Ig2 task 1 work sheet

Salford City College
Eccles Sixth Form Centre
BTEC Extended Diploma in GAMES DESIGN
Unit 73: Sound For Computer Games
IG2 Task 1
Luis Vazquez
1
Produce a glossary of terms specific to the methods and principles of sound design and production. Using a provided template, you must
research and gather definitions specific to provided glossary terms. Any definitions must be referenced with the URL link of the website you
have obtained the definition.
You must also, where possible, provide specific details of how researched definitions relate to your own production practice.
Name: RESEARCHED DEFINITION (provide short internet researched definition
and URL link)
DESCRIBE THE RELEVANCE OF
THE RESEARCHED TERM TO
YOUR OWN PRODUCTION
PRACTICE?
SOUND DESIGN
METHODOLOGY
Foley Artistry ‘A sound effects technique for synchronous effects or live effects.
Foleying is an excellent means of supplying the subtle sounds that production mikes
often miss. The rustling of clothing and a queak of a saddle when a rider mounts his
horse give a scene a touch of realism that is difficult to provide using other effects
methods. A steamy sex scene was probably created by a foley artist making
dispassionate love to his or her own wrist.
The good Foley artist must "became" the actor with whom they are synching effects or
the sounds will lack the necessary realism to be convincing.’
http://www.filmsound.org/terminology/foley.htm
Foley artistry is relevant to the
production of video games due to the
necessity for sounds to be appropriate
for any actions or events that occur in
the space of the game. The ability to
apply sounds to these actions makes a
large difference on the resulting
products ability to portray itself to an
audience realistically or appropriately.
Sound Libraries ‘A sample library is a collection of digital sound recordings, known as samples, for
use by composers, arrangers, performers, and producers of music. The sound files are
loaded into a sampler - either hardware or computer-based - which is then used to
create music.’
http://en.wikipedia.org/wiki/Sample_library
Sound libraries are important to a
production due to the large amount of
sounds that a given project will require,
meaning that a prepared ‘library’ of
sound assets will reduce the amount of
newly recorded sound necessary to
complete a project.
SOUND FILE FORMATS Uncompressed ‘Uncompressed audio files are the most accurate digital representation of a
soundwave, but can also be the most resource-intensive method of recording and
storing digital audio, both in terms of storage and management. Their accuracy makes
them suitable for archiving and delivering audio at high resolution, and working with
audio at a professional level, and they are the 'master' audio format of choice.’
http://www.jiscdigitalmedia.ac.uk/guide/uncompressed-audio-file-formats
When producing or recording sound,
very often the audio created will be
uncompressed, which is useful because
very often, it is undesirable to lose any
of the data from a file until it processed
and exported for use.

IG2 Task 1
Luis Vazquez
2
.wav ‘A Wave file is an audio file format, created by Microsoft, that has become a standard PC audio
file format for everything from system and game sounds to CD-quality audio. A Wave file is
identified by a file name extension of WAV (.wav). Used primarily in PCs, the Wave file format
has been accepted as a viable interchange medium for other computer platforms, such as
Macintosh. This allows content developers to freely move audio files between platforms for
processing, for example.
In addition to the uncompressed raw audio data, the Wave file format stores information
about the file's number of tracks (mono or stereo), sample rate, and bit depth.’
http://whatis.techtarget.com/definition/Wave-file
.wav is the format that the majority of
sound manipulation and production
programs export in, and as .wav files
will have the vast majority of the data
for a sound file; the format is useful as
‘raw’ data that can be modified into a
different file type.
.aiff ‘AIFF is short for Audio Interchange File Format, which is an audio format initially created by
Apple Computer for storing and transmitting high-quality sampled audio data. It supports a
variety of bit resolutions, sample rates, and channels of audio. This format is quite popular
upon Apple platforms, and is commonly adopted in professional programs that handle digital
audio waveforms.
AIFF files are uncompressed, making the files quite large compared to the ubiquitous MP3
format. AIFF files are comparable to Microsoft's wave files; because they are high quality they
are excellent for burning to CD. There is also a compressed variant of AIFF known as AIFF-C or
AIFC, with various defined compression codecs. ‘
http://www.abyssmedia.com/formats/aiff-format.shtml
.aiff is a file type that is commonly
associated with Apple computer
systems and programs. As it is an
uncompressed file format, files of this
type are fairly large in size.
.au ‘The Au file format is a simple audio file format introduced by Sun Microsystems. The format
was common on NeXT systems and on early Web pages. Originally it was header less, being
simply 8-bit µ-law-encoded data at an 8000 Hz sample rate. Hardware from other vendors
often used sample rates as high as 8192 Hz, often integer factors of video clock signals. Newer
files have a header that consists of six unsigned 32-bit words, an optional information chunk
and then the data (in big endian format).
Although the format now supports many audio encoding formats, it remains associated with
the law logarithmic encoding. This encoding was native to the SPARCstation 1 hardware,
where SunOS exposed the encoding to application programs through the /dev/audio interface.
This encoding and interface became a de facto standard for Unix sound. ‘
https://www.princeton.edu/~achaney/tmve/wiki100k/docs/Au_file_format.html
.smp ‘An ".smp" file may be one of several different types of audio file. For example, it could be a
SampleVision audio sample file. This 16-bit audio file was originally used by Turtle Beach
SampleVision; you can open it with Adobe Auction, Sound Forge Pro or Awave Studio. It could
also be a sample file for AdLib Gold, a PC sound card released in 1992; Scream Tracker, a mid-

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3
1990s music editing program; or Swell. Reason, a music recording and production program,
uses the ".smp" extension for sampler instrument patches.’
http://www.ehow.com/info_12198596_file-smp.html
Lossy Compression ‘Lossy file compression results in lost data and quality from the original version. Lossy
compression is typically associated with image files, such as JPEGs, but can also be used for
audio files, like MP3s or AAC files. The "lossyness" of an image file may show up as jagged
edges or pixelated areas. In audio files, the lossyness may produce a watery sound or reduce
the dynamic range of the audio.
Because lossy compression removes data from the original file, the resulting file often takes up
much less disk space than the original. For example, a JPEG image may reduce an image's file
size by more than 80%, with little noticeable effect. Similarly, a compressed MP3 file may be
one tenth the size of the original audio file and may sound almost identical.
The keyword here is "almost." JPEG and MP3 compression both remove data from the original
file, which may be noticeable upon close examination. Both of these compression algorithms
allow for various "quality settings," which determine how compressed the file will be. The
quality setting involves a trade-off between quality and file size. A file that uses greater
compression will take up less space, but may not look or sound as good as a less compressed
file. Some image and audio formats allow lossless compression, which does not reduce the
file's quality at all.’
http://www.techterms.com/definition/lossy
Lossy compression is a method of
processing audio into files that can
maintain similar a quality of audio while
reducing the amount of space that the
file takes up by ‘losing’ data from the
‘raw’ files in such a way that the
difference would not be very
pronounced from the output.
.mp3 ‘The name of the file extension and also the name of the type of file for MPEG, audio layer 3.
Layer 3 is one of three coding schemes (layer 1, layer 2 and layer 3) for the compression of
audio signals. Layer 3 uses perceptual audio coding and psychoacoustic compression to
remove all superfluous information (more specifically, the redundant and irrelevant parts of a
sound signal. The stuff the human ear doesn't hear anyway). It also adds a MDCT (Modified
Discrete Cosine Transform) that implements a filter bank, increasing the frequency resolution
18 times higher than that of layer 2.
The result in real terms is layer 3 shrinks the original sound data from a CD (with a bit rate of
1411.2 kilobits per one second of stereo music) by a factor of 12 (down to 112-128kbps)
without sacrificing sound quality.
Because MP3 files are small, they can easily be transferred across the Internet.’
http://www.webopedia.com/TERM/M/MP3.html
.mp3 files are a commonly used file
type for compressed audio. It is a lossy
compression audio type, which means
that it loses data and quality from the
‘raw’ audio files in order to create
smaller files.
AUDIO LIMITATIONS Sound Processor Unit (SPU) ‘A sound card (also known as an audio card) is an internal computer expansion card that
facilitates the input and output of audio signals to and from a computer under control of
computer programs. The term sound card is also applied to external audio interfaces that use
software to generate sound, as opposed to using hardware inside the PC. Typical uses of
sound cards include providing the audio component for multimedia applications such as music
composition, editing video or audio, presentation, education and entertainment (games) and
video projection.
Sound Processors are vital to a
production process, as the method by
which a computer interprets and
outputs sound. Sound cards also affect
how quickly and how well a computer
can receive and output sound of a given
quality, meaning that the capabilities

IG2 Task 1
Luis Vazquez
4
Sound functionality can also be integrated onto the motherboard, using basically the same
components as a plug-in card. The best plug-in cards, which use better and more expensive
components, can achieve higher quality than integrated sound. The integrated sound system is
often still referred to as a "sound card". ‘
http://en.wikipedia.org/wiki/Sound_card
and compatibilities of sound cards must
be taken into consideration when
producing audio based media.
Digital Sound Processor
(DSP)
‘Digital Signal Processors (DSPs) take real-world signals like voice, audio, video, temperature,
pressure, or position that have been digitized and then mathematically manipulate them. A
DSP is designed for performing mathematical functions like "add", "subtract", "multiply" and
"divide" very quickly.
Signals need to be processed so that the information that they contain can be displayed,
analyzed, or converted to another type of signal that may be of use. In the real-world, analog
products detect signals such as sound, light, temperature or pressure and manipulate them.
Converters such as an Analog-to-Digital converter then take the real-world signal and turn it
into the digital format of 1's and 0's. From here, the DSP takes over by capturing the digitized
information and processing it. It then feeds the digitized information back for use in the real
world. It does this in one of two ways, either digitally or in an analog format by going through
a Digital-to-Analog converter. All of this occurs at very high speeds.
During the recording phase, analog audio is input through a receiver or other source. This
analog signal is then converted to a digital signal by an analog-to-digital converter and passed
to the DSP. The DSP performs the MP3 encoding and saves the file to memory. During the
playback phase, the file is taken from memory, decoded by the DSP and then converted back
to an analog signal through the digital-to-analog converter so it can be output through the
speaker system. In a more complex example, the DSP would perform other functions such as
volume control, equalization and user interface. ’
http://www.analog.com/en/content/beginners_guide_to_dsp/fca.html
Digital Sound Processors are important
for a sound production that intends to
use recorded sound. DSPs allow signals
that have been converted to data to be
manipulated, and can also process
manipulated data to transfer it into a
digital or analogue form.
Random Access Memory
(RAM)
‘Alternatively referred to as main memory, primary memory, or system memory, Random
Access Memory (RAM) is a computer storage location that allows information to be stored
and accessed quickly from random locations within DRAM on a memory module. Because
information is accessed randomly instead of sequentially like a CD or hard drive the computer
is able to access the data much faster than it would if it was only reading the hard drive.
However, unlike ROM and the hard drive RAM is a volatile memory and requires power in
order to keep the data accessible, if power is lost all data contained in memory is lost.’
http://www.computerhope.com/jargon/r/ram.htm
Random Access Memory is an
important part of all processes of a
computer, but it is especially relevant in
sound production where a sound may
need to be played back and processed
repeatedly, so the faster and more
capable RAM is on the computer, the
speedier the production process.
Mono Audio ‘Mono or monophonic describes a system where all the audio signals are mixed together and
routed through a single audio channel. Mono systems can have multiple loudspeakers, and
even multiple widely separated loudspeakers. The key is that the signal contains no level and
arrival time/phase information that would replicate or simulate directional cues. Common
types of mono systems include single channel centre clusters, mono split cluster systems, and
distributed loudspeaker systems with and without architectural delays. Mono systems can still
Mono Audio is useful when producing
sound that does not need, or benefits
from not having directional
information. As all audio is directed
through a single channel in Mono the
signal is consistent for each system that

IG2 Task 1
Luis Vazquez
5
be full-bandwidth and full-fidelity and are able to reinforce both voice and music effectively.
The big advantage to mono is that everyone hears the very same signal, and, in properly
designed systems, all listeners would hear the system at essentially the same sound level. This
makes well-designed mono systems very well suited for speech reinforcement as they can
provide excellent speech intelligibility.’
http://www.mcsquared.com/mono-stereo.htm
outputs the sound.
Stereo Audio ‘True stereophonic sound systems have two independent audio signal channels, and the
signals that are reproduced have a specific level and phase relationship to each other so that
when played back through a suitable reproduction system, there will be an apparent image of
the original sound source. Stereo would be a requirement if there is a need to replicate the
aural perspective and localization of instruments on a stage or platform, a very common
requirement in performing arts centres.
This also means that a mono signal that is panned somewhere between the two channels
does not have the requisite phase information to be a true stereophonic signal, although there
can be a level difference between the two channels that simulates a position difference, this is
a simulation only. That's a discussion that could warrant a couple of web pages all by itself. ‘
http://www.mcsquared.com/mono-stereo.htm
Stereo Audio systems use two audio
channels to create the simulation of
sound that has directional information,
or so that sound can be processed as if
it did have directions associated with it.
This is useful when creating sound that
is simulated to exist within a ‘physical’
space (such as a character’s movements
in a video).
Surround Sound ‘Surround sound audio is, simply put, sound that completely surrounds you. It means a
speaker in virtually every corner of the room, projecting high-quality digital sound at you from
all angles just as though you were in a theater.
In terms of nuts and bolts, it means a set of speakers, usually five including the all-important
“center speaker,” and a subwoofer for powerful bass. This is where the term “5.1” comes from
-- five speakers and a subwoofer. ‘
http://peripherals.about.com/od/speakersandheadphones/a/whatis_ss.htm
Surround Sound is a phrase that can
refer to any number of audio systems
that have in common the assumption of
existing in an approximate 360 degree
space around a recipient of the audio
output.
Direct Audio (Pulse Code
Modulation – PCM)
‘Pulse code modulation (PCM) is a digital representation of an analog signal that takes samples
of the amplitude of the analog signal at regular intervals. The sampled analog data is changed
to, and then represented by, binary data. PCM requires a very accurate clock. The number of
samples per second, ranging from 8,000 to 192,000, is usually several times the maximum
frequency of the analog waveform in Hertz (Hz), or cycles per second, which ranges from 8 to
192 KHz.’
http://www.techopedia.com/definition/24128/pulse-code-modulation-pcm
PCM is a form of interpretation of an
analogue system into a digital form,
which is useful for allowing digital
systems to interpret and use data from
an analogue system or storage media.
AUDIO RECORDING
SYSTEMS
Analogue ‘Simply put, an analog system sends information by encoding it as a continuous change in
voltage or current. The receiving end then decodes these changes back into usable
information. For example, an analog clock represents the sun circling the sky. An analog device
converts a pattern, such as light or sound patterns, into electrical signals with similar patterns.
Analogue systems use a single
continuous signal as a pattern to
represent information, such as audio.
Since it relies on these small physical

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Luis Vazquez
6
The definition of an analog signal can also reveal its shortcomings. Analog signals, because
they rely on minute variations in voltage or current, are very susceptible to interference from
outside sources. Even a small change in the analog signal from an outside source, such as
electromagnetic interference, can cause dramatic changes in the signal and how the signal is
interpreted.’
http://www.theprojectorpros.com/learn-s-learn-p-theater_analog_digital.htm
changes from a single continuous signal
to interpret information, it is very
susceptible to interference from
physical sources of any magnitude.
Digital Mini Disc ‘One easy way to think about a MiniDisc is like a floppy disk -- you can record and erase files
on a MiniDisc just as easily as you can on a floppy disk. The big difference between the a
MiniDisc and a floppy disk is that a MiniDisc can hold about 100 times more data (about 140
megabytes in data mode, 160 megabytes in audio mode vs. 1.44 megabytes for a floppy).’
http://electronics.howstuffworks.com/question55.htm
DMDs were an older storage medium
that have largely fallen out of use. It
had the advantage over early CDs in
that DMDs were more easily recorded
onto.
Compact Disc (CD) ‘A compact disc [sometimes spelled disk] (CD) is a small, portable, round medium made of
molded polymer (close in size to the floppy disk) for electronically recording, storing, and
playing back audio, video, text, and other information in digital form. Tape cartridges and CDs
generally replaced the phonograph record for playing back music.’
http://searchstorage.techtarget.com/definition/compact-disc
CDs are one of the most commonly
used physical storage mediums, and are
compatible with most computers and
other disc-based devices, which are
often built with CD reading hardware.
Digital Audio Tape (DAT) ‘Digital Audio Tape (DAT or R-DAT) is a signal recording and playback medium developed
by Sony and introduced in 1987.[1]
In appearance it is similar to a compact audio cassette,
using 4 mm magnetic tape enclosed in a protective shell, but is roughly half the size at 73 mm
× 54 mm × 10.5 mm. As the name suggests, the recording is digital rather than analog. DAT has
the ability to record at higher, equal or lower sampling rates than a CD (48, 44.1 or
32 kHz sampling rate respectively) at 16 bits quantization. If a digital source is copied then the
DAT will produce an exact clone, unlike other digital media such as Digital Compact Cassette or
non-Hi-MD MiniDisc, both of which use lossy data compression.’
http://www.princeton.edu/~achaney/tmve/wiki100k/docs/Digital_Audio_Tape.html
MIDI ‘MIDI stands for Musical Instrument Digital Interface. The development of the MIDI system
has been a major catalyst in the recent unprecedented explosion of music technology. MIDI
has put powerful computer instrument networks and software in the hands of less technically
versed musicians and amateurs and has provided new and time-saving tools for computer
musicians. The system first appeared in 1982 following an agreement among manufacturers
and developers of electronic musical instruments to include a common set of hardware
connectors and digital codes in their instrument design. In 1983, the MIDI 1.0 Specification
was formally released by the International MIDI Association* as Roland, Yamaha, Korg, Kawai
and Sequencial Circuits all came out with MIDI-capable instruments that year.’
http://www.indiana.edu/~emusic/etext/MIDI/chapter3_MIDI.shtml
MIDI is the standard interface that is
common to the software that is used in
the production of music, which means
that I will use it to create sounds that
have not been recorded using a
microphone, such as music.
Software Sequencers ‘A sequencing software package designed to be loaded into a computer. Software sequencers
usually have more features and has the advantage of showing you a lot more information at
Software sequencers are important to
my production process, as I do not use

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7
once because they use the computer's screen and aren't locked into the knobs or buttons or
display of a hardware sequencer.’
http://www.wannaplaymusic.com/get-started/keyboard-terminology
specialised hardware to manipulate
sound files. These programs give me an
interface to manipulate and create
sound directly and independently.
Software Plug-ins ‘A plug-in is a (sometimes essential) piece of software code that enables an application or
program to do something it couldn’t by itself. One of the more common plug-ins is Adobe
Flash Player. Without Flash Player you won’t, for example, be able to view BBC News bulletins
embedded into web pages.
Other plug-ins are available for different things. There are plug-ins for social media
networking, foreign language alphabets and many other things. One plug-in allows for the
display of Microsoft Office 2007 documents within the browser.’
http://www.bbc.co.uk/webwise/guides/about-plugins
Plug-ins are necessary to my production
process, as the base software for
programs such as sequencers are
limited in the types of manipulations
that they can achieve. Using plug-ins, it
becomes possible to create new sounds
that were not possible with the options
available without them.
MIDI Keyboard Instruments ‘A Musical Instrument Digital Interface (MIDI) keyboard is a musical instrument like a piano
keyboard. The MIDI portion indicates that the instrument has a communication protocol built
in that allows it to communicate with a computer or other MIDI-equipped instrument.
The MIDI interface is now so easy to implement that almost all keyboards sold today are some
type of MIDI keyboard. This ranges from a simple 100 US dollar (USD) MIDI keyboard sold at
the local department store to a 30,000 USD grand piano with a built-in controller. Every type
can connect to any other type of musical instrument that sports a MIDI interface. The 30,000
USD instrument will sound much better than the 100 USD instrument, but both can be
controlled by the computer or other instrument in the same way.’
http://www.wisegeek.com/what-is-a-midi-keyboard.htm
MIDI instruments allow for in-software
productions to be created using real
world musical skill, as an interface to
that software. MIDI instruments also
themselves often are created and
distributed with options for the
creations of sounds using the interface,
which can be simulated in Software
Sequencers using plug-ins.
AUDIO SAMPLING File Size Constraints - Bit-
depth
‘Audio recorded at a lower bit depth will sound grainy and less defined therefore recording
audio at 24 bit will be of a higher quality than something recorded at 16 bit. So why doesn't
everyone record at 24 bit? Well we have identified the benefits of recording at higher bit
depths but what about the costs? If something is recorded at 24 bits then it is using more bits
in a file and as such the files are going to be much bigger. They will take up more disc space
and they also require more computing power to process. So how do you choose what bit
depth to work in? The best option is to provide yourself with the choice, buy an audio
interface capable of recording at higher bit depths but only use the higher bit depths if you
know your project will benefit from it. For example it may be more efficient to stick to 16 bits
if your project is aimed at FM transmissin or internet streaming.’
http://www.dolphinmusic.co.uk/article/120-what-does-the-bit-depth-and-sample-rate-refer-
to-.html
Bit depth is relevant in sound
production as a method of balancing
the quality of a given audio file, and the
amount of space that that file takes up,
which can be changed to suit a variety
of practical purposes.
File Size Constraints -
Sample Rate
‘Basically, sample rate is how often the ADC (Analog to Digital Converter) takes a snapshot of
the audio. Ideally, the sample rate would be infinite, so the ADC was continuously translating
the analog signal to digital, but that would require infinite disk space, so we settle for a less
than perfect translation, and hopefully we can’t notice the difference. ‘
The sample rate of an audio file affects
how precisely it is read by any given
software, which means that the sound
taken from that file will be of a higher

IG2 Task 1
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8
http://progulator.com/digital-audio/sampling-and-bit-depth/
quality. However, higher sample rates
increase the size of a file exponentially.

Ig2 task 1 work sheet

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