The Music Software Glossary
This music software glossary contains hard to find definitions of terms
somehow related to making music with computers
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#-Music Software Glossary-#
A surround sound format comprising 5 full
bandwidth channel (left, centre, right, left surround and right
surround) plus a low frequency effects channel.
64 bit doubles
An audio format that uses 8-byte doubles in binary form. 8 bytes per
sample mono, or 16 bytes per sample stereo interleaved. The 64-bit
doubles format has no header. It's purely audio data, just like the raw
8 bit signed
This format is popular for building MOD files, since audio in MOD files
is 8-bit signed. Many MOD editors allow samples to be inserted from or
exported to files in this format. Files with the .sam extension contain
8-bit signed raw data, and by default, they have no headers.
A-Music Software Glossary-A
Adaptive Differential Pulse Code Modulation; an audio compression scheme
which compresses from 16-bit to 4-bit for a 4:1 compression ratio.
Audio Codec '97; A specification for an audio system within the PC that
separates the analog and digital circuits. Formed in 1996 by Intel,
Analog Devices, Creative Labs and others, AC '97 enables the digital
controller chip to be placed on the motherboard and separated from the
"noisy" analog circuits which can be located near the connectors or on a
riser card. The digital controller can support up to four
codecs. AC '97 also provides support for modem
Audio Compression Manager; Microsoft ACM is part of all 32-bit versions
of Windows. The ACM driver enables you to open and save files in a
variety of formats other than those directly supported by a software.
Some of these formats come as a standard part of Windows, while others
are provided by third-parties. You may acquire ACM formats when you
install other software.
Audio Engineering Society; an American organization of audio engineers
which standardizes audio related technology and forms a common forum for
experts in the field.
AES/EBU digital audio bus
A digital sound transmission standard of stereo digital audio and
associated data (called sub-channel data). The standard has been
strongly influenced by CD technology, and is mainly
used between digital studio equipment. The standard specifies multiple
sample rates (32kHz to 48kHz) and sample bit depths (up to 24 bits per
sample). Originally developed by AES, later adopted by
EBU; hence the name.
The AES-31 standard is an open file interchange format, developed by the
Audio Engineering Society as a means of overcoming format
incompatibility issues between different audio hardware- and software.
It can be used for transferring projects via disk or network from one
workstation to another, retaining time positions of events, fades, etc.
AES-31 uses the widely used Microsoft FAT32 file system with Broadcast
Wave as the default audio file format. This means that an AES-31 file
can be transferred to and used with any digital audio workstation that
supports AES-31, regardless of the type of hardware and software used,
as long as the workstation can read the FAT32 file system and Broadcast
Wave files (or regular wave files).
Audio Interchange File Format; originally developed by Apple for storage
of sound in the data fork of Macintosh files. It has been adopted as a
standard audio format by the OMFI (Open Media Format Interchange) group
for cross-platform media exchange, which includes Silicon Graphics, Avid
Technology and others.
Compressed WAV format. A-Law (or CCITT standard G.711) is an audio
compression scheme common in telephony applications. It is a slight
variation of the u-Law compression format, and is found in European
systems. This encoding format compresses original 16-bit audio down to 8
bits (for a 2:1 compression ratio) with a dynamic range of about
13-bits. Thus, a-law encoded waveforms have a higher s/n ratio than
8-bit PCM, but at the price of a bit more distortion than the original
16-bit audio. The quality is higher than you would get with 4-bit ADPCM
formats. Encoding and decoding is rather fast and generally, widely
The Amiga IFF-8SVX format is an 8-bit mono format from the Commodore
The reconstruction of different guitar and bass amplifier models by
software alone. When its well programmed the program reacts exactly like
the real thing when you turn the knobs on the screen. Often you can also
choose from a variety of cabinets or speakers to make the sound even
API stands for Application Programming Interface. That's the way/method
an application can use to access a certain function. A soundcard driver
API is used by audio software to access the hardware. Common APIs for
sound- or audio cards are: MME (also called 'wave' or 'mmsystem'),
DirectSound (sometimes called 'DirectX') or ASIO.
(Audio Stream Input/Output). A multichannel audio transfer protocol
developed by Steinberg in 1997, for audio/MIDI sequencing applications,
allowing access to the multichannel capabilities of sound cards.
The ASIO 2.0 API was released later and added control functions for the
monitoring functions of the hardware. Cubase (and other apps supporting
ASIO 2.0) are able to control the input monitoring of the hardware
This format produces compressed Adobe Audition loop files, which are
essentially .mp3 files with a .cel extension. Each .cel file has a
header that contains loop information, such as the number of beats,
tempo, key, and stretch method.
The .cel format avoids a
potential problem with .mp3 files. During encoding, a very small amount
of silence is added to the beginning, end, or both of an .mp3 file. The
silence is very short--often only a few samples long. When you work with
a loop, though, it's enough to throw off the entire loop.
As it saves a .cel file, Adobe
Audition calculates how much silence will be added to the .mp3 file and
writes this information into the .cel header. Then, when Adobe Audition
opens a .cel file, it reads this information and automatically removes
the silence from the file so that it loops smoothly.
Encoded Audio format used by Sun and NeXT machines. It has many data
types. Like Windows PCM and AIFF, this format can support mono or
stereo, 16- or 8-bit, and a wide range of sample rates when saved as
linear PCM. The NeXT/Sun format is most commonly used for compressing
16-bit data to 8-bit u-law data. AU is used quite extensively on the Web
and in Java applications and applets.
A method to prevent clicks when cutting and pasting audio. The start and
endpoint of the selected audio are automatically set to the nearest
point where the waveform comes from a negative amplitude and crosses
Audio Video Interleave; Also called Video for windows. It is a special
case of the RIFF (Resource Interchange File Format), defined by
Microsoft and is the most common format for audio/video data on the PC.
The application of a defined sequence of different processes to a batch
of files. Every file gets processed the same way and usually stored in
another folder (output folder).
The basic rhythmic unit in a piece of music. When you have a straight
4/4 rhythm a beat means 1/4 note.
Just as the sample rate determines the frequency resolution, the bit
depth determines the amplitude resolution. A bit is a computer term
meaning a single number that can have a value of either zero or one. A
single bit can represent two states, such as on and off. Two bits
together can represent four different states: zero/zero, one/zero,
zero/one, or one/one. Each additional bit doubles the number of states
that can be represented, so a third bit can represent eight states, a
fourth 16, and so on.
Amplitude resolution is just as
important as frequency resolution. Higher bit depth means greater
dynamic range, a lower noise floor, and higher fidelity. When a waveform
is sampled, each sample is assigned the amplitude value closest to the
original analog wave. With a resolution of two bits, each sample can
have one of only four possible amplitude positions. With three-bit
resolution, each sample has eight possible amplitude values. CD-quality
sound is 16-bit, which means that each sample has 65,536 possible
amplitude values. DVD-quality sound is 24-bit, which means that each
sample has 16,777,216 possible amplitude values.
Higher bit depths provide greater dynamic range.
A display that shows how many bits are used, i.e. the resolution of the
audio being monitored. Steinberg's Wavelab provides this feature to get
some valuable information about internal processing and the "actual"
resolution of a sound file.
Brown noise has a spectral frequency of 1/f^2, which means, in layman's
terms, that the noise has much more low-frequency content. Its sounds
are thunder- and waterfall-like. Brown noise is so called because, when
viewed, the wave follows a Brownian motion curve. That is, the next
sample in the waveform is equal to the previous sample, plus a small
random amount. When graphed, this waveform looks like a mountain range.
Markers used in broadcast wav files by the Japanese radio, to mark
significant locations within the file.
BWF - Broadcast Wave Format
It is an extension of the regular
WAV audio format, specified by the
Union in 1997, and updated in 2001. Purpose of this file format is
the seamless exchange of sound data between different computer
platforms. It also specifies the format of metadata, allowing audio
processing elements to identify themselves, document their activities,
and permit synchronization with other recordings. This metadata is
stored as an extension chunk in an otherwise standard digital audio WAV
file. BWF is the audio format used by most file-based non-linear digital
recorders used for motion picture and television production. It contains
Description... Describes the audio
file in up to 256 characters.
Originator... Specifies the name of
the audio files producer in up to 32 characters.
Originator Reference... Specifies
reference information about the producer in up to 32 characters.
Origination Date... Specifies the
date that the subject matter was produced. The date should be in the
year-month-date format (yyyy-mm-dd). For example, specify June 8, 2004
Origination Time... Specifies the
time the audio file was produced. The format is hour:minutes:seconds,
with the hour represented in Universal Military Time (for example,
specify 10 p.m. as 22).
Time Reference... Specifies the
timecode of the audio file, calculated since 12:00 a.m. (midnight).
Select from the hh:mm:ss.ddd or Samples options. This option is commonly
referred to as the clips timestamp. It is used by Audition for spot
inserting in a multitrack session.
Coding History... Provides a text
box for you to describe all coding processes applied to the waveform.
Adobe Audition automatically adds information every time BWF data is
modified and the file is saved. You can manually edit this information.
Write All Metadata... Specifies
whether to write metadata At the Start of the File or At the End of the
File. Metadata written at the start of a Broadcast Wave file works with
most systems, but some expect metadata to come at the end of the file.
For these systems, choose the option to write metadata At the End of the
UMID Specifies... Unique Material
Identifier Data (UMID) according to the SPMTE 330M Standard. This
information is read-only.
C-Music Software Glossary-C
time to create a shimmering effect.
makes a single voice sound like multiple voices in unison. You can
by sending a sound through a series of delays whose
delay times are slowly being modulated.
Comité Consultatif International Téléphonique et Télégraphique; an
organization that sets international communications standards. CCITT,
now known as ITU (the parent organization) has
defined many important standards for data communications
Compact Disc; A compact disc is a polycarbonate with one or more metal
layers capable of storing digital information. The most prevalent types
of compact discs are those used by the music industry to store digital
recordings and CD-ROMs used to store computer
data. Both of these types of compact disc are read-only, which
means that once the data has been recorded onto them, they can only be
read, or played.
CD+G (graphics): a special Audio Compact disc that contains graphics
data in addition to the audio data on the disc. The disc can be played
on a regular Audio CD player, but when played on a special CD+G player,
can output a graphics signal (typically, the CD+G player is hooked up to
a television set). It's main use so far is the realization of Karaoke
CD index markers
An index is a sub-divisions of each track of a CD or DVD-Audio. Each
track may, if necessary, be divided into up to 99 indexes to provide
more than 99 'tracks' per disc. Your burning program lets you set index
markers to specify the exact place of the indexes on your CD or DVD.
Alternative name for Enhanced Music CDs, which are multisession CDs
comprising a CD Audio session (with up to 98 tracks) followed by a
single-track CD-ROM XA session, which contains the data. CD-EXTRA discs
are compatible with all CD audio players (as the data session is not
seen) and the data track can be played in a Windows 95 or 98 PC and/or a
Macintosh depending on how the software was written.
Compact Disc-Musical Instrument Digital
Interface; A CD-system on a computer that enables the user to work with
MIDI instructions for electronic instruments, including reading musical
scores and editing. CD-MIDI can display visual information that
corresponds with the sounds as they are played.
Compact Disc-Read-Only Memory, a type of optical disk capable of storing
large amounts of data -- up to 1GB, although the most common size is
650MB (megabytes). A single CD-ROM has the storage capacity of 700
floppy disks, enough memory to store about 300,000 text pages. CD-ROMs
are stamped by the vendor, and once stamped, they cannot be erased and
filled with new data. To read a CD, you need a CD-ROM player. All
CD-ROMs conform to a standard size and format, so you can load any type
of CD-ROM into any CD-ROM player. In addition, CD-ROM players are
capable of playing audio CDs, which share the same technology.
The data on all CDs is stored in chunks called sectors. There are
330,000 sectors on a CD each of which can hold 2352 bytes. This adds up
to a total data capacity of approximately 744MB (Mega Bytes). Different
CD formats use this capacity in different ways.. There are now a number
of 80, 90 and even 100 minutes CDs available but make sure that your
software is able to write those CDs.
A recent addition to the CD audio
specification allowing disc and track related information to be added to
standard audio CDs for playback on suitably equipped CD audio players.
The CD TEXT information, coded as characters for maximum efficiency, is
contained in the R to W subcode channels in the lead-in and/or program
area of a CD. CD TEXT is compatible with the ITTS
(Interactive Text Transmission System) standard. CD TEXT equipped
players can provide a range of display formats from one or two line, 20
character display to 21 lines of 40 color alphanumeric or graphics
characters. The specification also allows for the future addition of
additional data such as JPEG coded images.
Pronounced sisk, and stands for complex instruction set computer. Most
personal computers, use a CISC architecture, in which the CPU supports
as many as two hundred instructions. An alternative architecture, used
by many workstations and also some personal computers, is RISC (reduced
instruction set computer), which supports fewer instructions.
When circuits or transmission media are driven past the point of their
maximum input amplitude, they tend to limit the signal to its maximum
value. This can happen sharply (digital full scale) or softly (the
sigmoid type limiting action of analog tape) and results in the effect
of hard or soft limiting, respectively. Limiting produces heavy side
banding and, consequently, harsh and nonconsonant distortion. Synonymous
terms are overdrive (especially when speaking of amplifiers) and
saturation (taken from tube amplifier terminology).
Coder/decoder. When talking about data transmission, a coder/decoder is
a device or algorithm which works on a bidirectional data link, coding
transmitted and decoding received data. Audio codecs usually use
computer files, multimedia data streams or TV broadcast channels for
Color "depth" is defined by the number of bits per pixel that can be
displayed on a computer screen. Data is stored in bits. Each bit
represents two colors because it has a value of 0 or 1. The more bits
per pixel, the more colors that can be displayed. Examples of color
depth are shown in the following table:
1 bit color
4 bit color
8 bit color
24 bit color
No. of Colors:
Decreasing the size of stored information by reducing
the representation of the information without significantly diminishing
the information itself, usually by removing redundancies. Requires
decompression upon retrieval. Lossless compression allows the original
data to be recreated exactly. Lossy compression sacrifices some accuracy
to achieve greater compression.
dynamic range by lowering amplitude when an audio signal rises above a
specified threshold. For example, compressors can be used to eliminate
variations in the level of an electric bass, providing an even, solid
bass line. Compressors can also compensate for variations in level
produced by a vocalist who moves frequently or has an erratic volume.
acoustics, an echo is the convolution of the original sound with a
function representing the various objects that are reflecting it.
Convolution reverbs are sample based reverbs that sound more natural but
are usually less flexible than simulated digital reverbs. They are based
on the idea that one can record the way a room reacts to certain
frequencies. These responses are than triggered by the actual sound that
has to get reverberated. Most convolution reverbs even let you record
your own custom library of room responses.
special symbol that indicates where the next character you type on your
screen will appear. You use your mouse or the arrow keys on your
keyboard to move the cursor around on your screen.
Creative Sound Blaster
format is for Sound Blaster and Sound Blaster Pro voice files. Adobe
Audition supports both the older and newer formats. The older format
supports only 8-bit audio, mono to 44.1 kHz and stereo to 22 kHz. The
newer format supports both 8- and 16-bit audio. Files in this format can
contain information for looping and silence. If a file contains loops
and silence blocks, they expand when you open the file.
technique commonly used in editing audio. One sound is faded out while
another fades in, allowing for a smooth transition between the two.
Crossfading is also common in samplers, where it is used to smooth loop
transitions (crossfade looping), and sound design to create hybrid
sounds (one sound morphing or turning into another). While we often
think of this as a digital process, audio engineers have been using two
channel faders on a mixing console to crossfade between two signals or
tracks for many years.
D-Music Software Glossary-D
Digital Audio Tape;
Digital Audio Tape, a magnetic tape
format developed by Sony and Philips in the mid-1980's. DAT uses a
rotary-head format, where the read/write head spins diagonally across
the tape as in a video cassette recorder. Its proper name is "R-DAT",
where "R" for rotary distinguishes it from "S-DAT", an earlier
stationary design. Most computer DAT recorders use DDS (digital data
storage) format, which is the same as audio DAT, but it is not always
possible to read tapes from one system on the other.
stands for "Direct Current." A signal whose midpoint is skewed away from
zero is said to have a DC offset; this can result in clicks, clipping or
other problems. You can use the DC Offset process in sound editors to
negate a pre-existing offset and put the mid of your waveform in the
dynamics processor or plug-in that reduces the high frequencies that
contain excessive hissing or "S" sounds. They either just lower the
volume of the whole sound when those high frequencies appear or a
usually selectable part of the frequency range. In modern recordings
they also got used for another trick. You might have wondered about
these intimate sounds where the singer seems to crawl into your ear and
you understand every bit of what he or she says. This is done by heavily
turning on the high frequencies followed by a hard working de-esser.
The tone heard in a phone when the receiver is picked up, indicating the
line is available for dialing.
Diamond Ware Digitized
This format is used by
DiamondWare Sound Toolkit, a programmer's library that lets you quickly
and easily add high-quality interactive audio to games and multimedia
applications. It supports both mono and stereo files at a variety of
resolutions and sample rates.
DirectX is an API (application programming interface)
for demanding multimedia applications. It provides fairly
direct access to the hardware and handles
tasks like full-color graphics, games, video, and 3-D
animation. Windows had a very
limited multimedia feature set for many years and before
directX many programmers preferred DOS for multimedia purposes.
Now DirectX provides a standard development platform
for Windows-based PCs by enabling software developers to access
specialized hardware features without having to write code that is
Disc at once
If you want to create a CD-R to use as a master for a real CD
production, you must write the CD-R in Disc-At-Once mode. In this mode,
the entire disc is written in one pass, without ever turning off the
recording laser. There are other ways of writing a CD, namely
Track-At-Once and Multisession. If you use these writing
formats, the "link blocks" created to link the
various recording passes together will be recognized as "uncorrectable
errors" when you try to master from the CD-R. These links can also
result in clicks when playing back the CD. Disc-At-Once mode provides
more flexibility when specifying pause lengths between tracks. It is
also the only mode that supports sub-indexes.
Memory Access/Addressing. DMA is a method of transferring data from one
memory area to another without having to go through the central
processing unit. Computers with DMA channels can transfer data to and
from devices much more quickly than those in which the data path goes
through the computer's main processor.
Every frame :00 & :01 are
dropped for each minute change (60 X 2 = 120) except for minutes with
0s (00:, 10:, 20:, 30:, 40: & 50:) (6 X 2 = 12, 120 - 12 = 108)
Color video was slowly introduced into broadcast. It was therefore
necessary to make it compatible with black and white receivers and to
design color receivers or televisions to be able to receive black and
white programming as well. In order to accommodate the extra information
needed for color the b&ws 30 frame/second rate was slowed to 29.97 f/s
for color. Although usually not an issue for non broadcast applications,
in broadcast, the small difference between real time (or the wall clock)
and the time registered on the video can be problematic. Over a period
of 1 hour (SMPTE) the video will be 3.6 seconds or 108 extra frames
longer in relation to the wall clock. To overcome this discrepancy drop
frame is used.
Dual Tone Multi-Frequency signals are used by standard push-button
telephones. They are generated using combinations of 679-, 770-, 852-,
941-, 1209-, 1336-, 1477-, and 1633-Hz sine waves.
DVI is actually both the name of the Digital Video
Interactive hardware system sold by Intel and the file format associated
with that system. DVI technology is essentially a PC-based interactive
audio/video system used for multimedia applications. The DVI system
consists of a board for use in an Intel-based PC, drivers, and
DVI is a major competitor of QuickTime,
AVI, and MPEG for market share in
digital audio/video applications.
Intel's (DVI) and the
International Multimedia Associations (IMA) flavor of
ADPCM compresses 16-bit data to 4 bits/sample (4:1)
by using a different (faster) method than Microsoft ADPCM. It has
different distortion characteristics, which can produce either better or
worse results depending on the sample being compressed. As with
Microsoft ADPCM, use this format with 16-bit rather than 8-bit files.
This compression scheme can be a good alternative to MPEG; it provides
reasonably fast decoding of 4:1 compression, and it degrades sample
quality only slightly.
Digital Versatile Disc (formerly Digital Video Disc);
new type of
CD-ROM that holds a minimum of 4.7GB (gigabytes), enough for a
full-length movie. The DVD specification supports disks with capacities
of from 4.7GB to 17GB and access rates of 600KBps to 1.3 MBps. One of
the best features of DVD drives is that they are backward compatible
with CD-ROMs. This means that DVD players can play old CD-ROMs, CD-I
disks, and video CDs, as well as new DVD-ROMs. Newer DVD players can
also read CD-R disks. DVD uses MPEG-2 to compress video data.
DVD-Audio; About 15 years after the CD new formats are now available
offering higher quality and additional features, but still on the
familiar 12 cm optical disc. One of those is DVD-Audio. It offers at
least 74 minutes of very high quality, surround sound, plus additional
features (such as video and limited interactivity) that are not
available on CDs.
There are three basic types of DVD players:
This could either be an audio-only player (AOP), or a player capable
of displaying visual menus, text and still images.
This is referred to as a "V-Player" (Video Player), and is capable
only of playing back video contents contained in the VIDEO_TS folder.
Universal DVD-Audio/Video player
This is capable of playing back DVD-Audio data, displaying menus, text
and still images. It can also play "hybrid" DVDs with both DVD-A and
video content (contained in a VIDEO_TS folder), as well as standard
Dynamic automation lets you change parameters over time. Unlike static
automation, that just remembers a specific setting that you can recall,
it lets you record and play back dynamic changes. It is realized either
by recording real-time movements of faders and knobs or by drawing lines
that represent a parameter on a screen.
European Broadcasting Union; an organization formed originally by
national radio stations in Europe. Specializes in broadcast audio
distribution technology. Current standardization efforts include
terrestrial digital radio, both for audio and various kinds of data.
Hard disk recording system from Ensoniq, a vendor originally famous for
producing one of the first workable samplers. Ensoniq originally
developed the Paris hardware somewhere between 1995 and 1997 together
with Emu. later Emu released Paris with a new color scheme and called it
"Paris Pro". Unfortunately, the Paris hardware is no longer being
produced. Some software programmers however seem to continue developing
software on a private basis.
You can visit them at
Equalizer; An electronic
circuit, that boost or cuts specific frequencies or frequency bands in a
signal to modify and shape its sound.
This basic idea has been realized in some typical designs like Graphic
Equalizer, Parametric Equalizer, Notch Filter, Wah-Wah pedal etc...
Fast Fourier Transform; An algorithm based on Fourier Theory that music
software uses for filtering, spectral view, and frequency analysis.
Fourier Theory states that any waveform consists of an infinite sum of
sin and cos functions, allowing frequency and amplitude to be quickly
An audio effect caused by mixing a varying, short delay in roughly equal
proportion to the original signal. The name comes from
how the effect was produced on big tape reels whereby the flange of the
reel was tapped to slow down a copy of the signal hence produced phasing
effects in the output.
Internet & computing Flash is a vector-based moving
graphics format created by Macromedia for the publication of animations
on the world-wide web. Flash (.swf) graphics files can be created in
Macromedia's own Flash program, or else in software applications such as
Adobe's LiveMotion or Corel's RAVE (real animated vector effects)
package. Most web browsers still require a plug-in to be installed
before they can play Flash animations.
The term floating point is
derived from the fact that there is no fixed number of digits before or
after the decimal point; that is, the decimal point can float. This
improves the calculating capability of the CPU for arithmetic with real
Synthesis; a music simulation technique that approximates the sounds of
real instruments by modulating and bending raw electronic wave forms.
The first real popular instrument that used this sound generating
technique was the famous Yamaha DX7 keyboard.
A peak in the frequency response of a vocal tract or
musical instrument. Different vowel sounds are characterized by the
position and shape of their formants. The human vocal tract typically
has five formant regions.
A definition of the manner in
which data is stored; its organization. The pattern in which data are
systematically arranged for use on a computer. It is not only used at
computers. For example, VHS, SVHS, and Beta are three different formats
of video tape. They are not 100% compatible with each other, but
information can be transferred from one to the other with the proper
equipment (but not always without loss; SVHS contains more information
than either of the other two).
Frames per Second; The rate of how many complete pictures (frames) a
video or film shows per second to create the illusion of a motion
picture. A video frame consists of two interlaced fields
of either 525 lines (NTSC) or 625 lines (PAL/SECAM), running at 30
frames per second (NTSC) or 25 frames per second (PAL/SECAM). Film runs
at 24 frames per second.
(1) A single screen-sized
image that can be displayed in sequence with other slightly different
images to create motion pictures.
(2) The data on an audio CD
is divided into frames. A frame consists of 588 stereo samples. 75
frames make up one second of audio. Why? Well, 75 x 588 = 44100, and
since the sampling frequency of the CD format is 44100kHz (samples per
second), this equals one second of audio. The data on an audio CD is
divided into frames. A frame consists of 588 stereo samples. 75 frames
make up one second of audio. Why? Well, 75 x 588 = 44100, and since the
sampling frequency of the CD format is 44100kHz (samples per second),
this equals one second of audio.
An oscillator starts generating a waveform at a certain (usually low)
frequency and changes the frequency at a defined speed to another
frequency (usually high), without changing the amplitude of the
waveform. This is useful to determine the frequency response of
circuits, rooms, speakers etc...
Also called snipping, hacking or slicing. An effect that mutes and
un-mutes a signal more or less rhythmically. This effect has first been
realized with noise gates that had a side-chain function. The side chain
has been fed by a rhythmic signal like drums or a pulsating
Part of the CD and DVD disc manufacturing process that uses a laser
(which is modulated by the data to be stored on the disc) to expose
areas of a photo-resist layer on a glass disc, where the final pits are
required. These areas are then developed and the photo resist layer
metallised so that stampers can be grown by electroforming using this
metal layer. The stampers are then used for molding CD and DVD discs.
The GigaSampler-Interface support was invented by Nemesys (now Tascam)
to allow multichannel output for the GigaSampler (and later GigaStudio)
software. As ASIO it allows to access the hardware for multichannel
playback over one device. Recording is not possible. You may argue that
the same functionality is also possible with ASIO and GSIF would not be
needed and yes, probably you are right with that. Nemesys certainly had
reasons not to use the already well-supported ASIO interface invented by
a different audio software vendor. GigaSampler / GigaStudio also work
with DirectSound drivers but that does not allow output of multiple
channels at the same time. Because of that, special GSIF support is
needed for multichannel audiocards.
here are simple technical problems if you want to use different audio
applications at the same time. Now GigaSampler/-Studio originally was
not developed to be used with a Audio-/MIDI-sequencer software at the
same time on the same PC. Still many users demanded that functionality
from the vendors. The solution: a special multiclient driver that allows
the usage of ASIO or MME at the same time as GSIF, usually the physical
output channels need to be assigned to one or the other driver model.
Given a signal, we can decompose it with the Fourier transform. Then, a
harmonic (of some particular frequency present in the transform) is any
frequency (also present in the analysis) which is at a whole number
ratio to our base tone. If the signal is periodic, every partial present
in the analysis is harmonic. The term implies an underlying base tone to
which the harmonic in question is related. (Thus, one doesn't say that
components present in a composite signal are necessarily harmonics, even
though they may appear in integer frequency ratios.)
In file management, a header is a region at the beginning of each file
where bookkeeping information is kept. The file header may contain the
date the file was created, the date it was last updated, and the file's
size. The header can be accessed only by the operating system or by
Normally when creating a CD or a DVD-A, only the sections between CD/DVD
track markers are burned, and the pauses between tracks are replaced by
silence. Some programs however let you burn the exact image of the Audio
Montage to CD or DVD, including any audio between tracks. This makes it
possible to add audio (without track markers) either in between CD/DVD
tracks, before the first track or after the last track, thereby creating
"hidden" tracks or continuous playback between tracks.
High definition audio
Intel® High Definition Audio (HD Audio) is
the specification for the next generation integrated audio solution.
First announced at the Intel Developer Forum, Spring 2003, and gaining
rapid adoption by the PC audio community, it is intended to be a
complete audio specification from the jacks in the PC to the driver
layer of the OS. The HD Audio is featured with Intels upcoming
chipset codenamed Grantsdale and will be the evolutionary replacement
of AC 97.
The following table
shows a quick overview of the benefits of the Intel HD Audio
Specification over the current AC 97 Audio Specification.
High Definition Audio
11.5 MB/s max bandwidth
24 -48 MB/s bandwith
Bandwidth for more
channels and microphone array inputs at higher sample rates
assignment, fixed slot base protocol
Bandwidth delivered where
multi-streaming / multiple similar device types
Single stream support (in
Support for multiple
streams (in and out)
Support for new Digital
Home / Digital Office usage models
Clock provided by primary
Clock provided by the
Intel® I/O Controller Hub (ICH)
stable clock source for synchronization
Stability depending on
Unique Microsoft bus
Single bus driver for
more OS stability and base functionality
Limited device sensing /
Full device sensing /
Support for full audio
Plug and Play
2-element (stereo) array
More accurate, better
quality voice input and recognition
Head related transfer function; the transfer function of the system
resulting from the linear filtering action of placing the human body
(especially the head) in a sound field. The main components arise from
shoulder and ear lobe reflections and from diffraction effects on sound
traveling around the head. Also used to denote the impulse response of
such a system or any processing method simulating such a system (the
usage is quite fuzzy indeed).
I-Music Software Glossary-I
International Federation Of Producers Of Phonograms
IFPI represents the recording industry worldwide and
affiliated industry associations.
IFPI's international Secretariat is based in London and is linked to
regional offices in Brussels, Hong Kong, Miami and Moscow. According to
their own statements these are their priorities.
- Fighting music piracy
- Promoting fair market access and adequate
- Helping develop the legal conditions and the
technologies for the recording industry to prosper in the digital era
- Promoting the value of music in the development
of economies, as well as in social and cultural life
(1) International MIDI association; a consortium formed as a place for
users of MIDI and related software to discuss their problems and
propositions. IMA keeps close contacts with MMA to relay user input and
suggestions to manufacturers.
(2) International Multimedia Association; a professional trade
association of companies, institutions, and individuals involved in
producing and using interactive multimedia technology.
(1) A duplicate, copy, or
representation of all or part of a hard or floppy disk, a section of
memory or hard drive, a file, a program, or data. For example, a RAM
disk can hold an image of all or part of a disk in main memory; a
virtual RAM program can create an image of some portion of the
computers main memory on disk.
(2) A exact copy of
a CD as a
file on a computer. CD-images represent the finished CD. If you compare
your CD after burning with the image (verify), you can make sure that it
is perfectly written.
International Standard Recording Code; defined by ISO 3901, is an
international standard code for uniquely identifying sound recordings
and music video recordings. IFPI has been appointed by ISO as
registration authority for this standard. In addition to the basic PQ
codes, professional CD burning software allows you to specify an ISRC
code for each audio track.
The ISRC code is structured as
Country Code (2 ASCII characters).
Owner Code (3 ASCII characters or
Recording Year (2 digits or ASCII
Serial Number (5 digits or ASCII
The Red Book CD digital audio
standard enables the encoding of ISRCs onto CDs.
Interactive text transmission system; It provides the
mechanism for encoding sound associated data on prerecorded media and
for the transport of such data across equipment interfaces. Defines
those system characteristics which are independent of the recording or
International Telecommunication Union; an intergovernmental organization
through which public and private organizations develop
telecommunications. The ITU was founded in 1865 and became a United
Nations agency in 1947. It is responsible for adopting international
treaties, regulations and standards governing telecommunications. The
standardization functions were formerly performed by a group within the
ITU called CCITT, but after a 1992 reorganization
the CCITT no longer exists as a separate entity.
Intervoice, Inc. a company from Dallas, creates voice automation
solutions. Voice server software, text to speech, advanced speech
recognition and intelligent networks.
A high-level programming language developed by Sun Microsystems. Java
was originally called OAK, and was designed for handheld devices and
set-top boxes. Oak was unsuccessful so in 1995 Sun changed the name to
Java and modified the language to take advantage of the World Wide Web.
JKL keyboard commands. The keys J, K and L are used for transport
functions in Audio/Video workstations. It developed as a quasi standard.
A round dial found mainly on some VCRs
and DVD players which allows the easy movement forward or backward
within a software program (usually a movie). The jog/shuttle dial is an
intuitive and simple means of moving forward or backward at multiple
fast or slow motion speeds. In order to go forward, one simply turns the
dial right. In order to go backward, the dial is turned left. The
jog/shuttle dial is easy to use and easy to find on a remote control or
component and is also easy to use simply by touch in a dark room. Such
dials may also be found on other components to tune a radio station for
instance, but they are used primarily for video with VCRs, laserdisc
players and DVD players.
Latch Automation Mode
Latch Records adjustments you make to automation settings and creates
corresponding edit points on track envelopes. It begins recording, when
you first adjust a setting, and continues to record new settings until
In general the time lag
between a request and the action being performed. Latency can reflect
network time delays or MIDI and audio playback timing errors. Regarding
audio interfaces like soundcards it's the
delays occurring when you're recording or playing back. This delay is
known as soundcard latency. It depends on the processing power of the
soundcard, the drivers and the speed of the host computer.
A leveler reduces signal level differences in audio material. It can be
used to process recordings where the level unintentionally varies. It
will boost low levels and attenuate high level audio signals but on a
much more general basis than a compressor or even a limiter.
A limiter is designed to ensure that the output level never exceeds a
certain set output level, to avoid clipping in following devices.
Conventional limiters usually require very accurate setting up of the
attack and release parameters, to totally avoid the possibility of the
output level going beyond the set threshold level. Software limiters
often adjust and optimize these parameters automatically, according to
the audio material.
A specialized effect designed to
boost signals as well as increase the perceived level of the audio
signal. It uses the headroom on a CD to achieve the maximum perceived
loudness of a song or advertising clips.
A Symbol like a colored vertical line that identifies a specific
position in a sound or video file. You can save it with the sound file.
They can be used to mark looping points, for designating regions in a
file or just to remember points of interest like glitches, song start
and end or rough descriptions of the content. You normally can label a
marker and search for it later.
MADI or AES10
Multichannel Audio Digital Interface; a unidirectional multichannel
digital audio transmission standard originated by the AES. MADI is based
on the FDDI (Fibre Distributed Data Interface) transmission format, but
usually uses coaxial cable instead of optical fibre. Accommodates up to
56 channels and 24 bits per sample. Used for point-to-point multichannel
digital audio connections in studio and broadcasting environments.
Term for a bar of music, commonly used only in the U.S.
A musical notation for a repeating pattern of musical beats.
Metadata command markers indicate when an instruction (function) will
occur in a streaming media file. You can use command markers to display
headlines, captions, link to Web sites, or any other function you
Multi frequency or CCITT R1 signals are used internally by the telephone
networks. They are generated by a combination of 700-, 900-, 1100-,
1300-, 1500-, and 1700-Hz sine waves. It is usually used on trunk
circuits or pay telephones.
The Microsoft ADPCM format provides 4:1
compression. Files saved in this format expand automatically to 16-bits
when opened, regardless of their original resolution. For this reason,
use this format with 16-bit rather than 8-bit files.
Microsoft .NET Framework
The .NET Framework is
an integral Windows component for developing, building and running
software applications and Web services. It
should make the development of platform independent applications easier
and more effective.
Microsoft Windows Media®
The digital formats of audio and video files featured by Microsoft's
Musical Instrument Digital
Interface. A standard protocol for communication between electronic
musical instruments and computers. It transmits
information of which notes should be played how loud and other control
information like volume, pan etc. Normally it doesn't transmit any audio
data except sending short sound samples via Midi SDS.
A timing reference signal that can be sent within the
MIDI data stream used to synchronize pieces of equipment together. MIDI
clock runs at a rate of 24 pulses per quarter note. This means that the
actual speed of the MIDI clock varies with the tempo of the clock
generator (not like
MIDI time code, which runs at a constant
rate). MIDI clock itself does not carry any location information - the
receiving device does not know what measure or beat it should be playing
at any given time, just how fast it should be going. To transmit that
kind of information you need the MIDI song position pointer.
Sample Dump Standard; standardized by the Midi Manufacturers
Association, this protocol allows unified downloading of sample data to
synthesizers and samplers through the MIDI bus. Utilizes SysEx messages
and offers two separate modes: open loop and closed loop. Open loop
corresponds to the usual MIDI connection topology, in closed loop
configuration a separate return cable is used to provide feedback. SDS
is extraordinarily slow, even in the context of the MIDI physical layer.
In addition, operating SDS reliably is quite difficult (to use SDS in
closed loop mode, the physical cabling has to be changed, for instance)
and so the standard is not currently widely deployed in studio
A MIDI message that tells the slave device where to start playback from.
It is sent when both devices are stopped, not while they play. It just
synchronizes the starting point of the playback.
MIDI Time Code; is used to synchronize two or more devices. MTC messages
are an alternative to using MIDI Clock and
Song Position Pointer
messages. MTC is essentially digital SMPTE mutated
for transmission over MIDI. SMPTE timing is referenced from an absolute
"time of day". On the other hand, MIDI Clocks and Song Position Pointer
are based upon musical beats from the start of a song, played at a
specific Tempo. For many (non-musical) cues, it's easier for humans to
reference time in some absolute way rather than based upon musical beats
at a certain tempo.
Mixed mode CD
A special type of CD which contains different types
of information in separate tracks on the CD. The most common form of
mixed mode CD places computer data in track 1 and audio information in
the subsequent tracks.
Files with the extension .mod belong to tracker software which, in their
purest form, allow the user to arrange sound samples stepwise on a
timeline across several monophonic channels. A tracker song, when saved
to disk, typically incorporates all the sequencing data plus samples,
and thus during the format's heyday it became almost a sport to create
long, complex .mod (or .sng) files which were nonetheless
smaller than 880 kB.
Midi Manufacturers' Association; a consortium formed to promote and
refine the MIDI specification and to guide in the implementation of the
standard. MMA extensions to the original MIDI specification include MIDI
time signaling and SDS.
(Multi Media Extension) The MME devices (or drivers) have been invented
by Microsoft with the not very well known operating system called
"Windows with Multimedia Extensions 1.0" that was based on Windows 3.0
and was the base for later released Windows 3.1 and 3.11. The name of
the OS ("Multimedia Extensions") was also used for the API to access
soundcards. Windows MME 1.0 was really the first Microsoft operating
system that provided a universal API that worked with any soundcard
hardware exactly the same way (if there was a driver). The same API with
small modifications is used until today to playback and record wave
The variation of some characteristic of a signal or a parameter of an
algorithm producing the signal to achieve some specific goal. Examples
include amplitude modulation (time-variant scaling of a signal (AM,
tremolo)) and frequency modulation (variation of the repetition rate of
a (quasi-) periodic signal (FM, vibrato)).
MPEG1 or 2, layer 3 audio coding; a lossy, perceptual audio coding
format widely used for the transmission of stereophonic sound, both in
commercial and non-commercial environments. Layer 3 is the most
sophisticated of the 3 layers specified for MPEG1 and MPEG2 (They share
the same audio bitstream formats, only the allowed bitrates differ.
Funny enough, MPEG2 allows only three of the lower bitrates.). The
standard does not specify the codec, per se, only the bitstream.
However, implementation issues have stabilized fairly well by now. MP3
offers excellent audio quality for music and similar sound encountered
on soundtracks at relatively low bit-rates (in the range from 48kbps to
196kbps). Isn't suitable for very low bitrate speech coding, for which
different methods exist. The acronym comes from the common filename
extension used for files of this content. (FYI: MPEG1 layer 1 audio
coding is used in DCC under a different name.)
Motion Picture Experts Group; a joint consortium of motion picture
engineers. Standardizes movie related material. Commonly known for its
MPEG1, MPEG2 and MPEG4 standards, which pertain to the digital coding
and transmission of moving picture and associated sound (MPEG1-2), and
multimedia (MPEG4, in draft stage).
The capability of a computer with a single CPU to
simulate the processing of more than one task at a time. Multitasking is
effective when one (or more) of the applications spends most of its time
in an idle state, waiting for a user-initiated event such as a keystroke
or mouse click. There are two basic types of multitasking, preemptive
and cooperative. In preemptive multitasking, the operating system
parcels out CPU time slices to each program. In cooperative
multitasking, each program can control the CPU for as long as it needs
it. If a program is not using the CPU, however, it can allow another
program to use it temporarily. OS/2, Windows 95, Windows NT, the Amiga
operating system and UNIX use preemptive multitasking, whereas Microsoft
Windows 3.x and the MultiFinder (for Macintosh computers) use
Multiband compressors have been developed to overcome a problem that
came up when compressing mixes or complex sounds. A full band compressor
always compresses the whole signal and doesn't differentiate between
frequencies. If you tried to compress the bass drum of a complete drum
mix, also the hi-hat and the cymbals were compressed which contain
mainly high frequencies. So you had "pumping" cymbals. The multiband
compressor avoids this problem by dividing the frequency spectrum in
different bands, that get compressed separately. This leads to a very
present and tight sound when applied correctly and to maximum perceived
N-Music Software Glossary-N
When Steve Jobs left Apple, he decided to create the best computer
possible ! The result is the NeXT. This prodigious computer impressed a
lot of people when it was presented! Its technical features, its object
oriented operating system and its graphical interface, even its black
case were very far from the standards (remember how many black-cased
computers there were in 1988: not many)! And NeXTStep is always
considered as a reference. It was sold with a lot of great programs and
a very powerful 400 dpi laser printer. Some technical features were a
bit strange (grayscale display, no floppy drive, no hard disk), but were
modified in the next generation with the NeXT Station and the NeXT Cube
Gating, or noise gating, is a method of dynamic processing that mutes
audio signals below a certain set threshold level. As soon as the signal
level exceeds the set threshold, the gate opens to let the signal
through. Some gates offer additional features, such as auto calibration
of the threshold setting, a side chain input,
a look-ahead predict function, and frequency selective triggering.
There are basically two types of noise reductions in
use. Single and double ended systems. Single-ended noise reduction
systems use dynamic filters to reduce the level of audible noise in a
signal. Double-ended noise reductions process (compress) the signal
during recording and then 'unprocess' (expand) it on playback.
To boost the level of a waveform to its maximum
amount without clipping, limiting, compressing or changing it in any
other way. This maximizes resolution and minimizes certain types of
noise like aliasing.
filter that eliminates a narrow
frequency range from a
signal without affecting the rest of the sound.
O-Music Software Glossary-O
Ogg Vorbis is an open source audio codec designed to
compete with MP3. Since it is not licensed like MP3, software using this
codec does not have to pay royalties.
The software on your computer that controls the basic
operation of the machine. The operating system performs such tasks as
recognizing keyboard input, sending output to the monitor, keeping track
of files and directories on the disk, and controlling other connected
devices such as disk drives and printers.
Original Sound Quality. This is Wavelabs proprietary lossless compressed
audio format. It can significantly reduce the audio file size without
affecting the audio quality at all.
P-Music Software Glossary-P
Pulse-code modulation (PCM) is a modulation technique. It is a digital
representation of an analog signal where the magnitude of the signal is
sampled regularly at uniform intervals. Every sample is quantized to a
series of symbols in a digital code, which is usually a binary code. PCM
is used in digital telephone systems. It is also the standard form for
digital audio in computers and various compact disc formats.
Digital Assistant; a small hand-held computer that in the most basic
form, allows you to store names and addresses, prepare to-do lists,
schedule appointments, keep track of projects, track expenditures, take
notes, and do calculations. Depending on the model, you also may be able
to send or receive e-mail; do word processing; play MP3 music files; get
news, entertainment and stock quotes from the Internet; play video
games; and have an integrated digital camera or GPS receiver.
A meter that shows the relation of the left and right channel. If you
have a mono signal you get a straight line in the middle. A signal on
the right channel gets a 45° line to the right, a signal on the left
channel gets a 45° line to the left side. A real stereo signal gets a
waveform like the one below.
to flanging, phasing introduces a variable phase-shift to a split signal
and recombines it, creating psychedelic effects first popularized by
guitarists of the 1960s. The Sweeping Phaser effect sweeps a notch- or
boost-type filter back and forth about a center frequency. A phase is
similar to a flange except that instead of using a simple delay,
frequencies are phase-shifted over time. If a phase is used on stereo
files, the stereo image can be dramatically altered to create some
remarkably interesting sounds.
turns the signal "upside down", which is the same as inverting the phase
by 180°. No settings are needed for the operation. There is no audible
change when you invert the phase of a mono signal. However, if one
channel in a stereo pair is out of phase with the other, this will lead
to artifacts such as a drop in the bass register and a "blurred" stereo
image. The most common use for this function is therefore to fix a
stereo recording where one of the channels has accidentally been
recorded out of phase with the other. Another Application is to mix two
signals with inverted phase to listen to the difference of the signals.
This is useful to find out the effect that has been applied to a sound
file or to determine if sounds are identical.
Physical modeling tries to emulate parts of a real instrument by
algorithms. It's an extremely good choice for synthesis of many
classical instruments, especially those of the wind and brass families.
Its parameters directly reflect the ones of the real instrument and
excellent emulations can be produced. Original synthesis is fairly easy
on PM platforms. Serious processing power is needed, something that
limits the polyphony of current PM implementations. In addition,
instrument design can be very time consuming. Some types of instruments
are more difficult to model, as well, especially instruments with
significant two plus dimensional effects. These include e.g. drums and
plates, and, to some extent, string instruments. Sometimes these
problems can be solved by using modeling alongside other synthesis
methods or expanding our models to include samples as excitation or by
allowing traditional sound processing methods (effects, filtering etc.)
to be applied within our instrument. Sometimes not. Progress is fast,
techniques are developing constantly and the field will certainly get
even more attention as time goes by and serious commercial applications
continue to appear.
Pink noise has a spectral frequency of 1/f and is found mostly in
nature. It is the most natural sounding of the noises. By equalizing the
sounds, you can generate rainfall, waterfalls, wind, rushing river, and
other natural sounds. Pink noise is exactly between brown and white
noise (hence, some people used to call it tan noise). It is neither
random nor predictable; it is fractal-like when viewed. When zoomed in,
the pattern looks identical to when zoomed out, except at a lower
This effect varies the pitch of the source audio over time. You can
sometimes use a graph to "draw" a tempo to create smooth tempo changes
or other effects, such as that of a record or a tape speeding up or
Most Pitch Correction effects provide two ways to make pitch adjustments
for monophonic instruments or voices. Automatic mode analyzes the audio
content and automatically corrects the pitch based on the key you
define, without your having to analyze each note. Manual mode creates a
pitch profile that you can adjust note-by-note. You can even
over-correct vocals to create robotic-sounding effects. The Pitch
Correction effect detects the pitch of the source audio and measures the
periodic cycle of the waveform to determine its pitch. The effect can be
used on audio that contains a periodic signal (that is, audio with one
note at a time, such as for a saxophone, violin, or vocals). Nonperiodic
audio, or periodic audio with a high noise floor, can disrupt the
effect's ability to detect the incoming pitch, resulting in incomplete
A pitch shifter lets you transpose a sound with or without changing it's
length. The earlier versions like the famous Harmonizer from Eventide
didn't let you preserve the formants of the sound
and if you changed the pitch more than 2 or 3 halftones upwards it
sounded like Mickey Mouse. Modern pitch shifters let you preserve the
formants in a sound to avoid that effect.
Peak program meters; Unlike VU-meters, PPM meters can show you even the
fastest peaks in an audio signal. At least in the digital domain. Analog
PPM-meters still have some (very short) reaction time. The integration
time is the time how fast the PPM bar moves. If it is too short the bar
is flickering at some signals and if it is too high the PPM-bar reacts
too slowly. Here are some widely used integration times for PPM-meters.
PQ refers to the first two (of the eight, named from P to W) subchannel
bits on CDs. These are used to carry auxiliary data, such as track
information, the table of contents (TOC), catalog numbers, ISRC
(International Standard Recording Code) information, de-emphasis status,
SCMS copy propagation control and so on. The majority of this
information is carried over the Q channel, accumulated in 98 bit frames,
whereas the P channel carries a simplistic code denoting the starts and
ends of CD tracks, lead-in and lead-out areas. Most current CD players
are sophisticated enough not to use the P channel code at all, since all
relevant information is also available through the more sophisticated Q
coding scheme. The addition of PQ code is a major portion of the CD
mastering process - often manufacturing plant bound masters are simply
referred to as being PQ coded or PQed.
Pre-emphasis is a basic noise reduction process that is implemented by a
CD player. For pre-emphasis to occur, the CD player must support the
Even if you don't push record button, the input signal gets recorded
into a memory if your software supports that. This memory is called the
prerecord buffer. When you start recording the sound recorded in the
prerecord buffer just gets appended to the beginning of your recording.
Therefore the program has recorded even before you decided to push the
record button. This is especially useful when you want to make very
sure, that you don't cut off the beginning by pressing the record button
The term comes originally from analog tape machines. It means
action of switching a tape machine to record while it plays back and is
usually followed by a punch out. With most multitrack machines, both
punching in and out can be accomplished without stopping tape. Most new
machines have auto punch in/out where you set points in advance, and let
the machine do the work. Software basically just imitates that function
and lets you set the punch-in/out points by markers or on the fly.
Q-Music Software Glossary-Q
Since the introduction of stereo equipment in the 1950's, scientists and
listeners alike have set their sights on finding an even better audio
experience - true 3D audio. Throughout the sixties and seventies numbers
were crunched, algorithms designed and sound tests developed. But few
could create an effective 3D audio experience. Then in the early 1980's,
QSound's inventors were setting up a complex microphone arrangement when
they discovered that sound was coming from a location it wasn't supposed
to be coming from. They had inadvertently created 3D audio! Intrigued,
they tried to replicate the event in a reliable manner.
later, after thousands of hours of consultation with neurological and
medical specialists and half a million human listening tests, they
produced what some skeptics said was impossible: 3D positional audio.
They called the technology "QSound®" and it has become synonymous with a
rich and realistic sound experience.
A video and animation system developed by Apple Computer. QuickTime is
built into the Macintosh operating system and is used by most Mac
applications that include video or animation. PCs can also run files in
QuickTime format, but they require a special QuickTime driver. QuickTime
supports most encoding formats, including Cinepak, JPEG, and MPEG.
QuickTime is competing with a number of other standards, including AVI
R-Music Software Glossary-R
Access Memory; the place in a computer where the operating system,
application programs, and data in current use are kept temporarily so
that they can be quickly reached by the computer's processor.
This format is simply the PCM dump of all data for the wave. No header
information is contained in the file. For this reason, you must select
the sample rate, resolution, and number of channels upon opening the
file. By opening audio data as PCM, you can interpret almost any audio
file format--but make sure that you have some idea about the sample
rate, number of channels, and so on. When you guess at these parameters
upon opening a file, it may sound incorrect (depending on which
parameters are wrong). Once the file is opened and sounds fine, you may
hear clicks at the start or end of the waveform, or sometimes
throughout. These clicks are various header information being
interpreted as waveform material. Just cut these out, and you've read in
a wave in an unknown format.
format that allows a user to hear streaming audio files in real time
over the Internet as opposed to waiting for an audio file to download
before hearing it.
format that allows a user to watch streaming video files in real time
over the Internet as opposed to waiting for a video file to download
before watching it.
Red Book Standard was developed to define specifications for producing
audio CDs, and is the first of the book standards. The Red Book Standard
contains specifications on size of the media, maximum recordable area,
tracking information, etc. All subsequent books (Orange Book
multisession specifications for CD-R/RW, Yellow Book for data, White
book for CD-Interactive, etc.) are based on the physical specifications
contained in the Red Book.
Computing the final file in a sound editor or graphics program
(especially 3D program).
ReWire was designed as the software equivalent of a
multi-channel cable between two audio applications. The original thought
was to provide a link between ReBirth and other audio programs. But the
technology quickly won a foothold and has now been implemented in a lot
of other applications. ReWire 2.0 now takes the next logical step by
integrating MIDI, audio and transport control in one invisible
Resource Interchange File Format; Multimedia applications require the
storage and management of a wide variety of data, including bitmaps,
audio data, video data, and peripheral device control information. RIFF
provides an excellent way to store all these varied types of data. The
type of data a RIFF file contains is indicated by the file extension.
Examples of data that may be stored in RIFF files are:
Audio/visual interleaved data (.AVI)
of other RIFF files (.BND)
NOTE: At this point, AVI files are the only type of
RIFF files that have been fully implemented using the current RIFF
Pronounced risk, acronym for reduced instruction set computer, a type of
microprocessor that recognizes a relatively limited number of
instructions. Until the mid-1980s, the tendency among computer
manufacturers was to build increasingly complex CPUs that had
ever-larger sets of instructions. At that time, however, a number of
computer manufacturers decided to reverse this trend by building CPUs
capable of executing only a very limited set of instructions. One
advantage of reduced instruction set computers is that they can execute
their instructions very fast because the instructions are so simple.
Another, perhaps more important advantage, is that RISC chips require
fewer transistors, which makes them cheaper to design and produce. Since
the emergence of RISC computers, conventional computers have been
referred to as CISCs (complex instruction set computers).
Rubber band curves
Also called envelopes. Curves used to automate volume, pan, effect
parameters, tempo, etc... You can create dragging points and move them
or shift the whole graph up and down.
Rubber Band Curves
S-Music Software Glossary-S
S/PDIF or IEC-958
Sony/Philips Digital Interface; a consumer derivative of the AES/EBU
bus. Standardized by the International Electrotechnical Commission under
the name IEC-958, but marketed as S/PDIF for consumer applications.
(Technically, these are two different standards but in practice, they
are almost identical. They interoperate perfectly.) Uses simplified
AES/EBU (consumer mode) and includes provisions for copy management
through SCMS. Used primarily for digital audio transmission in consumer
applications, such as CD players, DATs, Minidisk players, and DCC
recorders. Applied on top of both electrical and optical interfaces.
Super Audio CD; It offers high quality audio and optional surround sound
but no images, video or interactivity. SA-CD discs can be hybrid and
include a CD audio layer which will play on normal CD players, albeit at
CD quality. SA-CD discs will not play on DVD-Video players unless they
are designed to play SA-CD.
digital recording of a sound, often a single note, or perhaps several
bars of a song, the length only restricted by the memory of the system.
Once the sound is "sampled" or digitized, it can be manipulated. The
sample can be trimmed, looped, pitch-shifted, reversed, slowed-down or
speeded-up, and altered in a myriad of ways. The most common
applications of sampling are recording a single note of an instrument,
then playing that sample back from a keyboard to simulate the original
instrument, or recording a few bars of a rhythm and using the sampler to
repeat that rhythm in a loop. But the sampler also allows the creation
of nonexistent sounds with some characteristics of natural organic
sounds, and the playing of natural sounds in ways impossible with the
The sampling rate determines the frequency range of an audio
file. The higher the sampling rate, the closer the shape of the digital
waveform will be to that of the original analog waveform. Low sampling
rates limit the range of frequencies that can be recorded, which can
result in a recording that poorly represents the original sound.
Two sample rates A. Low sample rate that
distorts the original sound wave. B. High
sample rate that perfectly reproduces the original sound wave.
To reproduce a given frequency,
the sampling rate must be at least twice that frequency. For example, if
the audio contains audible frequencies as high as 8000 Hz, you need a
sample rate of 16,000 samples per second to represent this audio
accurately in digital form. CDs have a sample rate of 44,100 samples per
second that allows sampling up to 22,050 Hz, which is higher than the
limit of human hearing, 20,000 Hz.
The SampleVision format is native to Turtle Beach's SampleVision
program. This format supports only mono, 16-bit audio. it supports loop
points (no looping, forward loop, forward/back loop, number of times to
Serial Copy Management System; a protocol used for restricting digital
copying of audio material in consumer applications. Based on sub-channel
coding of generation identifiers and copy protection bits on digital
audio media, such as DATs and CDs. Only implemented in consumer mode
applications, pro mode applications ignore SCMS. AES/EBU in pro mode
cannot even convey SCMS information. Three possible conditions are
defined by the SCMS flags contained in the Q-channel:
Single generation copy
No digital copying
The SCMS flags are output from CD players via the
S/PDIF (Sony/Philips Digital Interface) which is used to connect to a
CD-recorder or other recording hardware. CD-recorders should obey the
SCMS flags, inhibiting copying from a second generation copy or where no
copying is allowed. SCMS has no affect on analogue copying..
The frequency spectrum is sometimes divided into bands (usually 10, 15
or 30 bands). The frequency scale makes more sense when it is
logarithmic because if it is linear you have much more information about
the high frequencies than about the lows. You get valuable information
about the overall sound of instruments and especially the end-mix if you
can read this display well.
Another term for macro or batch file, a script is a list
of commands that can be executed without user interaction. A script
language is a simple programming language with which you can write
Ability to move forward/reverse in the audio while
listening to the audio. Similar to cueing during wind/rewind on analogue
Small Computer System Interface. Bidirectional, parallel
interface to connect up to 7 (15 with wide SCSI) external devices to a
Scott Studios Wave
Scott Studios is a large
digital air studio systems vendor. If
you're saving files for use with a Scott Studios system, you can add
different commands and information like Artist, title, unique ID, date
of recording, vocal begin etc...
In addition to the normal signal input, a noise gate, compressor or
limiter can have a side chain input. In normal use, the amount of
compression or expansion is related to the dynamics of the input signal.
The side chain allows to control the signal passing through the unit by
the dynamics of another, completely separate signal that you fed in by
the side chain input.
SCSI Musical Data Interchange; a data interchange standard originated in
1991 by Peavey Electronics. In the late 80's and early 90's, samplers
were coming into fashion and a standardized way to exchange sample data
was needed. As MIDI was quite old and extremely slow (MIDI choke was a
problem even then), it was seen that a new bus was needed. As the SCSI
(Small Computers System Interface) bus already existed and had proven to
be interoperable, SMDI leveraged the existing technology. Nowadays SMDI
can be used to convey all kinds of information besides pure sample data
and is invaluable whenever samplers need to be integrated to the rest of
the studio. As an added bonus, computer connectivity and use of existing
SCSI hard drives became possible.
Society for Motion Picture and Television Engineers; an organization of
motion picture and television technology experts that standardizes
technical aspects of moving picture and related data (such as audio)
transmission and coding, such as frame rates, time codes and modulation
techniques. Responsible for the time code format of the same name which
is commonly used in broadcasting, film production and professional audio
applications as a common synchronization standard to relate pieces of
audiovisual presentations together.
Sound Designer 1
This was the original file format developed by digidesign for their
programs Sound Designer and Pro-Tools on the apple platform. it was only
Sound Designer 2
This audio file format is used by Digidesign applications (such as Pro
Tools). 8, 16 or 24 bit resolutions supported. It can be stereophonic
too. The SDII file has become a widely accepted standard for
transferring audio files between editing applications. Most Mac CD-ROM
writer software, for example, specifies SDII or
Audio Interchange File Format as the file format needed
when making audio CDs.
Short for Scalable Processor Architecture, a RISC technology developed
by Sun Microsystems. The term SPARC® itself is a trademark of SPARC
International, an independent organization that licenses the term to Sun
for its use. Sun's workstations based on the SPARC include the
SPARCstation, SPARCserver, Ultra1, Ultra2 and SPARCcluster.
An instrument or software
which displays the frequency spectrum of a sound signal
separate low speed data channel on every CD. The subcode comprises 8
channels. The P and Q channels are used to provide control information
for CD discs. The R to W channels are used for CD Graphics.
Includes SCMS data, as well as the additional data
oriented applications standardized as CD+G and
CD+MIDI. Later, the same coding was transferred
to AES/EBU frames and DAT tape.
Digital Perfect Clarity
This technology allows users to compress music in a format that will not
sacrifice the fidelity of the original source audio recording. While
most audio compression technologies such as MP3 and
WMA are considered "lossy," Perfect Clarity Audio delivers audio output
that is identical to the original source and supports both 16- and
24-bit audio. During the editing process, Perfect Clarity Audio files
can be modified and recompressed without degradation that lossy codecs
add to every generation. Test files have shown compression ratios of 2:1
and as high as 5:1 with no loss in audio quality.
Sony Pictures Digital Wave
The WAVE-64 file format is defined as a true 64 bit file format that
allows to overcome the limitations of the RIFF/WAVE
format. The RIFF/WAVE file format
as defined by Microsoft allows to store up to 4 GB of audio data in a
single file. This is sufficient to hold about 6h 45min of uncompressed
PCM coded stereo 16-bit audio signals with a sample rate of 44.1 kHz.
However, for multichannel audio (e.g. 5.1 surround), high-definition
formats (24 bits, 96 or 192 kHz sample rate) or some special
applications in production and broadcasting, the file size limit of 4 GB
is rather inconvenient, since long recordings need to be split into
The file format was originally defined by Sonic
Foundry. In Summer 2003, Sony Pictures Digital acquired Sonic Foundry's
Desktop Software assets. Since then, the new format is officially
promoted as Sony Pictures Digital Wave 64(TM). Companies are
encouraged to support this format and no royalties have to be paid to
Sun Microsystems is a company based in Mountain View, California that
builds computer hardware and software. Sun Microsystems was founded in
1982 by Andreas Bechtolsheim, Vinod Khosla, and Scott McNeally. The firm
is best known for developing workstations and operating environments for
the UNIX operation system, and more recently, for developing and
promoting the Java programming language. Sun products include SPARC
workstations and the Solaris operating environment.
T-Music Software Glossary-T
If you drive the higher end of analog recording systems into overdrive,
there is a phenomenon called Tape Saturation, essentially a form of
distortion that is pleasing to the ear. It is easily overdone and also
highly overrated by some "sound authorities" (my opinion). I would
suggest not to believe anybody's story about this topic but to test this
effect for yourself if you have the opportunity. Digital systems use
Plug-ins to simulate this effect with more or less success. I have heard
some tape saturation Plug-ins that sound better than a real tape if you
put them in heavy overdrive but with all of this kind of effects also
this statement is quite subjective.
Also known as a "foo file" a temporary file is a file created to hold
information temporarily while a file is being created. After the program
has original file has been close the temporary file should be deleted.
Temporary files are used to help recover lost data if the program or
Track at once; In Track-at-Once recording, the recording laser is turned
off after each track is finished, and on again when a new track must be
written, even if several tracks are being written in a single recording
operation. Tracks recorded in Track-at-Once mode are divided by gaps. If
a data track is followed by an audio track, the gap is 2 or 3 seconds.
The gap between audio tracks is usually 2 seconds. There is nothing that
can be done by the software to suppress or reduce the gap, unless both
recorder and software support variable-gap Track-at-Once. All current CD
recorders support Track-at-Once. It is not recommended to write CD
masters in Track-at-once but in Disc-at-once
Latch Records adjustments you make to automation settings and creates
corresponding edit points on track envelopes. It begins recording, when
you first adjust a setting, but returns settings to previously recorded
values when you stop adjusting them.
The term tracker derives from Ultimate Soundtracker, a software written
by Karsten Obarski and released in 1987 for the Commodore Amiga. Tracker
is the generic term for a class of software music sequencers which, in
their purest form, allow the user to arrange sound samples stepwise on a
timeline across several monophonic channels. A tracker's interface is
primarily numeric; notes are entered via the keyboard, whilst length,
parameters, effects and so forth are entered in hexadecimal. A complete
song consists of several small multi-channel patterns chained together
via a master list.
A CD can contain up to 99 track IDs that identify the start of audio
tracks on a CD. If you need more identification points on your CD you
have to use Index points. A track can be
a minimum of 4 seconds long (600 sectors).
An amplitude modulation (usually with a sinus-wave). Contrary to vibrato
not the pitch, but only the volume of the sound is changed periodically.
A point at
which an effect can be seen. For example, the threshold of a compressor
is the minimum voltage that must be present bat the input before any
down a sound file to the desired size or shape.
When a tube is driven into saturation, it reacts by producing mostly
harmonic overtones. This is usually rather pleasant to the ear compared
to clipping of transistors or even digital clipping, which reacts by
producing mainly disharmonic overtones. This effect is used by guitar
amplifiers or similar equipment and can be emulated by software, mostly
Compressed WAV format. U-Law (or CCITT standard G.711) is an audio
compression scheme and international standard in telephony applications.
u-Law is very similar to A-Law, a variation of u-Law found in European
systems. This encoding format compresses original 16-bit audio down to 8
bits (for a 2:1 compression ratio) with a dynamic range of about
13-bits. Thus, u-Law encoded waveforms have a higher s/n ratio than
8-bit PCM, but at the price of a bit more distortion than the original
16-bit audio. The quality is higher than you would get with 4-bit ADPCM
formats. Encoding and decoding is rather fast and generally, widely
Universal Product Code/European Article Numbering
System; the first bar code symbology widely
adopted. Its birth is usually set at April 3, 1973, when the grocery
industry formally established UPC as the standard bar code symbology for
product marking. Foreign interest in UPC led to the adoption of the EAN
code format, similar to UPC, in December 1976.
V-Music Software Glossary-V
A more or less periodic change in pitch over time. Contrary to tremolo
not the volume, but the pitch of the sound is changed periodically.
The superimposition of the estimated varying short-term spectral
envelope of a signal on another. Used as an effect to create illusions
of singing instruments and other spectral hybrids of separate sound
Studio Technology; the audio engine created by Steinberg originally for
Cubase has been adopted as a standard by many other applications. It
allows software effects and instruments to "plug in" to a VST compatible
application like a sequencer.
Volume unit meter; Unlike peak meters which read instantaneous changes
in your audio signal the VU meters read a portion of the signal and
calculate the average level. The size of the signal that the meters read
is determined by the meters' integration time.
W-Music Software Glossary-W
The Wah-Wah effect is based on a band-pass filter with fairly high
resonance. As early as 1945 Leo Fender put one in his lap steel guitar.
Besides that ,some studio players found that by rotating the tone knob
it gave a Wah-Wah like sound. In 1965 Brad Plunkett of Thomas Organ Co
in the USA was working on a tone control and found the effect by
accident, word is when people heard this sound they all came in the lab
and where astounded by it. The effect was commercialized in 1966 by Vox
who called it the Vox Wah-Wah and Thomas Organ who gave it the name
Crybaby, since it sounded like a baby making noise.
Waveform Audio; uncompressed file
format was developed jointly by Microsoft and IBM as the standard format
for sound on PCs. WAV sound files end with a .wav extension and can be
played by nearly all Windows applications that support sound.
A waveform is the visual representation of wave-like phenomena, such as
sound or light. For example, when the amplitude of sound pressure is
graphed over time, pressure variations usually form a smooth waveform.
The WAVE-64 file format is defined as a true 64 bit file format that
allows to overcome the limitations of the RIFF/WAVE format. The file
format was originally defined by Sonic Foundry. In Summer 2003, Sony
Pictures Digital acquired Sonic Foundry's Desktop Software assets. Since
then, the new format is officially promoted as Sony Pictures Digital
Wave 64(TM). Companies are encouraged to support this format and no
royalties have to be paid to use it.
Short for Windows Driver Model. Microsoft invented this format to allow
hardware vendors to make one driver for all current and future Windows
operating system versions. All versions of Microsoft Windows after
Windows 95 have implemented WDM. WDM drivers can be installed under
Windows 98 SE, ME, 2000 and XP. Other Windows versions are not supported
(esp. Windows 95, Windows 98, Windows NT 4.0). WDM drivers have special
features that are not available on other driver models/formats. It
introduces another way to access the audiocard hardware, this method is
called WDM Kernel Streaming (WDM KS). The driver's kernel module is
accessed directly from the audio application. This method was first used
by Cakewalk in their SONAR software. By accessing the kernel module of
the driver directly from the application without the usage of any high
level API, very low latency figures can be achieved (similar to ASIO,
depending on the driver structure and hardware even lower than with
ASIO). Other (but not all) software vendors are now working to support
WDM KS inside their future audio applications.
WDM (Windows Driver Model) Kernel Streaming. The driver's kernel module
is accessed directly from the audio application. This method was first
used by Cakewalk in their SONAR software. By accessing the kernel module
of the driver directly from the application without the usage of any
high level API, very low latency figures can be achieved (similar to
ASIO, depending on the driver structure and hardware even lower than
with ASIO). Other (but not all) software vendors are now working to
support WDM KS inside their future audio applications.
White noise has a spectral frequency of 1, meaning that equal
proportions of all frequencies are present. Because the human ear is
more susceptible to high frequencies, white noise sounds very hissy.
White noise is generated by choosing random values for each sample.
Windows Media Audio; Microsoft's proprietary audio
codec designed to compete with MP3. Claims
competitive sound quality at lower bitrates.
(1) A type of computer used for engineering applications (CAD/CAM),
desktop publishing, software development, and other types of
applications that require a moderate amount of computing power and
relatively high quality graphics capabilities. Workstations generally
come with a large, high-resolution graphics screen, at least 64 MB
(megabytes) of RAM, built-in network support, and a graphical user
interface. Most workstations also have a mass storage device such as a
disk drive, but a special type of workstation, called a diskless
workstation, comes without a disk drive. The most common operating
systems for workstations are UNIX and Windows NT. In terms of computing
power, workstations lie between personal computers and minicomputers,
although the line is fuzzy on both ends. High-end personal computers are
equivalent to low-end workstations. And high-end workstations are
equivalent to minicomputers. Like personal computers, most workstations
are single-user computers. However, workstations are typically linked
together to form a local-area network, although they can also be used as
(2) In networking, workstation refers to any computer connected to a
local-area network. It could be a workstation or a personal computer.
(3) A Sound generating device, capable of playing different sounds or
instruments at the same time. It has built in sequencing or recording
abilities and usually a built in sound library of a variety of
X-Music Software Glossary-X
Metadata is information about the file, such as the authors name,
resolution, color space, copyright, and keywords applied to it. You can
use metadata to streamline your workflow and organize your files. This
information is stored in a standardized way using the Extensible
Metadata Platform (XMP) standard on which Adobe Bridge and the Adobe
Creative Suite applications are built. XMP is built on XML, and in most
cases the information is stored in the file so that it cannot be lost.
If it is not possible to store the information in the file itself, XMP
metadata is stored in a separate file called a sidecar file.
Y-Music Software Glossary-Y
Z-Music Software Glossary-Z