Acoustics

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 recognise.

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 when

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 instruments.

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 0’s. 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 Page � of �3 7

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 emitted.

From the perspective of a listener, the frequency of the noise being emitted will change as the object passes.

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 speaker.

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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A

C B

D

E GF

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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