Module 01 - Analog and Digital Audio PDF

Audio Masterclass Music Production and Sound Engineering Course
Module 01: Analog and Digital Audio
Module 01
Analog and Digital Audio

In this module you will learn about the nature of sound, soundproofing, acoustics and acoustic
treatment, analog audio electronics and digital audio.
Learning outcomes
To understand the way in which sound behaves in air, how sound interacts with hard and soft
materials; flat and irregular surfaces.
To possess the basic background knowledge of recording studio acoustic design.
To understand how sound is handled and transmitted as an electronic signal.
To understand how an analog electronic signal is converted to, handled and stored as a digital
signal, and how it is converted back to analog.
Assessment
Formative assessment is achieved through the short-answer check questions at the end of this
module.
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Module Contents
Learning outcomes 1
Assessment 1
The nature of sound 3
Frequency 5
The decibel 7
The inverse square law 10
Acoustics 11
Standing waves 14
Acoustic treatment 16
Soundproofing 18
Materials 18
The three requirements for good soundproofing 19
Concrete 19
Bricks 20
Plasterboard (drywall) 20
Glass 20
Metal 21
Proprietary flexible soundproofing materials 21
Construction techniques 22
Walls 22
Ceiling 23
Floor 23
Windows 24
Doors 25
Box within a box 25
Ventilation 26
The function of absorption in soundproofing 27
Flanking transmission 27
Cable ducts 28
Audio electronics 29
Passive components 30
Real world components 32
Resistors in series and parallel 33
Digital audio 34
Digital versus analog 34
Analog to digital conversion 36
Problems in digital systems 38
Latency 40
Clocking 40
Check questions 42
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The Nature of Sound

We all know the experience of sound, and we all
learned in school that it is a vibration of air molecules
that stimulates our eardrums. People who work with
sound every day tend not to think about the science
of sound and take it for granted. But unless you have
assimilated a good understanding of the nature of the
medium in which you work, how are you ever going to
make it really work for you?
Sound starts with a vibrating source, commonly the

vocal folds (formerly known as the vocal cords or
sometimes vocal chords), musical instruments and
loudspeaker diaphragms, as far as we are concerned.
Let us think of a loudspeaker diaphragm. It vibrates

forwards and backwards and pushes against air
molecules. On a forward push, it squeezes air molecules Vocal folds - illustration courtesy University
together causing a compression, or region of high of California, Berkley
pressure. On pulling back it separates air molecules
causing a rarefaction, or region of low pressure. The
compressions and rarefactions travel away from the
diaphragm in the form of a wave motion.
Wave motions are all around us, from the water

waves we see in the sea (best viewed from a ship -
the breaking effect near the shore disguises their true
nature), to all forms of electromagnetic radiation such
as x-rays, light, microwaves and radio waves.
The childs toy commonly known as the slinky spring

can display a wave very much like a sound wave. The
slinky is a spring the metal versions work best
of around 15 cm in diameter and perhaps 4 m long
when lightly stretched. If two people pull it out and
one gives a sharp forward and backward impulse, the Slinky spring - photo by Roger McLassus
compression produced will travel to the end of the (GFDL)
spring and if the other person holds his or her end
firmly reflect back.
This demonstrates a longitudinal wave where the

direction of wave motion is in the same direction of
the motion of the actual material (we can call the
motion of the material the particle motion). A sound
wave is a longitudinal wave.
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Contrast this with a water wave where the wave

moves parallel to the surface of the sea, but water
molecules move up and down. This is a transverse
wave. Electromagnetic waves are transverse waves
too.
One feature that the water wave demonstrates

perfectly is that if you look out from the side of a
ship at a piece of flotsam riding the wave, the wave
appears to travel from place to place, carrying energy
as it does so, but the flotsam simply bobs up and
down. Other than wind or tide acting directly on the
flotsam itself, it will bob up and down all day without
going anywhere.
This is true of sound too. A sound wave leaves a

loudspeaker cabinet, but this doesnt mean that air
travels away from the cabinet. The air molecules
simply vibrate forwards and backwards, never going
anywhere. (When air molecules travel from one place
to another that is called, in purely technical terms, a
wind!).
If this were not so then either a vacuum would

develop inside or around the cabinet and there would
be a danger of asphyxiation. Obviously this doesnt
happen. Oddly enough, if you put your hand in front
of a bass loudspeaker you will feel a breeze, if not a
full-on wind, on your hand. This is an illusion since
you feel the air molecules when they press on your
hand, but not when they pull back.
In a transverse wave, such as a water wave,

the direction of particle motion is at right angles
(perpendicular) to the direction of wave motion. In
a longitudinal wave, such as sound, the direction of
particle motion is parallel to the direction of wave
motion.
Although the longitudinal wave in the slinky spring

is similar to a sound wave, it doesnt quite tell the
Transverse wave created on a string. Photo
whole story. The slinky wave is confined within the
courtesy Union College
spring whereas a sound wave spreads out readily.
It is possible to think of each air molecule (actually
oxygen, nitrogen and an increasing amount of carbon
dioxide) that vibrates under the influence of a sound
wave as a sound source in its own right.
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Molecules are of course very small, and it is a feature of

small sound sources or point sources that they emit
sound equally in all directions, or omnidirectionally.
So where light travels over great distances in straight
lines, sound merely has a tendency to follow a straight-
line path, and readily spreads out from that path in an
ever-widening arc, particularly at low frequencies.
[Regarding point sources - it is also worth considering

the example of a small loudspeaker emitting a low
frequency tone. If the speaker is small in comparison
with the wavelength being emitted, then it will have
the characteristics of a point source and will obey the
inverse square law - sound pressure halves for every
doubling of distance from the source.]
Frequency
To compare the range of frequencies in human
experience, a satellite TV signal - for example - has
a frequency of around 10 to 14 GHz. The Olympic
Games have a frequency of 8 nanohertz (they happen
once every four years!).
1 hertz (Hz) means one cycle of vibration per

second
1000 Hz = 1 kHz
1,000,000 Hz = 1 Megahertz (1 MHz)
1,000,000,000 Hz = 1 Gigahertz (1 GHz)
Sound comes in virtually all frequencies but our

hearing system only responds to a narrow range.
The upper limit of young human ears is usually
taken to be 20 kilohertz (kHz) (twenty thousand
vibrations per second). This varies from person to
person, and decreases with age, but as a guideline
its a good compromise. If a sound system can handle
frequencies up to 20 kHz then few people will miss
anything significant.
At the lower end of the range it is difficult to know

where the ear stops working and you start to feel
vibration in your body. In sound engineering however
we put a figure of 20 Hz on the lower end. We can
hear, or feel, frequencies lower than this but they are
generally taken to be unimportant.
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Frequency is related to wavelength by the formula:
velocity = frequency x wavelength
This applies to any wave motion, not just sound. The

velocity, or speed, of sound in air is a little under 340
meters per second (m/s). This varies with temperature,
humidity and altitude, but 340 m/s is a nice round
number and well stick with it. If you work out the
math, this means that a 20 Hz sound wave travelling
in air has a wavelength of 17 metres!
The extreme physical size of low frequency sound

waves leads to tremendous problems in soundproofing
and acoustic treatment. At the other end of the scale,
a 20 kHz sound wave travelling in air has a wavelength
of a mere 17 mm. Curiously, the higher the frequency
the more difficult it is to handle as an electronic,
magnetic or other form of signal, but it is really easy
to control as a real-life sound wave travelling in air.
Low frequencies are easily dealt with electronically,
but are very hard to control acoustically.
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The decibel
The concept of the decibel a convenience that allows
us to compare and quantify levels in the same manner
through different media. In sound terms, decibels can
be used for every medium that can store or transport
sound or a sound signal, for instance...
real sound travelling in air

electric signal
magnetic signal
digital signal
optical signal on a film sound track
mechanical signal on a vinyl record
A change in level of 3 dB means exactly the same thing

in any of these media. Without decibels we would have
to convert from newtons per square metre (sound
pressure), volts, nanowebers per square metre, etc.
Decibels have another advantage for sound. The ear

assesses sound levels logarithmically rather than
linearly. So a change in sound pressure of 100 N/m2 Optical film soundtracks, variable density
(micro-newtons per square meter) would be audibly and variable area. Illustration by Iain F.
different if the starting point were quiet (where it
would be a significant change in level) then if it were
loud (where it would be hardly any change at all). A
change of 3 dB is subjectively the same degree of
change at any level within the ears range.
[Sound pressure is measured in newtons per square

meter. You may think of the newton as a measure of
weight. One newton is about the weight of a small
apple.]
An important point to bear in mind is that the decibel

is a ratio, not a unit. It is always used to compare two
sound levels.
To convert to decibels apply the following formula in

your scientific calculator:
20 x log10 (P1 /P2 )
where P1 and P2 are the two sound pressures you

want to compare. So if one sound is twice the pressure
of another then P1 /P2 = 2. The logarithm of 2 (base
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10) is 0.3, and multiplying this by 20 gives 6 dB.
Actually its 6.02 dB but we dont worry about the odd

0.02.
This is useful because we commonly need to, say,

increase a level by 6 dB, but it doesnt actually tell us
how loud any particular sound is because the decibel
is not a unit. The answer to this is to use a reference
level as a zero point. The level chosen is 20 N/m2
(twenty micro-newtons per square meter), which is,
according to experimental data, the quietest sound
the average person can hear.
We call this level the threshold of hearing and it can

be compared to the rustle of a falling autumn leaf
at ten paces. We quantify this as 0 dB SPL (sound
pressure level) and now any sound can be compared
with this zero level. Loud music comes in at around
100 dB SPL; the ear starts to feel a tickling sensation
at around 120 dB SPL, and hurts when levels approach This fruit weighs approximately 1 newton
130 dB SPL.
If you are not comfortable with math, it is useful to

remember the following, which apply to both sound
pressure and voltage (but decibels work differently
when referring to power):
-80 dB = one ten thousandth

-60 dB = one thousandth
-40 dB = one hundredth
-20 dB = one tenth
-12 dB = one quarter
-6 dB = one half
0 dB = no change
6 dB = twice
12 dB = four times
20 dB = ten times
40 dB = one hundred times
60 dB = one thousand times
80 dB = ten thousand times
Threshold of hearing = 0 dB SPL
Threshold of feeling = 120 dB SPL
Threshold of pain = 130 dB SPL
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Do you need to understand decibels to be a sound

engineer?
The answer is, Yes - to a point. You need to be able

to relate a change in decibels to a fader movement,
and from there to an image in your aural imagination
of what that change should sound like. In addition to
that, youll get producers telling you to raise the level
of the vocal a bit. How many decibels equal a bit?
Only the experience you will gain in the early years of
your career will tell you.
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The inverse square law

There is more to find out about the inverse square
law. Here is an interesting point...
The maximum rate of decay of a sound as you move

away from it is 6 decibels per doubling of distance
(the sound pressure halves). This is simply due to
the spreading-out of sound - the same energy has to
cover an ever greater area.
If the sound is focused in any way, by a large source

or by reflection, then it will fade away at a rate less
than 6 dB per doubling of distance. This fact is of great
importance to PA system designers.
The ultimate focused sound source is the old-fashioned

ships speaking tube. Sound is confined within the
tube and can travel over 100 meters and hardly fade
away at all.
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Acoustics
Its going to be a long time before anyone invents a
way to transfer an electronic or digital signal straight
into the brain, bypassing the ears. Until then, at some
stage sound must always pass through the air, and
this is the most difficult and least understood part of
its journey.
When sound is created, whether it is the human voice,

speaking or singing, a musical instrument or plain old-
fashioned noise, it travels through the air, bounces
from reflecting surfaces, bounces again and mingles
with its own reflection, then enters the microphone.
The same happens at the other end of the chain. Sound

leaves the speakers, and although part of the energy
will be transmitted directly to the listener, much of it
will bounce around the room over a period of anything
from half a second or less in a domestic environment
up to several seconds in a large auditorium.
Compare this with an electrical signal.
Once created, the signal travels in a one-dimensional

medium a cable or circuit track. The signal cant
An audio cable is a one-
escape until it reaches its intended destination, there
dimensional medium
is nothing that it can bounce off (unless the cable is
several kilometers long when it will reflect from the
ends unless measures are taken), and the worst that
can happen is that electrical resistance will lower the
level slightly.
This is a little bit of a simplification, but its fair to

say that everything about the behavior of electrical
signals can be calculated easily.
This is not the case with acoustics. Sound travels in

three dimensions, not one, and will readily reflect
from almost any surface. When the reflections mingle,
constructive and destructive interference effects occur
which differ at every point in the room or auditorium.
The number of reflections is, for all practical purposes,
infinite.
Even with todays sophisticated science and computer

technology, it is not possible to analyze the acoustics of
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a room with complete precision, accounting for every

reflection. It would rarely happen that the electrical
components of a sound system of any kind would be
installed (professionally of course) and then be found
not to work as expected.
It is normal however to complete the acoustic design

of a room or auditorium, and then expect to have to
make adjustments when the building work is complete.
Hopefully these adjustments will not cost more than
the margin of error allowed for in the budget.
Acoustics is a complex science in practice, but in

theory its all very simple. The acoustics of a room
(acousticians use the term room to mean an
enclosed space of any size) are determined by just
three factors: the timing of reflections, the relative
strengths of reflections, and the frequency balance of
reflections.
Look around you at the various surfaces in the room. If

you speak to a colleague, the sound of your voice will
travel directly to his or her ears. It will also bounce off
the nearest surface producing a reflection that arrives
at the ear after a certain number of milliseconds
(sound travels just under 34 cm in a millisecond
one foot per millisecond is often used as a handy rule
of thumb in non-metric countries even though it is a
little bit on the low side). It will bounce off the next
nearest surface with a slightly longer delay, then the
next. Then reflections of reflections will start to arrive.
At first they will be spaced apart in time but soon
there will be so many reflections that they turn into a
general mush of reverberation.
Some surfaces will be more absorbent, so reflections

are lower in level. Some surfaces will favour certain
ranges of frequencies. These three factors almost
completely determine the acoustics of a room.
There is a fourth factor that is worth mentioning

movement. If anything moves in the room source,
listener or any reflecting surface then the Doppler
effect comes into play.
The Doppler effect is best demonstrated by the siren

of a passing police car, which appears to drop in pitch
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as it goes past. Sound cant travel faster than its

natural velocity in any given medium, so if the sound
source moves, then velocity of the source converts to
a rising in pitch for an approaching source, a lowering
of pitch (acoustic red shift if you like) for a source that
is moving away.
In most contexts where acoustics are important, neither

the source nor listener will be moving significantly,
nor will the reflecting surfaces.
What will be moving however is the air in the room

due to convection effects and ventilation. You can see
this quite clearly if you anchor a helium balloon so
that it can float midway between floor and ceiling.
Even in a living room it will move around more than
you would expert.
This effect is often modelled in digital reverberation

units where it adds useful thickening to the sound, or
chorusing as some sound engineers might say.
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Standing waves
Although acoustics is a science, the ultimate arbiter of
good acoustics is human judgment. There are certain
basics that must be adhered to, derived from common
knowledge and experience, and also statistical tests
using human subjects.
Firstly, a room that is designed for speech must

maintain good intelligibility. Too much reverberation
obscures the words, as do reflections that are heard
by the listener more than 40 milliseconds or so after
the direct sound.
Late reflections cause phonemes (the sounds that

comprise speech) to overlap. Short reflections actually
aid intelligibility by making unamplified speech
louder.
For both speech and music there is the requirement

that the reverberation time (normally defined as the
time it takes for the reverberation to decrease in
level by 60 dB the RT60 ) is in accordance with that
commonly found in rooms of a similar size.
A small room with a long reverberation time sounds

odd, as does a big room with a short reverberation time.
We can thank the British Broadcasting Corporation
(BBC), who probably own and operate more purpose
designed acoustic spaces than any other organization
in the world, for codifying this knowledge.
Standing wave demonstration using a string
One of the most common problems in acoustics, that
particularly affects room-sized rooms, rather than
concert halls and auditoria, is standing waves. The
wavelength of audible sound ranges from around 17
mm to 17 m. Suppose that the distance between two
parallel reflecting surfaces is 4 m. Half a wavelength
of a note of 42.5 Hz (coincidentally around the pitch
of the lowest note of a standard bass guitar) will fit
exactly between these surfaces. As it reflects back and
forth, the pattern of high and low pressure between
the surfaces will stay static high pressure near the
surfaces, low pressure halfway between. The room will
therefore resonate at this frequency and any note of
this frequency will be emphasized. The reverberation
time at this frequency will also be extended.
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This will also happen at integral multiples of the standing

wave frequency. Smaller rooms sound worse because
the frequencies where standing waves are strong are
well into the sensitive range of our hearing.
Standing waves dont just happen between pairs of

parallel surfaces. If you imagine a ball bouncing off all
four sides of a pool table and coming back to where it
started; a standing wave can easily follow this pattern
in a room, or even bounce off all four walls, ceiling
and floor too.
Wherever there is a standing wave, there might

also be a flutter echo. Next time you find yourself
standing between two hard parallel surfaces, clap your
hands and listen to the amazing flutter echo where all
frequencies bounce repeatedly back and forth. Its not
helpful either for speech or music.
[At higher harmonics than the fundamental frequency,

the pattern of high and low pressure can be such
that there is high pressure in the centre between the
boundaries and low pressure elsewhere. The pressure
is always high at the boundaries.]
The solution to standing waves is firstly to choose the

proportions of the room so that the standing wave
frequencies are spread out as much as possible.
Square rooms concentrate standing waves into a
smaller number of frequencies. A cube shaped room
would be the worst. Non-parallel walls are good, but
these damned clever standing waves will still find a
way. We need...
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Acoustic Treatment
The function of acoustic treatment is to control
reverberation time and to reduce the levels of
standing waves. Well come back to standing waves
in a moment.
If surfaces can be made more absorbent then

obviously reflections will be reduced in strength,
hence reverberation time will be less.
Soft materials such as carpet, drapes and especially

mineral wool all find applications as porous absorbers.
Porous absorbers however only work well when they
are at least a quarter of a wavelength thick.
This means that they are only really practical for

high and high mid frequencies. If the only acoustic
treatment used in a room is porous absorption, then
the room will sound incredibly dull and lifeless.
Another type of absorber is the panel or membrane

absorber. A flexible wood panel (around 4 mm to 18
mm thick) mounted over a sealed air space (around
100 mm to 300 mm in depth) will resonate at low
frequencies, and as it flexes will absorb energy.
If damping material (typically mineral wool) is added

inside, or a flexible membrane is used, then this
type of absorber can be effective over a range of low
frequencies. Drill some holes in the panel and the
absorption becomes wide band.
Ideal! Panel absorbers with little damping can be

tuned to the frequencies of standing waves and control Panel absorber
them very effectively. The other way of dealing with
standing waves, and at the same time waving a magic
wand and making the room sound really great, is to
use diffusion. Irregular surfaces break up reflections
creating a denser pattern of low level reflections
than would occur with mirror-like flat surfaces. The
irregularities however have to be comparable in size
to the wavelengths you want to diffuse. Sound is
always difficult to control.
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Soundproofing
There has always been a lot of confusion between the
role of materials that reflect sound, and materials that
absorb sound. Sound-absorbing materials are NOT
good at blocking sound transmission.
This is not to say that they have no function in

soundproofing. Just that the general public consensus
is that to provide soundproofing, all you need is lots
of absorbent material. This is 100% absolutely not so.
Heres an example...
Suppose a partition is created from a very thick

layer of mineral wool (the most cost-effective sound
absorber there is). Suppose it is so thick that it absorbs
75% of the sound pressure that falls upon it, leaving
only 25% to transmit through to the other side. This
seems good, since the sound pressure has dropped to
a quarter.
However, when you consider this in decibel terms,

reducing sound pressure to a quarter is a change of
minus 12 dB. So if the sound pressure on the side
where the sound originates is 100 dB SPL, the sound
pressure on the other side of the partition is still a
very significant 88 dB SPL.
This is a noticeable difference, but its hardly

soundproofing. For really effective soundproofing we
need a drop of at least 45 dB, and preferably more.
Even then, the sound will very likely be audible on the
other side of the partition.
Materials
Effective soundproofing can only be provided by
materials which reflect sound energy. Such materials
would be massive and non-porous, such as concrete,
or a well-made brick or blockwork wall. Here is a list
of suitable materials:
Concrete
Bricks or non-porous blocks
Plasterboard (also known as drywall, sheetrock,
wallboard, & gypsum board)
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Plywood and dense particle board

Glass
Metal
Proprietary flexible soundproofing materials
The two characteristics that all of these have in common

is mass and non-porosity. The last item, proprietary
flexible soundproofing materials covers an immense
range of potential solutions, some of which - when
you look at their advertising material - seem to work
by magic rather than physics. They will only work if
they are massive and non-porous - simple as that.
The three requirements for good

soundproofing
Having looking briefly at the materials, we can
now consider the three requirements for good
soundproofing:
Mass
Continuity of structure
No defects
Mass means what it says. Double the mass of a

partition and you get 6 dB more insulation.
Continuity of structure not only means non-porosity,

it means that the soundproofing should enclose the
room in question absolutely 100%. If there is any
place where sound can get through, it will. You could
spend a lot of money and see it wasted because of
acoustic holes in the structure.
No defects really means the same as continuity

of structure. Except that it is one thing to design a
room with no acoustic holes, quite another thing to
see it through to completion successfully. Builders
do not always comply with architects plans 100%,
and the short-cuts they take could well compromise
soundproofing considerably.
Lets go back and look at the materials once again...
Concrete
Concrete is a wonderful building material. The only
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consideration is that it should be vibrated effectively

to make sure there are no air pockets.
Bricks
A house brick often has a hollow in one surface,
known as the frog. BBC practice is to lay bricks with
their frogs uppermost (which is not always the case
in conventional building practice) because then they
have to be filled completely with cement. This makes
the wall heavier than it may otherwise have been, and
therefore a little bit better at soundproofing.
Plasterboard (drywall)
Plasterboard consists of a layer of gypsum plaster
around 12 mm thick sandwiched between two sheets
of thick paper. With it you can make a dry partition.
A wooden framework is constructed and layers of
plasterboard nailed on.
The BBCs double Camden partition consists of two

such frameworks, onto which are nailed a total of
eight layers of plasterboard.
The advantage of dry partitions is that they can be

constructed while the rest of the studio complex is
still operational. Concrete and bricks are very much
more messy, making it more likely that operation
will have to be closed down totally. Dry partitions
are sometimes called lightweight partitions. This is
because you can divide a room into two using just two
sheets of plasterboard on the wooden framework. But
by the time you have added enough extra layers for
good soundproofing, it is no lighter than a brick wall
providing the same degree of insulation.
Plywood and particle board such as

chipboard and MDF
These are all good materials - obviously the denser,
and therefore heavier, the better. They are more
expensive than plasterboard however, so they are
only used where they are needed.
Glass
Glass is a very good material for soundproofing, but it
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is expensive. Therefore it is only used when you need

to see through the soundproofing.
Metal
Once again, this is a very good material for
soundproofing, but it is expensive in comparison to the
alternatives. It is only used where its high density is
important in achieving a relatively thin soundproofed
partition. It is most commonly found in soundproofed
doors, which may have a lead lining.
Proprietary flexible soundproofing

materials
With regard to the comments made above, these are
generally expensive in comparison with their acoustic
worth. They should only be used where flexibility or
ease of installation is important. They may also be
used as damping material.
For instance a metal panel in a car may vibrate and

transmit energy to the passenger compartment. If it
were damped, then not only would the vibration would
be reduced, but significant energy would be taken out
of the sound wave.
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Construction techniques
A room is made up from a variety of surfaces and
components, all requiring their own construction
techniques:
Walls
Ceiling
Floor
Windows
Doors
Cable ducts
Ventilation ducts
Walls
Whatever material the wall is made out of, it is better
to use two thin walls spaced apart rather than one
thick one of equivalent mass.
Remembering that soundproofing is best achieved by

reflection, and that reflection occurs at the boundary
between one material and another, it makes sense to
provide four boundaries rather than two.
At first thought, it may seem that if a partition has a

sound transmission class (STC) of say 35 dB, and two
such partitions are provided, then the overall STC will
be 70 dB. This is not the case.
By doubling the mass you get an extra 6 dB, and by

spacing apart the two leaves of the partition you might
gain another 3 dB.
This may not sound like much, but it costs hardly

anything so it is worth having.
The reason why you dont get twice as many decibels of

sound reduction is that the two leaves remain coupled
together. In fact the more closely they are coupled,
the more the object of the exercise is defeated.
Double-leaf brick walls (cavity walls) are often

constructed using wire or plastic ties which couple the
leaves together for mechanical strength. For a wall
that is designed for good soundproofing, the use of
such ties should be minimized. Care should be taken
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not to allow cement to fall onto the ties. In normal

building, this would not matter.
Also, builders are known to have a habit of depositing

rubbish between the leaves of a cavity wall. This of
course must not be allowed to happen as it couples
the leaves of the partition.
The space between the partitions should be filled

with absorbent material such as mineral wool. This
is where absorbent material does have a place in
soundproofing. If the cavity is left empty, sound will
bounce back and forth between the leaves and some
of the reflected energy will end up being transmitted.
If this can be absorbed significantly, then the insulation

will be better.
Ceiling
The difficulty in building a sound proofed ceiling is
mounting sufficient mass horizontally. The brute force
solution is to lay concrete on top of metal shuttering,
preferably as a double-leaf construction. The concrete
could be up to 175 mm thick in total. As always, mass
wins.
For a less heavily engineered solution, the BBC

recommend woodwool slabs. Woodwool is a sheet
or board made from a mixture of thin strips of wood
and cement, which are bound together through
compression within a mould. Layers of plasterboard
can be used too, providing they are adequately
supported.
Acoustic tiles are virtually useless as sound

insulation, although they do find application in acoustic
treatment.
Floor
Once again, mass rules. But also there is a technique
known as the floating floor, which is widely used in
studio construction.
The fully engineered floating floor would consist of a

concrete slab formed on metal shuttering, supported
on rubber pads or even heavy-duty springs. The mass
of the slab is important as the mass-spring system
Page 22
will have a resonant frequency, at which the sound

insulation properties will be worse than if the floor
were not floating! A massive slab can push this
resonant frequency below the audio band.
Where restriction on cost or loading prevents a

heavy floating floor, a lighter-weight version can be
constructed to BBC recommendations by putting
down a resilient layer of high-density mineral wool
(approximately 30 mm) covered with a plastic sheet,
and then laying a 70 mm reinforced concrete slab on
top of that, with a further 40 mm concrete screed on
top.
A wooden domestic floor may be improved by adding

two layers of 18 mm particle board on top, with the
joints staggered to avoid gaps. The gaps around
the edges can be filled with a mastic material, or
by compressing mineral wool tightly into the gaps.
There would be no harm in floating this on top of an
old carpet, but the additional benefit, other than for
impact noise, would be slight.
Windows
Glass is a good soundproofing material if it is thick
enough.
There is one type of window that is absolutely useless

for sound insulation however, and that is an open
window!
If a window has to be opened to provide ventilation

then all of the rest of the soundproofing in the studio
is rendered worthless. Some windows have to be
opened only for cleaning, in which case the opening
section should be surrounded by a compression seal,
not a brush seal (or no seal at all).
A properly constructed window would have two panes

of differing thicknesses (to avoid resonance effects
allowing the same band of frequencies through each
pane), set in mastic in a well constructed frame.
The flexible mastic decouples the panes from the

frame and from each other, reducing the opportunities
for sound transmission. Since this is in effect a double
leaf partition, it would be sonically advantageous to
Page 23
fill the gap between the panes with mineral wool.

Unfortunately, the window would now no longer
function as intended.
The best compromise is to line the reveals (the edges

between the panes) of the window with absorbent
material to soak up the energy that would otherwise
bounce around inside until it found an outlet.
Often, windows are constructed so that the panes

are angled to each other. This has some value in
preventing standing waves being set up between the
otherwise parallel panes, but its greater value is in
cutting down on the visual reflections there would be
otherwise.
Doors
The best solution to access is simply to buy a sound
proofed door. This will be expensive, but it will be worth
it. It will probably have magnetic seals around the top
and sides, and a compression seal at the bottom. It
will also be very heavy, meaning that the wall it fits
into will have to be strong enough to support it.
A reasonable alternative is a heavy fire door, with

the jamb fitted with rubber compression seals and
extended all the way around the door, including the
bottom.
To gain better insulation than a single door can

achieve, a sound lobby is sometimes constructed so
there are two doors between one side of the wall and
the other.
Box within a box

The ultimate in studio building is the so-called box
within a box structure. Here, the external building
provides shelter from the elements, and office
facilities, but within it is a completely enclosed and
self-supporting structure standing on rubber pads or
springs. Naturally, this is expensive.
Ventilation
Ventilation and air conditioning, sometimes known as
HVAC (the H stands for heating) is a vitally important
Page 24
topic to study in conjunction with soundproofing.

When a studio is soundproof, it is also air proof, unless
steps are taken.
Ventilation and air conditioning are not synonymous.

Ventilation means access to fresh air from outside
the building, air conditioning means cooling and
maintaining the humidity of the air that is already
inside. An air conditioning system may provide
ventilation, but many do not at all.
There are a number of problems caused by such

systems:
Noise caused by air turbulence within the ducts

Fan noise transmitted through the ducts
Noise in the structure of the building
transferring to the ducts and being transmitted
through them
Fan noise transmitted through the metal of the
ducts
The ducts create transmission paths through
the building
These are the solutions...
Turbulence is reduced by having ducts with

a large cross-sectional area. This allows the
air velocity to be lower and any remaining
turbulence will be lower in frequency.
Any airborne noise can be reduced by the
incorporation of plenum chambers. A plenum
is a large space through which the air must
travel, lined with absorbent material. The air
temporarily slows down and allows time for any
sound it carries to be absorbed.
The ducts are also lined, bearing in mind
that the absorbent material must not give off
particles (like mineral wool does), unless the
air is being extracted. Baffles are generally not
used as they increase turbulence.
Noise that would otherwise travel through the
metal of the duct is reduced by suspending the
ducts flexibly, and by having flexible connector
sections every so often to absorb vibration.
Page 25
Noise from the fan that would otherwise enter

the structure of the building can be reduced
by mounting the fan on a heavy plinth, itself
resting on resilient pads. Obviously, a fan that
is intrinsically quiet should be used.
Studio ventilation and air conditioning systems should

be installed by contractors who have experience in
doing this in a studio environment. Otherwise it is
likely that the result will not be satisfactory.
The function of absorption in

soundproofing
If soundproofing were carried out using only
reflective materials, then the sound energy would
continue to reflect back and forth, each time offering
another opportunity for some of the energy to be
transmitted.
If there is sound absorbing material within the room

or cavity, this energy will sooner or later be absorbed
and converted to heat.
Using absorption in this way is useful when the level

of the sound source is fixed - a noisy fan could be
enclosed and the enclosure filled with absorbent
material, for example. It is also useful in cavities.
In a recording studio control room however, adding

more absorbent material for this purpose is not useful
since it will lower the sound level in the room and the
engineer will simply turn up the level to compensate.
Flanking transmission
Flanking transmission occurs where are partition is
built up to the height of a suspended ceiling, or down
to the level of a raised floor, but not all the way to the
solid structure of the building.
No matter how good the soundproofing qualities of

the partition, sound will take the flanking path over or
under the obstacle.
Cable ducts
Where a cable duct passes through a partition there
Page 26
will be the opportunity for sound to leak through the

duct.
To prevent this, the space not occupied by cables must

be filled with pugging. This can take the form of sand
in bags, or tightly compressed mineral wool.
Note that there is no l in pugging.
Page 27
Audio Electronics
Sound engineers commonly feel quite close to the
electricity and circuitry that form and guide their
signals. What goes on inside the equipment is, to
many, as interesting as the sound that comes out
of it and is often the subject of heated debate. Next
time you bump into a sound engineer, whisper, Tubes
(valves) or transistors? in his or her ear.
An understanding of the nature of electricity is

important for a sound engineer to be able to do their
work properly. As a starting point, lets take the
concept of voltage, or electrical pressure.
Its useful to think of voltage, measured in volts (V),

as the motive force behind all electrical interactions.
In fact we sometimes refer to voltage as EMF, meaning
electromotive force, although the full term isnt in
common use.
An audio signal is always an alternating voltage (where

the voltage swings continually between positive and
negative) if there is a DC component then it shouldnt
be there and the frequency of the voltage is exactly
the same as the frequency of the original sound.
Since the frequency range of human hearing is taken

to be 20 Hz to 20 kHz, then these are the frequency
bounds of an audio signal. The level of the signal may
vary from microvolts (millionths of a volt), produced
by a microphone in quiet conditions, through a volt
or so in a mixing console, up to 100 V or more at the
outputs of a power amplifier (keep your fingers well
away!).
Electricity flows from positive to negative voltage.

At least that is what the early pioneers of electricity
thought. It actually flows from negative to positive,
transported by electrons, but we still generally think
in terms of conventional current. Yes it is confusing.
Nearly always we use a zero voltage reference, which

in mains powered equipment is connected to earth,
ultimately through a copper spike, or equivalent, sunk
deep into the ground.
Page 28
When a voltage is applied to a circuit component, or

to anything for that matter, then a certain current will
flow, measured in amperes (A). Its good to think of
current as the rate of flow of electrons the more
electrons that are moving, the greater the current.
The magnitude of the current fairly obviously depends
on the magnitude of the voltage. It also depends on
the resistance of the object or component that is
subjected to the voltage.
Some materials have a high electrical resistance,

measured in ohms (), and are therefore good
insulators. The lower the resistance, the greater the
current for a given voltage:
I = V/R, where I is current, V is voltage and R is

resistance
When the voltage is alternating we also commonly

think about impedance. Impedance (Z) is analogous
to resistance but accounts for situations where the
current and voltage are not alternating in step, or are
not in phase, as we would put it. Impedance is also
measured in ohms.
There is also reactance (X), as found in a capacitor or

inductor, which restricts the flow of current but there
is no, or negligible, resistive component. Impedance
is the combination of resistance and reactance.
Electrical power is another important concept. Power

in general terms is the rate of flow or conversion of
energy. Energy, as contained in a battery for example,
is measured in joules (J).
A battery contains a certain quantity of energy and

when that is used up the battery is dead. You can use
it at a faster or slower rate. Connecting the battery
to a low resistance circuit will result in a high rate of
energy release, and a high power will be developed.
Power (P), measured in watts (W), can be calculated
in two ways:
P = V2/R
P=VxI
Page 29
Passive components
The three principal passive components are the resistor,
the capacitor and the inductor. They are passive in the
sense that they do not add to the signal in any way.
They can reduce it in various ways, but they cant
increase the power of the signal and generally do not
change the shape of the waveform (distortion).
The resistor is commonly used to reduce the level of a

signal, or to develop a voltage from a current flowing
in a conductor. It can also be used to reduce current
flow. Fig. 1 shows a potential divider. The voltage
applied to the input creates a current...
I = Vin/(R1 + R2 )
The voltage at the output is therefore...
Vout = I x R2
If the resistors are of equal value then this arrangement

halves the voltage.
Potential divider
Fig. 2a shows a wire of low resistance. Apply a voltage
and current will flow so easily that the difference in
voltage at the two ends will become instantly zero,
or at least very close to zero. To take advantage of
this current to create a voltage, insert a resistor as in
Fig. 2b. There will now be a voltage between the two
terminals of the resistor...
V=IxR
Of course the current will now be less, simply because

there is a resistance present in the circuit.
The capacitor, formerly known as the condenser,

is used to control the high frequency content of a
signal.
Capacitors allow high frequencies to flow readily but

have a higher reactance to low frequencies. (Note the
use of the word reactance since voltage and current
are not in phase and there is no resistive component).
The relevant equation is: capacitative reactance
X=1/2fC, where f is frequency and C is capacitance
measured in farads (F). Measuring voltage
Page 30
Fig. 3 shows a circuit in which a capacitor is used

to control the level of high frequency- signal
components. You will notice that it is similar to the
resistive potential divider. If the frequency is very low
then the capacitor will appear as a high impedance.
If you plug an arbitrarily high number of ohms into
the potential divider equations, then you will find that
Vout is almost equal to Vin. (It may seem odd that Vin
is not reduced by passing through a resistor. This is in
fact the case if no current is drawn by the measuring
instrument.
You can imagine the measuring instrument to be

another R2 in another potential divider. If its input
impedance, as we would call it, is high, then the
voltage will not be reduced).
However if the frequency is high, then the impedance

of the capacitor will be low and it will allow current
to flow readily to earth where it is lost. Vout will
therefore be low.
An inductor (formerly known as a choke) is very much

like a capacitor, except that it will pass low frequencies
and restrict high frequencies.
Where two or more coils of wire are wound around a

metal former, the result is a transformer. A transformer
can act on an electrical signal to exchange more
voltage for less current, more current for less voltage.
The transformer, as a passive component, cannot
increase the power of a signal.
Page 31
Real world components

Resistors, capacitors and inductors are idealized
concepts that in the real world have to be manufactured
using available materials and technologies.
Resistors are easily manufactured in any value from

a fraction of an ohm up to 1 Mohm (1 million ohms)
or more from a variety of resistive materials. Small
resistors are used in low power circuits. Enormous
wire-wound resistors are available for high power use.
The desired accuracy and stability of the value can be
specified, commonly to 1% but better is available.
Capacitors are more of a problem. Capacitors consist

of two metal plates (sometimes coiled) separated by a
dielectric. Small value capacitors (1 pF would be about
the smallest useful value) can readily be made from
ceramic materials, mica etc. Large value capacitors
are more difficult.
So-called electrolytic capacitors may have high values

up to 100,000 F but they have to be polarized so that
one terminal experiences a positive DC voltage with
respect to the other. Electrolytic capacitors are neither
accurate nor stable in their values. They are also
bulky. Tantalum electrolytic capacitors are smaller,
but cannot withstand high voltages.
An inductor is simply a coil of wire, sometimes wrapped

round a former of soft (i.e. low coercivity) magnetic
material. Inductors are more costly to manufacture
than resistors and capacitors and their use is avoided
whenever possible. A capacitor, for instance, can
control high frequencies when placed in parallel (Fig.
4a) with the signal. It controls low frequencies when
place in series (Fig. 4b). At high power levels, such as
in loudspeaker crossovers, inductors have to be used.
One problem is that the metal core of an inductor
can become magnetically saturated i.e. it cannot
be magnetized to any greater degree. This causes
clipping and therefore distortion of the waveform.
Passive components are useful in a variety of ways,

but their limitation of not being able to increase the
power of a signal means they can only do so much.
Page 32
Resistors in series and parallel

Fig. 5a shows three resistors in series. Each resistor
gets a chance to block the current, therefore their
values add to give the total resistance: Rtotal = R1 +
R2 + R3 . When resistors are placed in parallel, as in
Fig. 5b, each resistor presents another opportunity for
current to flow, therefore the total resistance is lower
than the value of any of the individual components:
1/Rtotal = 1/R1 + 1/R2 + 1/R3 . This is important
in sound engineering when connecting a number of
loudspeakers to the same power amplifier.
For instance, four identical loudspeakers can be

connected as two pairs in series, and the pairs
connected in parallel. The resulting impedance will be Resistors in series and parallel
exactly the same as one loudspeaker by itself.
Page 33
Digital Audio
Digital versus analog
In analog audio, the varying pressure of a sound wave
travelling in air is represented by a varying electrical
voltage. When the sound pressure goes higher, the
voltage goes higher. When the sound pressure goes
lower, the voltage goes lower. Sound can also be
represented by other kinds of analog signals, such as
a magnetic signal, mechanical signal on a vinyl record,
optical signal on a film sound track. In all cases, some
property of the medium varies in correspondence with
the original sound pressure.
The problem with analog audio is in the detail. Where

there are very fine variations in air pressure, then
there must be very fine variations in the storage or
transmission medium. Although electrical signals can
represent sound very accurately, there is a problem
with magnetic storage, as in the old analog tape
recorders. That problem is noise.
The medium of magnetic tape is inherently grainy, Studer A80 analog tape recorder
for want of a better word. This granularity causes
variations in the magnetic signal that are confused
with the actual signal being recorded and stored. On
playback, the granularity of the medium manifests
itself as noise. And quite a lot of noise too.
So to combat noise on a magnetic tape, the signal

being recorded is made higher in level, causing
stronger magnetism on the tape. This is so that the
variations in magnetism caused by the signal are
very much greater than those caused by the noise.
The problem with this is that we run into magnetic
saturation. There comes a point where the tape is
reluctant to become any more strongly magnetized.
This causes distortion. So analog tape recording is
always a compromise between noise and distortion.
In audio, this was the primary reason why we moved

from analog to digital. It is possible to make a digital
recording system with very low noise; it is impractical
to do that with analog recording technology.
Page 34
Digital audio has another advantage it can be

copied without loss of quality. When making an analog
recording, there is always a certain loss of quality, even
in the original recording. When that recording is copied
there is a further loss which is significant and audible.
In the normal course of production, a recording may
be copied several times before it reaches the listener,
each generation of copying increasing the noise and
distortion.
However, when a digital recording is copied, it is a

simple matter of reading the numbers, which are just
ones and zeros, from one piece of media and writing
them onto another. As long as the numbers are copied
correctly, the copy will be perfect, and is often referred
to as a clone.
To put this another way, if some defect in the analog

copying process causes the voltage to wiggle, then
this wiggle will cause an audible distortion in the copy.
If some defect in the digital copying process causes a
zero to be a little but squashed in the middle, well it
is still a zero and has exactly the same meaning as it
did before.
So there are our two primary reasons why we

transferred from analog audio to digital noise and
the ability to make perfect copies.
Another very good reason developed over a period of

time. Digital processing is very much cheaper than
analog processing. So a digital mixing console that
costs a couple of thousand dollars has the same mixing
and processing capability as an analog console that
costs tens of thousands of dollars, and maybe more.
Although it is not specifically an advantage of digital
audio itself, it is also an operational convenience that
the entire settings of a digital mixing console can
be stored at the push of a button. There were some
analog consoles that could do this, but they were
fiendishly expensive, and resetting the controls was a
lengthy manual process.
Page 35
Analog to digital conversion

An analog signal, represented as a varying voltage,
must be converted to numbers. This is done by
sampling the signal so many thousands of times per
second.
There is a simple theory Nyquists theory that

states that to successfully digitize a certain frequency,
you have to sample it at at least twice that frequency.
So since the conventionally accepted limit of human
hearing is 20,000 Hz, or 20 kHz, then an analog to
digital convertor must sample at at least 40 kHz. In
practice there needs to be a safety margin above this
so we sample at 44.1 kHz or 48 kHz.
Why 44.1 kHz? Its for historical reasons. Digital audio

was originally recorded onto modified video recorders.
In video terms, 44.1 kHz is a nice round number that
happens to work.
Why 48 kHz? This is preferred in video and Analog to digital convertor

broadcasting and also stems from historical reasons.
When audio signals were first transmitted by satellite,
the sampling rate was 32 kHz. This is too low for
really good quality audio, but 48 kHz has a simple
mathematical relationship with 32 kHz so it was easy
to convert between the two. Its easy to convert from
anything to anything these days, but 48 kHz remains
as the alternative standard.
It is also common to sample at 96 kHz, and feasible to

sample at 192 kHz. These higher rates allow a better
high frequency response. This may not be audible but
the argument that 130 kilometers per hour in a car
capable of 200 kph is a smoother experience than in
a car that can only do 131.
So to summarize, the voltage of the analog signal is

sampled forty-odd thousand times per second.
What we meaning by sampling is that at any one

sampling period, the voltage is grabbed and stored,
just for a tiny fraction of a second. It is stored to allow
time to measure it. Measurement is not instantaneous
in practical convertors. We talk of the sample and hold
circuit doing this job while the voltage is measured.
Page 36
At this point the signal is still an analog voltage. The

next step is to convert it into a number.
This is straightforward the voltage is divided into

65,536 different allowable values. At any instant,
the allowable value that is closest to the actual value
is selected. We now have a number. This number
is expressed in binary form as a sequence of zeros
and ones, all the way from 0000000000000000 to
1111111111111111.
Two questions...
Firstly, where does this magic number of 65,536 come

from? The answer is that if we use a sequence of
sixteen binary digits for encoding a 16-bit number
then there are 65,536 different possible combinations.
Clearly if we encode to 20-bit or 24-bit resolution then
there will be more.
Secondly, what happens when the actual signal falls

between two allowable levels? Wont there be an
error here? The answer is yes. We call the process of
selection of the nearest allowable level quantization.
Where the actual signal level is different, which it will
be most of the time, this is the quantization error.
Quantization error leads to distortion and noise. But

the more bits you have in the system, the smaller this
error is. As a rough guide, each bit is worth around
6 decibels of signal to noise ratio. So in a 16-bit
system, the theoretical signal-to-noise ratio is 96 dB.
In practice this will never be completely attained. In
a 24-bit system the theoretical signal-to-noise ratio is
144 dB, but achieving better than 115 dB or so is still
proving difficult in practice.
To round off this section, a little bit of common

parlance. We would describe the compact disc system
as being 16-bit/44.1 kHz because it has sixteen bits
of resolution and is sampled 44,100 times per second.
Modern recording systems can be 24/96 an even
shorter but common way of saying it which means
of course twenty-four bits, sampled 96,000 times per
second.
Page 37
Problems in digital systems

We have already discussed frequency response and
signal-to-noise ratio. Distortion in digital systems is
very closely linked to noise. Although the distortion
is at a very low level in practical systems, digital
distortion is highly offensive to the ear.
The cure for digital distortion is dither. Dither is a

very low level noise signal that is added to the analog
signal being digitized. This might seem like a crazy
idea, to add noise to the signal. But the problem is that
digital distortion is correlated to the signal, and the ear
easily picks up on that. Adding noise randomizes the
distortion so that the noise is smooth and continues.
The ear is happy to ignore that. And in a 16-bit system
the noise is at a very low level anyway.
A relatively jitter-free digital signal shows
Another problem in digital systems is jitter. Jitter this distinctive eye pattern. When there is
is caused when a digital signal is transmitted at an jitter, the eye closes.
uneven rate. This can occur because of faulty, poorly-
designed or cheap components. Or by long cable
paths and physical problems in the system. Jitter
translates as noise in the output. Sometimes a lot of
noise. Fortunately, jitter can be completely cured by
re-clocking the signal so that it is even and regular.
For practical purposes, it should be said that you
will rarely come across significant problems caused
by jitter using properly designed equipment. It is
something that is worth knowing about however.
One of the features of digital audio is that it uses

zeros and ones to transmit and store the signal. So
for instance a zero could be represented by a low
voltage, a one by a high voltage. It would be very
easy to distinguish the difference between a zero and
a one. But sometimes there can be excessive noise
or interference in the transmission or storage system
and zeros and ones can be confused, causing errors.
Digital systems therefore are able to detect errors and

compensate for them.
The first thing to do is to detect whether an error has

occurred. The digital code used to store a signal has
special data added to make this possible. For example
if we devised a numerical code to represent letters of
Page 38
the alphabet, and we specified that only even numbers

would be used, if you received a message from us and
some of the numbers were odd, you would know that
there had been an error.
When a digital system detects an error, it will attempt

to correct it. Error correction means that the data is
as good as it was before and the defect is completely
inaudible. This is done by storing extra redundant
data that can be used if some of the signal data is
damaged. So if there is a scratch on a compact disc,
the error handling system will recognize that the data
is damaged and will look for the redundant data to
replace it.
Sometimes however the extent of the damage is too

great to do this, or the redundant data is damaged
as well. In this case error concealment will take
place. The system will make an intelligent guess as
to what the data would have been. Audio signals are
somewhat predictable, so this is easily possible. The
result however will be a slight degradation that might
be audible.
If the worst comes to the worst and the data cannot be

corrected or concealed, the system will mute. At least
this is what is supposed to happen. Digital glitches
can be very high in level and be unpleasant to the ear
and even damage loudspeakers. Clearly though not
all equipment that is meant to mute on coming across
seriously damaged data actually does that.
Error correction and concealment takes place in

compact disc, DVD, digital radio and television. It is
also used on digital tape, but that isnt found so much
these days.
In a hard disk, CD-ROM and DVD-ROM a more powerful

error correction system is used so that in normal
operation the data that comes off the disk is perfect.
Banks and financial institutions, that use exactly the
same storage systems, would be rather less than
happy if this were not so. However, where there is a
problem that is beyond the error correction systems
ability to cope, then the data is often irretrievably
damaged.
Page 39
Latency
All digital systems exhibit the phenomenon of latency
where there is a delay between input and output on
even the shortest signal paths. In a large analog
mixing console there may be literally a kilometer
of cable inside, but a signal takes for all practical
purposes no time at all to go all the way through the
console from input to output. This is so even in a large
analog studio complex. Analog audio has no latency.
However in a digital system it takes time to convert

from analog to digital, and time to convert back again.
So even the shortest signal path has a latency of at
least a couple of milliseconds. This is generally not
audible, although care has to be taken not to mix
a signal with a delayed version of itself, or phase
cancellation will take place.
Where more processing is involved, the latency will be

longer. The worst example would be a computer-based
recording system where the signal is processed by the
computers main processor (high-end systems use
specialized digital signal processing cards). Processors
such as this are designed for general data and are
not optimized for signals. Also, the computer has to
apply its attention to monitoring the keyboard and
mouse for input, driving the display and other tasks.
In this case the latency could be well into the tens of
milliseconds, which is noticeable and sometimes off-
putting.
Clocking
An analog signal starts, ends, and goes its own
sweet way in between. It has no external reference
to time. An analog recording depends absolutely on
the mechanism of the recorder running at a precisely
controlled speed during recording and playback.
There is no information in the recording to monitor or
correct the speed.
Digital systems on the other hand are sampled

typically 44,100 times per second. This therefore binds
digital signals to a time reference. A single signal on
its own can happily work to its own time reference,
which will simply be the period between samples.
Page 40
However, when two digital signals are mixed, there is

a problem.
It is impossible to have two digital clocks that run at

the same speed. So one signal might be running a tiny
bit faster than 44. 1 kHz, the other a tiny bit slower.
So if the two signals were mixed by simply adding up
the numbers they contain, all might start off well, but
sooner or later one of the signals will have to skip a
sample to keep pace. This will cause a click, so clearly
it is undesirable.
A single signal contains its own clock. So if you wanted

to copy a signal from one machine to another, then the
machine you are copying onto can be set to external
clock and it will derive its own clock reference from
the incoming signal and run at the same speed. Often
this happens automatically so the operator is unaware
of it.
A two-machine system can run perfectly well in this

way. But as soon as you add a third digital device, which
might be a mixer or processor and not a recorder, then
it becomes difficult deciding which should be the clock
master and which slave devices should synchronize
to the master.
So in larger digital systems it is common to provide a

master clock, which is an independent unit. Everything
in the entire system will use this clock source, so
everything is sure to run at the same rate.
Page 41
Check Questions
Describe compression and rarefaction.
Describe the direction of particle motion in a transverse wave.
Describe the direction of particle motion in a longitudinal wave.
Do air molecules move from place to place under the influence of a sound wave?
Comment on point source.
What is the accepted range of frequency of human hearing?.
What is the mathematical relationship between sound velocity, frequency and wavelength?
If a sound wave travelling in air has frequency 20 kHz, what is its wavelength? (Take the
velocity of sound in air to be 340 m/s).
If a sound wave travelling in air has frequency 20 Hz, what is its wavelength? (Take the
velocity of sound in air to be 340 m/s).
Describe the convenience aspect of decibels.
Describe the logarithmic nature of the ears perception of sound levels.
If sound pressure is doubled, by how many decibels does it increase?
If sound pressure is quadrupled, by how many decibels does it increase?
If a sound has a level of 100 dB SPL, what does this mean?
Why are acoustic sounds more complex than electrical signals?
Is acoustics a completely understood science?
What are the three main factors that determine the acoustics of a room?
What is the effect of air motion on acoustics?
Why do reflections arriving later than 40 ms after the direct sound reduce the intelligibility of
speech?
What is meant by RT60?
What is a standing wave?
Comparing wavelength and room dimensions, what is the requirement for a standing wave to
be created?
Is the pressure of a standing wave high or low close to a boundary?
Can standing waves occur other than between parallel surfaces?
What is a flutter echo?
What is the worst shape for a room, acoustically?
What is the function of acoustic treatment?
What is a porous absorber?
What is a panel absorber?
Are materials that are good reflectors of sound also good sound insulators?
Are materials that are good absorbers of sound also good sound insulators?
How is sound absorbing material used in conjunction with sound reflecting material to improve
Page 42
insulation?
With reference to the question above, when is this not effective?
What property of a material controls its sound insulating ability?
What are the three requirements for good soundproofing?
What is flanking transmission?
What is pugging?
What is a dry partition?
What is the disadvantage of proprietary flexible soundproofing materials?
What should be avoided when building a double-leaf partition?
What is a floating floor?
Describe the construction of a window between a studio and control room.
What types of seals would a soundproof door have?
What is a box within a box?
What problem does soundproofing cause with regard to ventilation?
What problems do ventilation and air conditioning cause for soundproofing?
With regard to the above question, describe some of the solutions.
What is voltage?
How many microvolts (V) are there in one volt (1 V)?
Describe the difference between conventional current and real current.
What is the voltage of electrical earth?
Describe electrical current.
Describe resistance.
If voltage is divided by resistance, what is the result?
Compare resistance with reactance.
Compare resistance with impedance.
In what units do we measure impedance, resistance and reactance?
Describe electrical power.
What is a potential divider?
How may a capacitor be used to reduce the level of high frequencies?
What is the function of a transformer?
What causes noise in an analog tape recording?
If the level of an analog recording is raised to combat noise, what is the undesirable
consequence?
Describe the problems that occur in copying analog recordings.
Explain why a digital copy can be an exact clone of the master.
Briefly describe the advantages of a digital mixing console over an analog mixing console.
Briefly explain Nyquists theorem.
State the two common sampling frequencies that lie between 40 kHz and 50 kHz.
Page 43
State two other sampling frequencies that are in common professional use.
Briefly explain why it is better to sample at a higher frequency.
Describe the task of the sample and hold circuit.
Describe quantization.
What is the relevance of the number 65,536?
What is quantization error?
Roughly, what is the signal-to-noise ratio of a digital system that has 20-bit resolution?
What is the purpose of dither?
Describe jitter and the problem it can cause.
How can jitter be cured?
Describe error detection.
Describe error correction.
Describe error concealment.
If an error is detected but can neither be corrected nor concealed, what should be done, and
why?
What is latency?
Why is the term latency not relevant to analog systems?
What causes latency in digital systems?
Describe the flow of the clock signal when a digital recording is copied from one machine to
another.
What would happen if two digital unsynchronized digital signals were mixed together?
What is a master clock
Page 44

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Audio Masterclass Music Production and Sound Engineering Course

Module 01: Analog and Digital Audio

Analog and Digital Audio

The Nature of Sound

Sound starts with a vibrating source, commonly the

Let us think of a loudspeaker diaphragm. It vibrates

Wave motions are all around us, from the water

The childs toy commonly known as the slinky spring

This demonstrates a longitudinal wave where the

Contrast this with a water wave where the wave

One feature that the water wave demonstrates

This is true of sound too. A sound wave leaves a

If this were not so then either a vacuum would

In a transverse wave, such as a water wave,

Although the longitudinal wave in the slinky spring

Molecules are of course very small, and it is a feature of

[Regarding point sources - it is also worth considering

1 hertz (Hz) means one cycle of vibration per

Sound comes in virtually all frequencies but our

At the lower end of the range it is difficult to know

Frequency is related to wavelength by the formula:

velocity = frequency x wavelength

This applies to any wave motion, not just sound. The

The extreme physical size of low frequency sound

real sound travelling in air

A change in level of 3 dB means exactly the same thing

Decibels have another advantage for sound. The ear

[Sound pressure is measured in newtons per square

An important point to bear in mind is that the decibel

To convert to decibels apply the following formula in

20 x log10 (P1 /P2 )

where P1 and P2 are the two sound pressures you

10) is 0.3, and multiplying this by 20 gives 6 dB.

Actually its 6.02 dB but we dont worry about the odd

This is useful because we commonly need to, say,

We call this level the threshold of hearing and it can

If you are not comfortable with math, it is useful to

-80 dB = one ten thousandth

Do you need to understand decibels to be a sound

The answer is, Yes - to a point. You need to be able

The inverse square law

The maximum rate of decay of a sound as you move

If the sound is focused in any way, by a large source

The ultimate focused sound source is the old-fashioned

When sound is created, whether it is the human voice,

The same happens at the other end of the chain. Sound

Compare this with an electrical signal.

Once created, the signal travels in a one-dimensional

This is a little bit of a simplification, but its fair to

This is not the case with acoustics. Sound travels in

Even with todays sophisticated science and computer

a room with complete precision, accounting for every

It is normal however to complete the acoustic design

Acoustics is a complex science in practice, but in

Look around you at the various surfaces in the room. If

Some surfaces will be more absorbent, so reflections

There is a fourth factor that is worth mentioning

The Doppler effect is best demonstrated by the siren

as it goes past. Sound cant travel faster than its

In most contexts where acoustics are important, neither

What will be moving however is the air in the room

This effect is often modelled in digital reverberation

Firstly, a room that is designed for speech must