
level and assigned to the track you want your final vocal to land up on. Now go write out all the
lyrics on a sheet and mark each take of each line that's OK like this:
Now, playing the track back mute and unmute each take that's OK and bounce them down to the
final track like so: You will note how some get a double tick - I do that when I really like a
delivery.
This could take some time as working out each line, line by line, is tedious but believe me the
singer will love you for it (providing their ego is in tact) because you will have captured the best
they can do and that is what recording is about!! Incidentally, I do all this on a computer now
because it's so much easier, then I archive all the out takes in case I need to revert back to them
later.
The Harmony Vocals

Recording harmony vocals really depends on how many harmony parts there are. You can put
two or three singers on one mike, or two singers on one mike in a figure 8 position with one one
each side or you can give each singer their own mike. It's up to you really, so long as the
outcome is properly balanced parts. Headphone balance with a three part vocal group can be a
problem so I recommend that if one can't hear themselves tell them to remove one can.
Multi-tracking Harmony Vocals

Tracking harmony vocals can be a problem - do I put each part on a separate track or mix them
together etc. It really depends on how many tracks you can afford and if you are going to double
track them. I usually double track harmony vocals as it creates a blend of parts. So I'll mix the
parts straight away onto one track and then double it again onto another. If we then decide that
there is another harmony to add and I don't have lots of tracks free I'll mix the first track with the
new harmony and record it onto another track. Then I'll do the same with the double track and
end up with three parts on one track and three parts on another. Then I'll wipe the first two
tracks to free them up for other things.
Alternatively you can use one singer and put all the different parts onto different tracks and mix it
all down later. Using one singer gives the harmony vocals a character of their own. The blend one
singer has with themselves is great.
So who sings what

Let's say you've got three singers. You can either:
Give each singer a part each and record three part harmony straight out.
Get all three singers to sing one part, then all three sing the second etc.
It really depends on the ability of the singers. The first is the hardest because the singers must be
good to hold their own part, the second way is really good when you have a band where the
backup singers aren't that good plus it's easier to balance.
RECORDING STRINGED INSTRUMENTS
So what is similar about all stringed instruments? Basically, all stringed instruments work on the same principle of a
tightened string between two points thus:
If you check out a guitar and a piano they are both the same. Both have a tunable string stretched between two points. In
the guitar the sound board is the flat front of the guitar where the bridge is mounted, in a piano there is a sound board
below the strings serving the same purpose. The sound actually varies depending at which point you place the microphone,
so that the sound near the bridge end is different from the sound in the middle of the string:
These three different mike positions all present a different aspect of the sound.
The microphone over the bridge is harder sounding and has more of the mid frequencies than the low frequencies.

The microphone over the middle of the string has a full version of the sound and when placed near the striking
method area has the transient sound of the string and the strike.

The microphone near the soundboard has no string impact sound but has all the body of the sound.
The way the string is vibrated or striking method - also determines how it will sound.
You can strum it as in a guitar.

You can hit it as in a piano.

You can bow it as in a violin.

You can pluck it as in a harp.
Either way, the three positions of miking it will always apply.
Select the different pages from the index list above and
see how this principle applies to each instrument.
RECORDING EQUIPMENT
Since posting the Recording Studio Design site on the web I've been asked by numerous e-mails
to suggest what gear the writer should purchase to establish their studio. What a question! In
this section I will cover the three main pieces of equipment a studio will need. Understanding
what each unit does and what to expect of it can quickly assist decision making.
Microphones are your front line in the studio. If you have the wrong type of microphone on an
instrument you are starting on the wrong foot right away. Understanding the different types and
how they work can help you decide what you need.
Consoles come in all shapes and sizes so I've described the workings of a console from the mike
input to the monitor output. You probably can't afford the super console but if you know what
each section does and how it is applied you can work out what your minimum requirement and
go looking for that or you've already got a console that mightn't have everything but this section
will help you understand how to use what you've got properly.
The Recorders section describes the different types of recorders and data storage systems
available as of now which includes analogue tape, digital tape, and digital hard disk - tomorrow
could be a different story. I've also covered Digital Sound. We are entering a totally new era in
sound recording since the advent of digital sound, it's not perfect yet, but I believe it very soon
will be.

MIXING
Without doubt the hardest part of recording is mixing, yet it is also the most enjoyable as this is
when everything starts to come together and all the hard work justifies itself. A good mixer
paints a picture in sound that attracts the listener and conveys the song clearly and simply. I
could sum up a good recording as a series of priorities which are:
The song
The singer
The "feel" or "groove"
The "fiddly bits"
The song is set from the start and a good producer will have chosen a song that has 'something
to say' and a good mixer will convey that something to the listener.
The singer is the next most important aspect and a good mixer will allow the singer to be heard
and the lyrics to be conveyed clearly but with style. There is nothing more annoying than
hearing a track and not being able to distinguish the lyric amongst a babble of instrumentation.
Fortunately you don't hear recordings like that on commercial radio as they just don't get a look
in. Engineers are often guilty of cluttering up tracks with all sorts of tricks and garbage that
distract from the song and the singer because they know the song so well after days in the
studio that they think everyone hears it like they do. If the track is to have a chance of
commercial success it must be understandable from the first hearing. Always underestimate
the ability of the listener as they are not professional listeners like you.
The "feel" or "groove" is what catches the listeners attention initially and sets up the mood
and emotion of the track. This is created by careful balancing of the rhythmic aspects of the
track be it drums, percussion or a great guitar groove.
Finally there is the "fiddly bits" as I call them; they are the musical phrases linking lyrics,
joining verses to choruses and filling solo sections etc. that are created by the guitar licks, the
piano fills, the answering vocal phases etc.
So where to start?
Monitoring Speakers
Monitoring speakers come in two types. Nearfield and Main. I like to use both. I work
primarily on the nearfield to establish my balances etc. and then every now and then I will
switch it up to the big speakers as they give a better idea of the low frequency balance, plus it
sounds good eh! (I was a Yamaha NS10 freak for years but now I'm totally sold on the Event
20/20. Well done Event!) To me a big speaker system is like a magnifying glass, it blows the
sound up and you can hear more but for a big system to be really good you have to flush mount
them and have good speakers and a good amplifier system. Can I say here that I don't like
equalised speaker systems. If they don't sound good flat, get another speaker!!
Level Structure
The first important procedure is to setup your console for mixing. The first requirement is to
setup your levels to and from your master recorder, usually a DAT. If your console has an
oscillator send tone to the DAT and balance left and right channels. Then check that the return
to your console, which is what you'll monitor, is balanced correctly left and right. At this stage it
is also recommended that you insert your master compressor either in the master stereo output
inserts or inline between the console and the DAT and line up correct left/right balance. This
procedure is very important as it effects your level structure from then on and if you don't do it
now you can end up with your levels all over the shop later.
Aux Sends and Returns

Next you must establish your auxiliary sends and returns.
One of the best ways to get perspective and separation within your mix is to what I refer to as
"putting everyone in their own space". You can achieve this through the use of reverb and
effects. I like to have one reverb unit dedicated to the drums. No other instruments are sent to
this effect, only the drums which will put them in their space. The choice of reverb for drums
depends entirely on the track but I start by putting reverb on the snare and going through the
presets to find the one that works best for the track. I find it usually ends up with a bright
reverb of shortish length around 1 - 1.2sec reverb time.
Note: A very fine producer in OZ was once quoted as saying "Give me a studio with 10
Midiverbs over a studio with one Lexicon 224XL" We all know what fantastic units the Lexicons
are but if it's all you've got you are limited to only one perspective.
Next I'll dedicate a reverb unit to act as my overall reverb effect. I look for the best (not
necessarily most expensive) unit in the studio for this will be my master reverb for vocals etc.
In the example above there are 6 sends with 5 & 6 being an option over 3 & 4. I therefore like
to use 1 for my drums and 2 for my master verb. Then I can assign the others for effects. I do
this so that I can always add master reverb as well as effects if necessary and if I had used say
3, I couldn't put master verb on channels where the effect was assigned to 5. Should I use a
stereo or mono send to the effects?? To be perfectly honest I don't think it matters. Most of
the stereo input reverb units I find have a mock stereo input, not a true stereo. If you use two
sends it really doesn't make a difference unless you are working with the more expensive units
like the aforementioned Lexicon, and even then I question the validity of two inputs especially if
you are limited in the number of sends.
I then assign the sends 3 - 6 to additional effects like delay, pitch change etc. to act as
perspective enhancers. When establishing delays I set them to the track tempo. See Tempo
Chart. The idea is to add these perspective effects so you only just hear them when in solo
and they appear to disappear when mixed into the track. Bob Clearmountain - the world famous
mixer - always had two delays going, one on eighths and the other on 16ths. It puts an air
around instruments and if mixed in correctly you won't actually hear them, just sense them.
Pitch change is another effect to consider with say the left channel set to -.008 cents and the
right to +.008 cents. This effect is great on harmony vocals and it puts them in a different space
form the lead vocal. Finally a soft flange or chorus is another effect I'll have as an option for
guitars etc. See Effects pages for settings.
Make sure that all your effects are returned through the effect returns and assigned to the
master stereo output. If you are fortunate enough to have spare channels on your desk you can
return your delay and chorus type effects back through a console channel as this gives you the
option of adding master reverb to them and using the channel EQ. Delays can soften if master
reverb is added to their returns plus you can attain your feedback from the console instead of
using the control on the effect unit. Say you are using send 3 to a delay unit you can feed back
to the delay by sending the send 3 on the return back into the unit. N.B. Incidentally, make sure
that the dry/wet or mix controls on your effect units are set to wet as you are only wanting
the effect from the units and you won't need any dry sound. (If you are using the Alesis
Quadraverb check this as all the default settings have 50% dry and 50% wet.) The returns from
effects are usually panned full stereo L/R, but you may wish to bring the drum reverb back half
L/R to separate the two.
Your console should now be setup like this

Mixing
Some mixers start with the drums, others start with the vocal. I must admit I start with the
drums as they convey the dynamic of a song. Hopefully you will have automation on your
console, if not, you must now start setting up a series of moves and remember where and how
they occur because, let's face it, the balance within a mix is not static, it varies continuously
throughout a song. For example lets say the drummer plays a rimshot snare through the verses
and full snare in the chorus. The EQ required on the rimshot snare sound is probably different
from the chorus snare sound so I often split the snare return from the recorder into two console
channels so I can EQ and effect each separately and automate the switch between the two. For
example, the snare in the chorus will probably require more reverb than the rimshot so having a
separate channel allows for that. Automation also allows for the tom mikes to be muted when
not needed thus reducing the spill of the rest of the kit and cutting out the constant ringing of
the toms which occurs with undampened toms. The overhead mikes also will need to be ridden
throughout the track, I tend to lower the overhead mikes when the rimshot is playing to achieve
a tighter sound, then I lift them in the chorus when the full snare comes in. Reverb on the
overheads gives reverb on the cymbals but it also adds reverb to the snare in the chorus and
lifts the whole ambient sound of the kit. This has the effect of changing the perspective of the
drums in a mix. You can also change the perspective by putting master reverb on the overheads
which blends with the drum reverb.
Once we have achieved a reasonable balance of the kit and the dynamics are set in place we
can add the bass. The bass and the kick drum will determine the bottom end of the track so the
balance between the kick and bass is critical. The kick will give the bass punch and attack when
they hit together.
Note: I must say a few words here about bottom end. The big mistake in mixing is to make the
bottom end sound too big by adding lots of bottom end EQ to the kick and the bass. You must
bear in mind how the track will be played back by the listener. Nowadays everyone has a stereo
system with bass boost as an option either as a loudness switch or as a sub bass control.
Everyone who has this option has it switched on!! If you get out a few of your favourite
recordings and listen to them on your mixing speakers you will find that they are relatively shy
in the bottom end and yet when played through your average boom box sound tight and fat.
You have to start to understand what a flat response really means and learn to mix that way.
If you put a bass on a VU meter you will notice how much energy there is in the bottom end. A
bass peaking to zero will have the same apparent loudness as a highhat peaking to -30db.
That's because a hithat has no real bottom end compared with a bass so be careful with your
low end EQ on basses and kick drums. I like to solo the two together and EQ them so that they
are tight but not boomy.
Add the vocal

OK, so the bass and drums are now at their first mix level so next I will add the vocal and mix it
sitting just above the bass and drums. This might mean an EQ change so they all sit tightly
together. The vocal might need to be ridden with the automation and I'll probably compress it
again to keep the dynamic range within the boundaries of the whole track. I often find that the
reverb on the vocal will need to be ridden so that the screaming high notes need more reverb
that the quiet intimate sections in the verse. Here I take a feed from the direct out of the
vocal channel and bring it up on another channel on the console. I then deselect this channel
from the stereo mix output so it goes nowhere but the aux sends are still working. By adding
reverb to this channel I can use this channel to ride the reverb on the vocal as an automated
send.
Adding the rest
Now we can start to add the fiddly bits like the rhythm guitar and keyboard pads etc. adjusting
their balance to fit tightly but not overpowering the vocal. (Please understand I am not
defaming guitars etc. by calling them fiddley bits, they are just as important as every other
part) The track should now be starting to take shape. If the dynamics of the drums and vocal
have been set correctly the placement of the additional instruments will fall into place easily.
The vocal harmonies, and solo instruments can now be mixed into the track and we are nearing
the completion of the first mixdown.
Note: It is important to keep checking your mix in mono. Unfortunately stereo and mono are
not compatible. When you switch to mono, instruments that are panned centre are 3db higher
than in stereo so your vocal, kick and snare, for example, will come up in the mix. Some
engineers actually make two mixes of a track: One that is full wide stereo with full dynamic
range for home listening and one where all the hard left and right signals are panned to the
centre or half centre and compressed for radio. It's really hard because if you make a mix sound
great on a good home hi-fi it won't have the tightness and punch a mix made for commercial
radio will have where the dynamic range is low. It's common practice to make separate mixes of
the singles from an album for radio whereas the remaining tracks are mixed totally for home hi-
fi. I think you will find that most commercial records are mixed to sound great on FM Radio.
Rest and Recreation

It is important that you constantly give your ears a break during the mixing process as your
ears have little compressors in them that will progressively shut your ears down. Have you
noticed that when you've been in a loud club with a loud band when you go outside you can't
hear as well. It's part of your ears protection system and a cup of coffee in another room
watching TV or something will allow them to start opening back up. I like to "mix from the
kitchen" as I call it. This means playing the automated mix and listening to it from an adjacent
room with the control room door open, you'd be surprised how clearly you can hear the balance
between instruments when you get away from the direct sound from your speakers. The
relationship between the bass and kick, the balance within the harmonies, the clarity of the
vocals etc. all become clearer when you relax and listen from another room.
Monitoring Level
Unfortunately the human ear is not flat at all levels. Some guys called Fletcher and Munson
worked out what the response curve of the ear was and found that at low levels the ear missed
out on the low frequencies and the high frequencies, whereas at loud levels it was the opposite.
From the above chart you can see that around 80 - 90db the ear is the flattest. The fact that we
don't hear low frequencies and high frequencies at low levels created the Loudness switch on
stereo systems which boosts the low and high frequencies to compensate for the ear.
Unfortunately, Joe Public doesn't know this but knows that when it is switched in things sound
fatter and brighter so they leave it in all the time. It is generally recognised that a level of 85db
is where the ear is at it's flattest so don't mix too loud if you want a flat response.
The important thing about mixing is apparent loudness, or relative loudness. If I whisper into
a mike and then I shout into a mike the shout will appear louder because I know that shouting
is loud. It's the same with mixing. You create an illusion of loudness, everything is relative. You
can't get bigger if you are already at your maximum. If I mix a soft acoustic guitar and vocal
and peak to zero then bring in a full kit and grunge guitar also peaking to zero it will apparently
get louder because I know that drums and guitar are loud. Mixing is the art of making signals
that all peak to zero sound as if there is a dynamic range. Nowadays with the excellent
compression systems we have most recordings are heavily compressed. I was told of a producer
who hired a mixing engineer to mix an album. The guy turned up with racks and racks of
compressors and set about compressing every track. He had one compressor for this and
another for that etc. In the end the whole mix was pumping away and almost mixed itself. That
album went on to sell millions of copies world wide. Those of you who have played with Waves
Ultramaximiser will know what compression can do for a mix. If you watch most modern pop
recordings on a VU meter the needle is almost static varying only a few db yet the tracks go
from quiet intros to full on chorus and solo sections yet still there is only a small variation in
level. So setting compression (and limiting) levels is important. I will always have a compressor
across the output of my mixes as it helps control the peaks and brings up the loudness of the
track but I may use individual compressors on separate channels.
Finally - do take the time to get a good mix. If you don't you have not given justice to all the
effort you put into recording it in the first place. It may take a few remixes, so what - it's the
final product that counts.
ACOUSTICS
So you want to setup a studio so why worry about acoustics
and what are acoustics anyway?
Let me tell you a story as I heard it from the late Dean Jensen who I stayed with in the early
seventies when he was the top techo in LA. --- According to Dean in the early seventies some of
the engineers of The Record Plant (one of LA's top studios ) were sitting on the roof of their
building after a long party. They had taken some speakers from the studio up to the roof for the
party and were discussing the fact that the speakers sounded great on the roof but were pretty
awful downstairs in their control room! Why?? Well, on the roof the speakers were in what is
called an anechoic environment - i.e. no reflections or reverberation! The sound left the
speakers, went past them and didn't come back. Try it yourself - take your favourite speakers
outside and set them up in your backyard or in an open field and have a listen to them.
Suddenly the bottom end will be clean and tight and the top end imaging will be amazing. The
centre will be really tight and defined and you will hear all the mistakes you made in the
recordings. Unfortunately your neighbours won't let you set up a control room in your backyard.
I might mention here that it amazes me how much EQ front of house engineers apply to
speakers when mixing outside concerts where there are no room modes etc. I remember an
engineer being highly offended (and confused da!) when I suggested that JBL must make
terrible speakers when they needed +/- 12db EQ in the open air. (I was the first in Australia to
add 1/3 Octave EQ my studio monitors in 1974 when I got back from LA. Within 6 months I'd
removed them.)
So what is the ideal internal listening environment? Well I reckon if you did a vote you would
find that over 50% would say - In the CAR!!!. I agree, I often check a mix in the car and I know
a lot of other engineers who do the same. So why does the car sound so good? There are a
number of factors and it is these factors that go into making a good listening environment.
The main factors in a car are:

The Shape: There are no parallel walls in a car and what walls there are are thin and
curved.
The Speakers. In a car the speakers are almost always flush mounted. i.e. They a
mounted into a flat surface like below the rear window or in the side door panels. As a
result there are no out of phase signals coming from the rear of the speaker. Also the
rear speakers are mounted in a big cabinet - the boot.
High Frequencies: In the car the windows are the main high frequency reflectors but
they are all at angles (approx. 6 - 12 degrees)and are usually curved as well. The highs
also get diffused evenly throughout the cabin by the dash board. Also the ceiling, sides
and floor are covered in high frequency absorption.
Mid Frequencies: The seats, door panels and passengers are all low mid to high mid
absorbers. Modern cars have deep pile carpet on thick underfelt which also acts as a mid
frequency absorber. Most of the car's acoustic treatment for cutting down engine and
road noise is also on the inside and acts as acoustic treatment for the car stereo.
Low Frequencies: The beautiful thing about cars is the bottom end response. With a
couple of hundred watts a side, a sub-woofer under the seat and the loudness switch on
the bottom end thumps away and sounds great. Actually most of the low end goes
straight through the walls and disappears, consequently it doesn't hum around the
internal body causing phase problems. Any vibration is dampened by the foam lining and
carpet and as far as the low end is concerned the car is equal to open air. Next time you
play a tape/CD in your car get out and listen to what actually leaves the car (most of it!!
especially if the windows are open)
The problem with recording studios is that to keep external sounds out you land up keeping
internal sounds in. People who build studios in the city have to worry about trains underneath,
traffic noise outside, planes flying overhead etc. Obviously the best thing to do is to build it in
the middle of a 50 acre paddock in the country where your only external noise concerns are
birds, wind and rain. Then you can build a simple skin to keep the rain out and allow all the
internal sound to get out so it doesn't muck up the sound within the room.
So how do you create the effect of your car, or the open air, in your studio? - By using
Acoustics!! Treating the walls in your control room and studios so as to control the sound and
thus improve the quality of the sound that you hear and record. I reckon good acoustics can
beat a fancy effect unit any day and they cost about the same.
You can now select any of the topics listed in the adjacent column and progressively gain a good
understanding of the basics of building a quality studio for yourself..
ABSORBERS
By using different construction techniques it is possible to treat the walls so they absorb sound
at various frequencies. The following pages deal with each of the frequency bands-high, mid and
low. By using these construction techniques you can dramatically change the acoustics of a
room. High frequencies are the easiest to absorb and it gets harder as the frequency lowers.
Most home studio enthusiasts only seem to treat the high frequencies in the room yet it is the
mid and low frequencies that cause all the room problems.
If you study the treatment for the low and mid frequencies you will notice that the high
frequencies are not affected by this construction and it is possible to have a room where all
frequencies have a similar reverberation time which is the idea of the whole exercise.
CONSTRUCTION
Isolation
Before you can consider your construction you must consider your isolation requirements. Pages
could be written on this subject but you must consider how much isolation you really want. The
idea of perfect isolation from external noise started in the days when loose miking techniques
were used. One microphone suspended over a string section meant the mic was wound up fully
and was extremely sensitive to ambient noise. Nowadays a mic 6" from a marshal amp is a
totally different story. At Big Toe Studios I often have a window open and the artist will say -"
Hey I can hear the birds, should I close the window?" To which I reply, "No, the only person
who will hear it is some stoned out freak with headphones on who will remark excitedly - wow
man I can hear birds on this track!" But if you have problem neighbours who don't like drums
pounding all day I suggest you apply a certain amount of sound isolation.
The acoustic term here is Transmission Loss. When sound hits a wall there is a certain
proportion of the sound reflected back into the room, some is lost in the absorption of the wall
and the rest travels through the wall and is called the transmission loss.
TRANSMISSION LOSS
The measure of the amount of sound that is transmitted through the wall is called the:
Sound Transmission Class- STC
The transmission loss obviously varies relative to frequency - the STC is a specially weighted
reading across all frequencies and is centred around 500Hz.. Every different wall construction
has a different transmission class.
When sound hits a wall the energy is transferred through the plasterboard to the other side via
the connection to the stud. This problem can be reduced via two ways:
Staggered Studs. Here you use two studs for each side of the wall. The plasterboard on
one side is attached to one stud and the plaster on the other side is attached to the other
stud. The two studs are connected to a common base and top plate.
Flexible Channel. Here a metal channel is attached to the stud and the plasterboard
attached to the metal channel thus reducing the connection to the stud. The channels are
mounted horizontally at 600mm (2 feet) centres. This system is extremely effective -
check out the figures in the STC Chart.
Studs:
Except for staggered stud systems, substituting timber studs for steel studs
generally results in a significant decrease in sound isolation.
Increasing the thickness of steel studs from 0.55BMT to 0.75BMT or 1.15BMT will
decrease sound isolation
Decreasing the stud spacing will decrease the sound isolation.
It is also interesting to note here that the higher the transmission loss the less reflected sound.
In other words in a tent there is a high transmission loss but also a low amount of reflected
sound so a tent makes a good recording room!! So people in the country who can afford a high
transmission loss because there are no close neighbours can allow their sound to get away thus
reducing the amount of treatment required to handle the reflected sound.
The standard gypsum wall in a house has a high transmission coefficient at 100Hz as well as a
high absorption figure because the gypsum panel's resonate frequency is around that figure.
Therefore the reverb in the room is low around 100Hz but higher around 300Hz where the
transmission and absorption are lower. That is why most rooms in a house have a reverb peak
around 300Hz. (You know the one you keep taking out of kick drums and toms.) Check it out on
the reverberation calculator.
Perfect isolation can cost heaps because there is only one thing that will stop sound and that is
MASS. The following solutions apply:
Floating the rooms. A typical construction consists of creating new rooms within your
existing rooms. This means building a floor on top of the existing floor with neoprene
isolation pads and Rockwool on the underside and then building walls and a ceiling using
the new floor as a base. This is the ideal system for total isolation because the new room
is not mechanically connected to the main room but it is also the most expensive system.
The main advantage of floating rooms is the low frequency isolation it gives. If you are
building in a block of flats it would be essential as it is the only real way of achieving total
sound isolation but if you are building a garage studio it really would depend on how
much isolation you require. If you are an acoustic folk band - forget it - if you are a
heavy metal power band it is essential because it is the only way you will stop the bass
and kick drum from annoying your neighbours.
Double Walls. This basically means building two timber/gypsum sheet walls between
each room with Rockwool in the cavity. If you wish to go further you can double the layer
of gypsum and even further by sandwiching a layer of fibre board between the two sheets
of gypsum. (This is extremely effective because sound doesn't like going through
changing medium densities). There are some companies who make sound isolation
gypsum which is thicker and heavier than normal gypsum sheet. In Australia it's called
Soundcheck and is 16mm(5/8") thick. The hollow concrete block -(Besser Block) is an
excellent wall construction as it attenuates sound efficiently and cheaply.
Sealed Environment. There is one important factor that must be understood about
sound isolation. If you build a beautifully sealed wall between two rooms you will get
good isolation BUT if you put one nail hole in the wall you will loose a lot of the isolation!!
Sound is really not the waves we keep on describing but air pressure difference so if you
allow the two air masses to join at any point the pressure gradient will transfer, so make
sure that all walls are sealed tightly. Also make sure all joints around doors, windows
and air-conditioning ducts are sealed.
Double Wall with Floating Floor
In the drawing above I've shown a single gypsum (pale blue) layer but adding to the layer can
dramatically increase the transmission loss. The options are
Adding another layer of gypsum which is glued (not nailed) to the first sheet and should
be a different thickness than the first sheet. i.e. 16mm (5/8") and 12mm (1/2")
sandwiching a layer of fibreboard between the two sheets.
This really works well, a double wall with a triple layer as described above on a floating floor
will create a room that will allow you to set up a band and not hear it outside the room! Just
remember that all the sound is now trapped inside the room and heavy acoustic treatment is
required to control it all.
POWER & LIGHTS
I'll deal with power first because the audio wiring comes after the power installation and
because they both interact we'll sort the power out first. The important thing about power is not
how much is coming in or

do I have enough grunt
but " Is it earthed correctly??"
Balanced and unbalanced electrical circuits

We use balanced and unbalanced leads all the time in the studio but what does it mean? The
following diagram illustrates the difference.
Here we have the two standards. The mic lead and the guitar lead. Balanced and unbalanced.
Three wire system and two wire system. Note that in the mic lead the positive and the negative
don't contact the earth whereas in the guitar lead the negative and the earth are one in the
same thing. The earth - (ground) is exactly that. The green earth wire goes to a copper stake in
the ground so that any short circuit between the positive and earth will send the current to
ground. But because the positive and the negative don't contact the earth it is said to be
floating above ground. The shield acts as a protection from interference by sending any
extraneous electrical interference like hum, to ground. Unfortunately in the unbalanced circuit
negative is ground!
So you would expect that your standard electrical feed from your power supplier would be
balanced. Well unfortunately here in Australia it's not. Sure we get a red positive and a black
negative from the power companies transformer but by Australian regulations the electrician
must link the negative to the earth so we become unbalanced. I understand that is not the
system in the US which is why Marshall amps hum in OZ but don't in the US. I would be
interested in any information I could receive on this matter from anyone from the US.
The system designed to get around this is called the Star Earthing System where you ask your
electrician to earth each power outlet individually like this:
STAR EARTHING OF MAINS POWER

In this setup each power point sees the same ground directly and a unit earthed to outlet 1 and
connected with a patch lead to something earthed to outlet 2 won't see outlet 2 as it's earth
because it has it's own more direct route to ground.
The earth, as stated before, is connected to a copper stake in the ground. It is definitely
advisable to increase this factor by getting your electrician to put two or more stakes in the
ground and connecting them together to increase your ground connection. I've seen systems
where designers have put a whole web of copper stakes under the concrete slab before it is
poured to ensure a good ground connection. In this country where it gets very hot the ground
around the stake can dry out and the connection gets weaker and weaker. It can be solved to a
certain extent by pouring salty water around the stake but two or more stakes is a better
solution.
Lighting: It is advisable to have your light circuit separate from your power circuit. This
decreases the chance of lighting interference in your power circuits. (See lighting further
down this page)
Three Phase Power: Ideally you should have three phase power into your studio. Obviously
the home studio owner won't have it but if you are looking at a professional facility it is a
beneficial addition. The advantage of three phase power is that you can spread your electrical
circuits over the three phases:
Phase 1: Equipment and studio power.

Phase 2: Lighting.
Phase 3: Air-conditioning
Transformer Isolation/Power Conditioning: It is now becoming common to install a power

conditioner in a studio. The advantage here is that you have a transformer between you and the
supplier so that spikes are smoothed and with additional circuitry you can have a voltage
stabiliser that keeps your power voltage stable no matter what the supplier is giving you. You
can also have an added feature that adds battery backup in the case of power failure. This is
great when you have computers as it allows you to save your current work. All these features
are advantageous but can be very expensive! In a three phase setup you can put the
conditioner over your equipment and studio power phase only. One of the common annoying
items is the fridge. Fridges are prone to sticking spikes in the power so watch out for that one.
It gets really tricky setting up your earthing requirements but if you start with your mains
power installed correctly you've got a better chance when it comes to your audio wiring.
LIGHTING
Good lighting is essential in a studio and ideally a separate circuit should be allowed for it.
Downlights over the console and effects area are advisable plus additional downlights for the
client etc. Lighting dimmers can also make for a comfortable environment but be careful here.
You will probably find that the standard light dimmer will cause a buzz interference in your
electrical circuits. I suggest you discuss this with your electrician. There are light dimmers
available (zero crossing) that don't interfere with your electrics but they can be expensive!!
It's not a bad idea to test a few different dimmers before you purchase the full set. In a studio
situation you often require full lighting if the musicians are reading charts through to low mood
lighting when the vocalist is performing a soft ballad. I believe dimmers are the only way to
go.
It's not a bad idea to have control over your studio lights from the control room with a lighting
panel mounted somewhere in your control room. That way you can control your lighting from
one place.
A recent addition to the lighting system is the 12 volt lighting system. This is a good idea in a
studio as these lights are already transformer isolated through the power supply which delivers
the 12 volts.
I really could go over the top here. What I've tried to do in this site is to go through all the fundamentals
so that you can fully understand all the acoustic ramifications and thus be in a position where you can
design your own studio. Layout of rooms etc. is really up to the basic room shapes you are starting with
so instead of me sweating over my computer to give you a whole lot of inappropriate shapes I will give
you the basic styles that have worked for me. If you follow the basic directions of what you need to
acoustically treat each room you can take any shaped room and come up with a plan. I have not included
dimensions as it is not necessary and it would obviously change for each situation. The proportions are
near enough though and the angles are what make it work.
Important thing to remember are:
Stereo room symmetry around your speakers.

Glass windows or doors for communication.

Low-mid frequency absorption from 150 -550Hz.

High frequency absorption.

Absorption across the rear of the control room wall.
Whatever low frequency absorption you can fit in the space.
Things to avoid are:
Having to go through the studio to get to the control room!! (I hate this because you always get
interruptions as people move in and out of the studio)
Creating studios with no visual communication. There is nothing worse than recording someone
you can't see.
Big studios with a small pokey control room and visa versa.
So here are a few ideas that might start you off, use the selector for the different options.
THE BIG FACILITY

THE CORNER CONTROL ROOM
THE GARAGE STUDIO 1
THE GARAGE STUDIO 2

THE CONTROL ROOM
THE BEDROOM STUDIO

LINKS
The following links are to pages I have found surfing and have kept for further reference - I hope
you can also use them.
SAE Institute the largest Institute for Multimedia, Audio Education and
Digital Film Education worldwide
Audio Related Internet, WWW This, I have to say, is the best resource on the web for
the home recording artist. There are pages and pages of
and FTP sites
discussion and fun. I thoroughly recommend it.
Audio Related Internet, WWW a massive directory by Steve Ekbald

and FTP sites
Audio Links from Geoff Martin at the University of Ottawa (using McGill's server)
Malcolm's studio design page one of my favorites! Malcolm has a great site with good
information based on 40 years of experience.
Home Studio Acoustics a good page on home studio acoustics.
Physics of Sound a good rave for all the boffins who can understand it.
Physics of Sound another page with a rave on the physics of sound
Noise-Busters Noise-Busters Huge links page.
Eddy's recording studio RPG Theory page with lots of discussion papers to
download
Tom Hidleys Page This page is a tribute to my hero Tom Hidley who started
the whole studio design trend in the seventies.
Low frequency Room Acoustics a rave on low frequency room acoustics
Mix Magazine Online My favourite magazine
StudioMenu.Com huge page with all listings in the music industry. well
worth a visit.
Studio Guru.Com this is a great pace with a site search to all aspects of the
professional music industry
NAPRS extensive Links Page
Audio & Acoustic Web Sites a comprehensive site by Herb Singleton
Reverb Calculator Page This is where I got the reverb calculator from
Ground Loops This is a great page for those who want to go into the
details of grounding.
Acoustic Calculator This is the mother of all calculators for working out your
room acoustics. great site.
Music Solutions This site has some great pages on hard disk recording
and how to maximise your Win95/98 for recording.
1212 Music This is a huge directory of everything within the music
Industry

SAE Recording Guide

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Index

Drums and Percussion

Vocals and Harmonies

Absorbers - Fader - channel Panel Absorbers

The Banjo and Mandolin

The Violin Family

Lets look at the different positions.

Solo guitar as in folk singer.

Strummed Rhythm Guitar.

There are basically two ways to control low frequencies.

False Wall with Acoustic Hangers

The mass or density of the panel.

A panel absorber is made like this:

Variable Panel Absorber

VARIABLE PANEL ABSORBER

THE HELMHOLTZ RESONATOR

The timber slats can be either finished or

The formula for calculating the helmholtz resonant frequency is:

f = 2160 x sqrt ( r / (( d x D ) + ( r + w )))

f = resonant frequency in Hertz (Hz)

Remember that full absorption is 1 whilst full reflection is 0

Absorption coefficients of common building materials and

Floor materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz

carpet 0.01 0.02 0.06 0.15 0.25 0.45

Concrete (sealed or painted) 0.01 0.01 0.02 0.02 0.02 0.02

Marble or glazed tile 0.01 0.01 0.01 0.01 0.02 0.02

Wood parquet on concrete 0.04 0.04 0.07 0.06 0.06 0.07

Wood flooring on joists 0.15 0.11 0.1 0.07 0.06 0.07

Seating materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz

Benches (wooden, empty) 0.1 0.09 0.08 0.08 0.08 0.08

Benches (cushioned seats and backs,

Benches (cushioned seats and backs,

Benches (cushioned seats and backs,

Seats (fabric-upholsterd, empty) 0.49 0.66 0.8 0.88 0.82 0.7

Seats (fabric-upholsterd, fully

Reflective wall materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz

Brick (natural) 0.03 0.03 0.03 0.04 0.05 0.07

Brick (painted) 0.01 0.01 0.02 0.02 0.02 0.03

Concrete block (coarse) 0.36 0.44 0.31 0.29 0.39 0.25

Concrete block (painted) 0.1 0.05 0.06 0.07 0.09 0.08

Concrete (poured, rough finish,

Glass (small pane) 0.04 0.04 0.03 0.03 0.02 0.02

Plasterboard (12mm (1/2") paneling

Plaster (gypsum or lime, on

Plaster (gypsum or lime, on wood

Plywood (3mm(1/8") paneling over

Plywood (3mm(1/8") paneling over

Plywood (5mm(3/16") paneling over

Plywood (5mm(3/16") panel,

Plywood (6mm(1/4") paneling,

Plywood (19mm(3/4") paneling,

Absorptive wall materials 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz

Drapery (10 oz/yd2, 340 g/m2, flat

Drapery (14 oz/yd2, 476 g/m2, flat

Drapery (18 oz/yd2, 612 g/m2, flat

Drapery (14 oz/yd2, 476 g/m2,

Drapery (18 oz/yd2, 612 g/m2,

Fiberglass board (25mm(1") thick)

Open brick pattern over 75mm(3")

Pageboard over 25mm(1") fiberglass

Pageboard over 50mm(2") fiberglass

Pageboard over 75mm(3") fiberglass

Performated metal (13% open, over

Ceiling material 125 Hz 250 Hz 500 Hz 1 kHz 2 kHz 4 kHz