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Introduction
The music / sound
design / computer / film industry is so full of…jargon…someone
had to translate all the terminology into Plain English. Since we are
fluent in this language at Bob & Barn, we felt the responsibility
fell on us. Impress your dinner party companions, as a result, with your
knowledge of adaptive differentials and pitch bending. (Just a suggestion.)
5.1
Term used to describe
a multi-channel speaker setup / environment used primarily for cinema
sound but is slowly seeping into consumer home cinema setups. The ‘5’
refers to the 5 main speakers – 1 each for left, centre, right,
left surround and right surround. These all need to be placed in the correct
orientation in the sound field to give the desired effect. The ‘.1’
refers to the sub-woofer. This is where all the lower frequencies of any
given soundtrack are piped. There is no need to place it anywhere specifically
in the sound field as it is difficult for the ear to identify the direction
and the location of low frequency sounds. The most common format is Dolby
Digital; other formats are DTS and Sony’s SDDS.
ADC
/ DAC - Analogue to Digital Converter / Digital to Analogue Converter
Converts audio sound
pressure waves (any form of analogue sound) to discreet digital information
(binary 1’s and 0’s) and vice versa. The quality of these
converters can vary enormously from ‘consumer’ home CD-Players
costing hundreds to hi-end professional digital equipment costing thousands
and tens of thousands.
(Tech Bit) The benchmark
for these converters are based on what’s called the Signal to Noise
Ratio (SNR). It’s measured in Decibels, a logarithmic scale from
0 to 120 DB (0 being absolute silence and 120 being the threshold of pain
– a plane taking off next to you!). The higher the value is to 120,
the better the converters will be. There are other issues to take into
consideration, but it’s a good starting point.
ADPCM
- Adaptive Differential (or Delta) Pulse Code Modulation
A form of audio file
compression that roughly equates to 3.5:1 file size by converting a 16-bit
file to 4-bit (and some clever maths!) It was initially adopted by Nintendo
as the hardware audio compression (called Bit Rate Reduction) on the Super
Nintendo game console in the early 90s but it’s biggest boost came
with Sony adopting the standard for Playstation audio and Microsoft supplying
it as standard with Windows 95. It is now used on the phone exchanges,
Nintendo GameCube, Microsoft X-Box and Sony’s Minidisc.
The reason behind
its popularity is due to the small amount of CPU overhead required to
decode the signal and relatively low perceptual loss in audio quality.
(Mp3, although a lot better compressor, requires a significant amount
of processing power to decode).
AIFF
- Audio Interchange File Format
This is the standard
file format for audio on Apple Macs. Similarly to Windows, files can support
different sample rates and compression codecs (see codec definition.)
Unlike WAV files, AIFF files only support mono or stereo.
(Tech Bit) Files are
coded Little Endian / Big Endian on a Mac and vice versa on a PC for a
WAV. Most PC / MAC audio programs will now be able to open (and may even
save) files in both format to give cross platform compatibility.
API
- Application Programming Interface
Essentially, this
is a computer program / self-contained module written to control something
specific. We use it to describe an audio API - a program that runs alongside
the main game controlling all of the audio in real-time allowing the programmer
to manipulate sounds on the fly. An example of this is to make a wind
sample more blustery the higher up the player goes.
AU
This is the audio
format for Sun Unix workstations. Can support 8-32 bit samples in mono
or stereo. Native compression is U-Law.
CODEC
- Compression-Decompression
A codec is an algorithm
that compresses or decompresses audio / video. ADPCM and MP3 both require
specific codecs. For compatibility purposes, if you wish to create a file
that can be decompressed on any PC or MAC, make sure that is one that
is installed by default.
CUE
A film and TV term
used to describe a piece of music of any length written to accompany moving
picture.
EQ
- Equalisation
Everyone is familiar
with the bass and treble controls on an amp or stereo, this is the most
basic form of EQ. Mixing Consoles and other Pro-Audio equipment just take
this to the nth degree by assigning complete control over the entire frequency
range (20 – 20,000 Hz). An EQ is just another name for a filter.
(Tech Bit) This is
done normally with 2 shelving EQ’s which control the low and high
frequencies (e.g. 80 Hz and 12,000 Hz) and 1-3 peaking EQ’s in the
middle ranges which allow you to specify which frequencies you would like
to control and how much you add or subtract to them. The ‘Q’
setting (if you have one) will control the width of that frequency response
– (how much you want that setting to control frequencies either
side).
FILTERING
A filter can be viewed
as a “sieve” on sound! It is simply a device (whether it be
in software or hardware) that alters the frequency range of a sound. A
low pass filter will only let the lower frequencies through and cut off
any frequencies over a given value making it sound muffled. A high-pass
filter is the reverse. You can filter things over time very smoothly to
use as a musical effect.
A good example is
listening to someone walking towards you whilst you are behind a heavy
closed door. You can hear the footsteps coming, but only the lower frequency
components. Once you open the door, you can hear it with far greater clarity
because the door isn’t obstructing the sound and absorbing the hi-frequencies.
FMV
- Full Motion video
Term coined by the
video Games industry to describe the fancy pre-rendered animations normally
seen in-between levels to help develop the story line. Because they are
effectively linear animations streamed from the main storage device (be
it hard drive or CD/DVD) they don’t suffer from the same limitations
that in-game graphics do. Therefore they can be as lavish and over the
top as the technology used to create them will allow. They can be thought
of as short films.
FOLEY
Foley is the name
given to the techniques employed for creating synchronous effects or live
effects, named after Jack Foley, a sound editor at Universal Studios.
Foley-ing is an excellent
means of supplying the subtle sounds that production mikes often miss.
The rustling of clothing and the squeak 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 in a film was probably,
in reality, created by a Foley artist making passionate love to his /
her own wrist.
(From filmsound.org’s
definition of Foley – see http://www.filmsound.org/terminology/foley.htm)
FTP
- File Transfer Protocol
The pre-cursor to
HTTP (which is the programming protocol that allows us to view web pages)
FTP was the first standard by which files were transmitted over the Internet
and as such, is very robust and powerful. You can view public FTP sites
anonymously using standard web browsers, or log onto a private FTP site
using a username and password. This connection, when initiated can be
seen as a secure way to transmit large files either way over the Internet.
We recommend CuteFTP for the PC and Fetch for the Mac.
INSTRUMENTATION
This is the choice
of which instrument you choose to perform certain musical functions i.e.
which instrument should carry the melody, is the instrument capable of
playing the notes in the desired range, and is it realistic to expect
a trumpet player to mimic some of the complicated lines often played by
the violin?
MIDI
- musical Instrument Digital Interface
MIDI is the protocol
by which synthesizers, samplers and other electronic musical instruments
communicate with each other. Audio itself is not transmitted, just the
data to instruct the Midi device as to which notes to play, on which sound,
how loudly and for how long (in a nutshell!) Midi is a serial interface
capable of transmitting up to 500 notes per second down a standard 5-pin
din cable. Midi devices normally have 3 inputs, a Midi In, Midi Out and
Midi thru (which is a direct copy of what comes into the Midi in to allow
you to daisy chain it to other devices). Each cable can support up to
16 unique Midi channels – the theory being that you would use 1
channel for each device.
MP3
The most popular compression
format for delivery of hi-quality music over the Internet (a medium requiring
small file sizes.) It offers very high compression ratios (10:1 or higher
are quite typical) and very low perceivable loss in quality of sound.
For example, 1 minute of stereo CD quality audio will have a file size
of around 10 MB. The equivalent MP3 file of a similar quality would be
only 1MB in size.
A common misconception
is that it means Mpeg 3. It actually refers to Mpeg 1 (the video format
– precursor of Mpeg2 used in DVDs) layer 3 audio (3rd generation
of Mpeg 1 audio).
(Tech bit) The compression
scheme is based on the algorithm that sounds can be decomposed into additive
sine waves (using the FFT - Fast Fourier Transform). Once the signal is
represented in the frequency domain, various tricks (huffman coding, small
blocks, etc) are used to compress the frequency data and make the file
small.
OFFSET
An offset in programming
terms is simply a number added or subtracted to a fixed number to give
a different result. In game audio terms you could have a random volume
or pitch offset. This is where the programmer attempts to imbue a sense
of real life into a sample by either making it randomly louder or quieter,
or higher or lower in pitch than simply repeating it time and time again
at the same pitch or volume. The important factor is that this will only
happen within a given range so using an offset can be viewed as a neat
way of adding randomness in a controlled manner.
A footstep is a good
example. If you assign a volume and pitch between 0 and 127 for each sample,
set the base value to 64 for each, but add an offset or +/- 10, then each
time you hear a footstep sound, it will playback at a volume or pitch
of between 54 & 74.
OUTBOARD
EFFECTS GEAR
A recording studio
term used to describe the use of different external hardware devices to
alter a sound. Whether that is by adding an effect to the sound (a reverb
or delay) or modifying the existing sound (an EQ, filter or compressor).
PCM
Pulse
Code Modulation.
This is the most common
method for digitisation – the transfer from analog to digital sound.
Each pulse measures a difference in modulation between that and the previous
sample. The modulation here is a difference in amplitude. (See sampling).
Most commonly used in reference to uncompressed digital audio.
PITCH
BEND
Pitch bend is the
gradual and smooth manipulation of pitch over time. A guitarist going
from one note to another has the choice of either simply jumping to the
2nd note (which would be deemed a pitch shift), or gradually bending the
string so that the pitch smoothly moves from the 1st note to the 2nd.
This is pitch bend.
SAMPLING
Sampling is an accurate
word to describe the process of converting audio from an analogue source
into the digital domain. (E.g. recording from vinyl records onto your
computer hard drive). A simple description is that it measures snapshots
of volume over time. A CD player takes 44,100 different snapshots of amplitude
(measure of loudness) and turns each sound level into a number.
(Tech bit) A CD player
works in 16-bits. This gives you 216 different volume levels to work from
(65536 – in computer speak although it’s actually –32767
to +32768). For every minute of recorded stereo sound, a PCM stream will
take up approximately 10 MB of space on your hard drive.
SD2
- Sound Designer II
This is the native
file format of Protools. Protools is the industry standard digital audio
workstation. It runs exclusively on Apple Macs.
(Tech Bit) The format
itself is essentially a RAW file with the description of type and creator
held in the Macintosh resource fork (a small file that provides a MAC
computer with information about the file with which it is associated).
It can be either mono or stereo.
SEPARATION
Separation is the
ability through careful and skilful mixing to hear all of the component
parts of a piece of music. In busy orchestral music it is often possible
to be unable to hear an important accompaniment because another element
of the music is obstructing it.
SOURCE
music
Just to clarify this,
source music does not mean reference music – i.e. music that is
used for simply reference purposes to aid the composer in his task of
composing original music. Source music is the music that plays out during
a scene (whether it be in a film, TV show or game is irrelevant) that
is not written to accompany the scene, it’s just playing in the
background – something on the radio, a band playing in a park etc.
SURROUND
Generic term used
to describe sound that literally surrounds you. This normally involves
having speakers placed around the room at key places so that the sound
can be projected from anywhere in a 360 degree sound field. Analogue based
surround such as Dolby Prologic and more recently Prologic II have made
way for digital formats such as Dolby Digital, DTS & Sony SDDS where
the same limitations as the analogue systems do not apply.
(Tech Bit.) Analogue
systems use clever phase technology to push the sound from a conventional
2 channel stereo cable into 4 channels. Unfortunately on stereo systems
this can make the sound feel slightly muffled, but also you have a reduction
in frequency response in the rear speakers (which are mono). With digital
systems, you do not have those restrictions. So not only do you have 5
discrete full frequency audio channels but also room for a sub.
VIBRATO
Vibrato is a musical
term to describe small but fast pitch variations over time. Take the example
of a vocalist singing a long continuous ‘aaahhh’. Over time,
it is common for vocalists to add a bit of ‘warble’ to the
note to add polish to a performance. That’s vibrato.
VOICINGS
This is how you decide
which note of the chord to give to which instrument playing the chord.
Sounds easy, but your decision can drastically alter the impact and sound
of each chord and they have to be chosen carefully.
WAV
- Windows Wave
The standard file
format for audio on Microsoft Windows based PCs. Just to confuse the issue,
files can have different sample rates or compression (ADPCM, MP3 etc.)
and still retain the same ‘.wav’ extension. The format allows
for mono, stereo or multi-channel audio in a single file (only a few programs
support the latter).
XA
Another compressed
audio format providing audio file size reduction in a ratio of around
3.5:1. It was originally a component of the CD-ROM eXtended Architecture
standard developed by Philips and Microsoft in the late 80s for video
playback.
(Tech Bit) This is
whereby the CD-ROM device would read a file from the disc that is split
into 8 equal sized interleaved streams, 7 of which are devoted to the
video component, and 1 to the audio (a 37.8k 16-bit stereo file compressed
using ADPCM). Various game audio people realised that the potential for
this was far greater if you could use all 8 sectors for audio. This would
maximise the amount of audio that we could get on a single CD. The loss
in frequency response going from 44.1k to 37.8k was negligible so most
felt it was worth it.
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