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Section 1 - The Physics of Sound

Sound Waves are the compression and rarefaction of
molecules through a medium. For sound to exist, vibrations
must occur.
For example, when this tuning fork is hit, the right fork bends
outwards. This causes all of the air molecules around the fork
to be pushed together and causes a compression of air. As
the fork then moves back to try and reach its originally
position, it spreads out the air molecules causing a
rarefaction of the air. This will repeat until the fork has
stopped vibrating and has regained its original position.
The diagram to the right shows compression and rarefaction
in a 2D form but as sound is omnidirectional, these waves will
be emitted in every direction simultaneously.
In most instances, people will recognise a sound wave as looking something like the below
diagram. This is know as a waveform and is a visual representation of the compression and
rarefaction of the air. As shown below, the compressions of air molecules results in the peaks of
the sound waves and the
rarefactions result in the
troughs and this is what gives
us the waveform that we
However, not all waveforms
will look the same as this
diagram and this is
determined by the frequency
of the sound. Frequency is
measured in Hertz (Hz) and is the amount of complete compressions and rarefactions in a second.
The higher the note, the more compressions and
rarefactions occur every second and the lower the note, the
less happen every second. The diagram below shows an
example of a high frequency and a low frequency.
Waveforms are also affected by the Attack, Decay, Sustain
and Release of the sound. The Attack of the sound is how
fast it takes for the sound to reach its peak volume. The
Decay of the instrument is the time taken for the sound to
drop until the sound is just resonating. The Sustain is how
loud the
note is
held on with the release being how long the note
takes to extinguish when the note has been let
go of.

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Sound travels at a speed of 344 m/s. Some aeroplanes are capable of exceeding the speed of
sound and this results in a sonic boom where by the object emitting the sound is traveling in front
of the sound waves it is emitting. It is similar to the Dopler effect where by the sound waves bunch
up to create a perceived sound with higher pitch due to the object moving.
Instead, the object surpasses the sound waves and this causes a high pressure sound wave and it
is this that causes the boom which can be heard from miles away.
When there are two microphones picking up the same sound source, the wave lengths can
sometimes be out of line. For example, the compression of one microphone could be at the same
time as the rarefaction of when combined, this will result in each microphone cancelling each other
out and the guitar loosing its tone and volume. Sometimes the waveforms will not align and this
can make the recordings sound unnatural. This is called being out of phase and can be fixed by
slightly shifting the audio recordings so that they match up and this will mean that the recordings
sound more natural and of a better quality.
Every instruments sound is built on a multiple of different notes. The fundamental note is the one
which the instrument is tuned to. For example, when a C note is played on a piano, you can tell it is
a piano and this is due to harmonics. Without these harmonics, the note would just be a pure Sine
wave and would just sound like a pure note. Instead,
each instrument has multiple different frequencies that
play at the same time as the note and this is what gives
the instrument its tone and distinct sound above other
Sound is measured in Decibels. The Decibel scale is a
logarithmic measure of the power produced by sound.
As it was originally intended to be used to measure the
power intensity along telephone lines, it was known as
the Bel scale. It is used in comparison between the
threshold of hearing and the comparison This allows
large intensity values to be reduced to smaller numbers,
simply by counting the number of 0s. However this
would have meant that there would only be 12 numbers
on the scale.
This is because the threshold of pain has a power
intensity of 1,000,000,000,000. Instead it was decided
that the Bel system would be multiplied 10 meaning the
scale would run from 1-120 which is where the Deci-bel
system came from.

Section 2 - The Principles of Musical Instruments

All musical instruments are divided into families depending on how their sound is created. Each
family has different characteristics. For example, within the woodwind family, there are two main
types of instrument; reed instruments and flute instruments
The Reed instruments generate their sound by focusing air at a reed which then sends vibrating air
down a large column to produce the sound. The pitch the of the instruments can be changed by
covering the holes in the column and this extends how far the air has to travel - resulting in a lower
pitch. An Alto Saxophone will usually produce frequencies ranging from 147Hz to 880Hz.

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Flute instruments create their sound by having a focused stream of air across the the hole in the
side of a tubular column. This creates the vibrating air which then resonates down the column of
air. The method used to change the pitch of the instrument can also be changed by covering the
holes and extending how far the air has to travel. A flute will produce frequencies between 262Hz
to 2.6KHz.

Stringed instruments create their sound by causing a string to vibrate. Classical stringed
instruments such as Violins, Cellos and Double Basses are all played by dragged a bow across the
strings. This causes the string to vibrate causing the noise. On these classical stringed
instruments, the sound is amplified by vibrating the bridge which rests on a hollow body and this
causes the sound to be amplified.

Other stringed instruments such as guitars can be plucked or strummed to form a chord. Stringed
instruments can be tuned by tightening the or loosening the string. The tighter the string the higher
the pitch of the note. Stringed instruments can have their pitches changed by pressing down to
shorten the string. This means a faster
vibration and will result in a higher
pitch being played. Acoustic guitars
work very similar to orchestral
instruments by using the bridge to
amplify the sound. Electric guitars
have solid body made from wood and
use pickups and electric circuits to
amplify the strings being played.

Percussion instruments are played by

hitting or striking an object. In the
instance of drums, a material is
stretched tightly across a circular
wooden shell and this produces a
sound. All percussion instruments are
tuned in some way whether this be by the owner of in the factory during production. Percussion
instruments such as the Xylophone have wooden blocks that when hit with mallets produce a
pitched note.
A tuned instrument is one that is designed and engineered to vibrate at a specific frequency. For
example, one block of the Xylophone will be designed to vibrate at 330Hz if it is intended to
produce an E note. Tuned instruments are those which can have their note changed either by
tightening a string or a drum head ect. The diagram above shows the frequencies that are
produced by different instruments and what notes they can produce.

Section 3 - The Mechanisms of Human Hearing

The human ear is composed of six different main parts.
The Pinna is the cartilage on the outside of the head that
are what we all know as Ears. They are designed to
channel sound waves down the ear canal. The ear canal
then focuses the sound towards the Tympanic membrane
or eardrum. As the eardrum moves, it causes three tiny
bones behind it to vibrate. These are called the ossicles.
The third bone of the ossicles knocks into the Cochlea or
Inner Ear and this causes the liquid in the Cochlea to move
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and this causes a response in the auditory nerve.

When talking about hearing, there are two limits that are important. The Threshold of Hearing is the
lowest audible sound that a human can hear. This has been identified as a sigh 10 meters from
your ear. This has a relative loudness and is the equivalent to 0dB. The limit of human hearing is
known as the Threshold of Pain and this is identified as your ear being a few centimetres from a
road drill and this is the equivalent to 120dB.
The brain can perceive sound in many different ways and the study
of this is called Psychoacoustics. There are many different
experiments that are used to demonstrate the brains way of seeing
sound. The Cocktail Party Effect is the brains ability to be able to
specifically shut out background noise and focus on one sound or
voice. For example at a cocktail party, you can listen to one
conversation and ignore all background noise and this is therefore
known as the Cocktail Party Effect.
Another experiment used to demonstrate the brains power is known as Haas. This is where
multiple sounds of different frequencies are played at exactly the same time and appear to the
listener as one sound and the brain is unable to distinguish between them.
Masking is another example of the brains capacity to control the way that sounds are perceived.
When a quiet sound is met with a louder sound, the quieter sound is over powered and left
inaudible or unable to be heard easily.
When an object moves it emits sound. If the object is moving fast enough, it will start to catch up
with the sound waves. This means that in front of the vehicle, the sound waves will bunch up and
this will cause an increase in frequency in front of the object. Behind the object, the sound waves
will be more spread out and this will give the
impression of a lower frequency sound being
From the perspective of a listener,
the frequency of the noise being
emitted will change as the object

In a work place, no employee is exposed to noise above an average level of 85 decibels over eight
hours, or a peak level of 140 decibels - whether or not the employee is wearing a personal hearing
protector. This is to ensure that no permanent damage is done to anybodies ears whilst working.
Employers should aim to have employees being exposed to no more than 85dB if they are wearing
ear protectors and should also provide ear protection should it be requested.

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Section 4 - The Acoustic Characteristics of Spaces

The Picture to the right shows a small recording studio. The
red arrows show the first point of reflection from the right
When sitting and listening in this position, the engineer
would hear both the original mix straight from the speakers
as well as the reverb of the room. This would give a false
impression of reverb on the final mix.
If you look at the back of the room you can see a staggered
surface. This is designed to disperse the sound reflections
and stop them from bouncing off of the wall. This staggered
surface would most likely be made from a semi-hard
surface like foam or fibreglass.
Softer more dense materials absorb sound waves better
than harder surfaces. Studios are designed with no
opposing surfaces. For example, all walls are offset and
this is so that sound waves are dispersed rather than
continuing to bounce off of each other.
Places such as the Royal Albert Hall are designed to have
a large amount of Reverb. The Royal Albert Hall is
designed to have a violin player stand on stage and be
heard by every person in the venue without any amplification.
In a recording environment, an engineer would want to record the best raw sound as engineers are
able to add reverb and other effects after it has been recorded but once recorded, this cannot be
removed in the original audio track.
Most studios are acoustically treated. Every material used to acoustically treat a space is given a
sound absorption coefficient rating. This is rated on a scale on 0 to 1. One is the most absorptive
with zero being no absorption and full
reflection. Each material is given a
different number for various frequencies
as various frequencies
Rooms can be sound proofed by using
soundproofing insulation which is usually
made from fibreglass which absorbs
sounds as they pass through the wall.
Floors and ceilings are also insulated to
prevent very little sound coming in or out.

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Section 4.2 - Designing a studio


Using graphic design software I have created a floor plan of my bedroom and added studio
furniture to help explain the best way to adapt a space. When designing a studio from scratch you
should try to eliminate any parallel walls and 90 degree corners. Opposing parallel walls with cause
you lots of problems when trying to listening back to your mix as the sound waves will continue to
bounce of the walls until they loose their power. This will result in an unattractive reverb being
generated by using your speakers.
To adapt a space into a studio, you must first place all of your equipment in. In the above design, i
have added a desk, five speakers in a surround sound formation, a large wall mounted TV as a
monitor for my system as well as six acoustic panels. Each panel is labelled with a letter and i will
use this to try and explain why i have placed it in the position shown. When referring to the
speakers, i will describe them from the point of view of the engineer.
Panel A is placed at the point of first reflection for both the front and back left hand speakers. This
is essentially where the sound produced from the speakers will first hit the wall. If there were no
panel here, the sound would be reflected around the room giving a false impression of the mix.
Panels B & C are placed behind the front left and right speakers. This is because although the
speakers are pointing forwards, they also produce slight sound from the back of the speaker.
Therefore, panels B & C are there to prevent the back of the speakers from reflecting any sound
back towards the engineer. There would also be a panel placed behind the central front speaker for
the same reason, however to show where the TV would be placed, I had to leave this out.

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Panel D is similar to Panel A in the fact that it is placed at the point of first reflection on the wall. It is
noticeable bigger than Panel A. This is because the speakers are further from the wall and so the
first reflection for the front speaker will be further back and for the back speaker it will be further
forwards. This means that it is more efficient to place one large panel that will absorb both
speakers sound.
Panels E & G are placed behind the back two speakers to absorb any back reflections from the
speakers and are similar to C and B in relation to purpose. Again this is to prevent any unwanted
sound from bouncing back to the engineer and giving a false impression of the mix.
Panel F would be used to stop the reflection of the central front speaker from then bouncing back
to the listener. Due to the speaker being straight, the sound would travel straight forwards resulting
in it hitting the point where the panel is.
The panels used would be Acoustic tile mounted directly to the wall. This would be an absorption
panel that would absorb the sound energy that has already reached the listeners ears. The aim of
the acoustic panels is to aim for the most neutral and dead room with no reverb from the room
itself. The panels would have an absorption coefficient of 0.8 and these are one of the most
efficient panels for their size.

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