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-#

5.1 Surround Sound
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 PCM format.

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.

AC '97
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 codecs.

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.

AES 31
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 supported.

Amiga IFF-8SVX
The Amiga IFF-8SVX format is an 8-bit mono format from the Commodore Amiga computer.

Amp modeling
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 more realistic.
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.
Asio 2.0
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 remotely.

Audition Loop
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.

Auto zero-cross
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 "Zero" amplitude.
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.

B-Music Software Glossary-B

Batch conversion
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.
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
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 European Broadcasting 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 information about...
Description... Describes the audio file in up to 256 characters.
Originator... Specifies the name of the audio file’s 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 as “2004-06-08.”
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 clip’s 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 File.
UMID Specifies... Unique Material Identifier Data (UMID) according to the SPMTE 330M Standard. This information is read-only.

C-Music Software Glossary-C

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 CDs.

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.

CD Extra
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.

CD Text
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.

time to create a shimmering effect. makes a single voice sound like multiple voices in unison. You can implement chorusing by sending a sound through a series of delays whose delay times are slowly being modulated.

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 their data.

Color depth
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:

Color Depth:
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.

Reduces 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.

Convolution reverb
In 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.

A 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
This 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.

A 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.

DC offset
DC 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 center again.

A 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.

Dial tone
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 hardware-specific.

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.

Direct 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 0’s (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&w’s 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 associated software. 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.

DVD Players
There are three basic types of DVD players:

  • DVD-Audio player
    This could either be an audio-only player (AOP), or a player capable of displaying visual menus, text and still images.

  • DVD-Video player
    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 video DVDs.

Dynamic automation
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.

E-Music Software Glossary-E

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.

Ensoniq Paris
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

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...

F-Music Software Glossary-F

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 analyzed.

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.

Floating point
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 numbers.

FM synthesis
Frequency Modulation 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.

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.

Frequency sweeps
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...

G-Music Software Glossary-G

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

Glass Mastering
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.
GSIF Multiclient
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.

H-Music Software Glossary-H

Harmonic (overtone)
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 specialized programs.

Hidden tracks
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 Intel’s 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.

AC ’97 High Definition Audio Benefit
11.5 MB/s max bandwidth 24 -48 MB/s bandwith Bandwidth for more channels and microphone array inputs at higher sample rates
Fixed bandwidth assignment, fixed slot base protocol Dynamic bandwidth assignment Bandwidth delivered where it's needed
Pre-defined DMA usage General purpose DMAs Support for multi-streaming / multiple similar device types
Single stream support (in and out) Support for multiple streams (in and out) Support for new Digital Home / Digital Office usage models
Clock provided by primary codec Clock provided by the Intel® I/O Controller Hub (ICH) Single, high-quality, stable clock source for synchronization
Stability depending on software provider Unique Microsoft bus driver Single bus driver for more OS stability and base functionality
Limited device sensing / jack retasking Full device sensing / jack retasking Support for full audio Plug and Play
2-element (stereo) array microphone support 16-element array microphone support 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 And Videograms; 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 copyright laws
  • 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 computer’s 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 follows:

  • Country Code (2 ASCII characters).

  • Owner Code (3 ASCII characters or digits).

  • Recording Year (2 digits or ASCII characters).

  • Serial Number (5 digits or ASCII characters).

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 interconnection medium. 

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.

J-Music Software Glossary-J

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.

Jog/Shuttle Dial
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.

K-Music Software Glossary-K

L-Music Software Glossary-L

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 playback stops.

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.

Loudness maximizer
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.

M-Music Software Glossary-M

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.

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 Commands
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 define.
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.

Microsoft ADPCM
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 Windows.

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.

MIDI clock
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.

Midi SDS
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 environments.

MIDI song position pointer
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.

MOD files
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 audio.
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 cooperative multitasking.

Multiband compressor
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 loudness.

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 040.

Noise Gate
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.

Noise reduction
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.

Notch filter
filter that eliminates a narrow frequency range from a signal without affecting the rest of the sound.

O-Music Software Glossary-O

Ogg Vorbis
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.

Operating System;
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.

Personal 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.

Phase-Meter demonstrated

Similar 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.

Phase reverse
This 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
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
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 amplitude.

Pitch bender
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 slowing down.

Pitch correction
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 pitch correction.

Pitch shifter
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.


Integration Time

UK PPM 10 ms
EBU PPM 10 ms
DIN PPM 5 ms
Nordic PPM 5 ms

PQ code(d)
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 pre-emphasis flag.

Prerecord buffer
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 too late.

The term comes originally from analog tape machines. It means
the 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. Eight years 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.

Quick Time
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 and ActiveMovie.

R-Music Software Glossary-R

Random 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.

Real Audio
An audio 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.

Real Video
An video 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
The 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 application link.

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)
  • Waveform data (.WAV)
  • Bitmapped data (.RDI)
  • MIDI information (.RMI)
  • A bundle 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 specification.

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 demonstrated
  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.

A 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 original instrument.

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. Sample Rate Demonstration
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.

Sample vision
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 loop).

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:

  1. No restrictions

  2. Single generation copy

  3. 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.

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 scripts.

Ability to move forward/reverse in the audio while listening to the audio. Similar to cueing during wind/rewind on analogue recorders.

Small Computer System Interface. Bidirectional, parallel interface to connect up to 7 (15 with wide SCSI) external devices to a computer.

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 monophonic.

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.

Spectrum analyzer
An instrument or software which displays the frequency spectrum of a sound signal
. 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.
A 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.

Sony Pictures 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 64™
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 several files.

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.

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

Tape saturation
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.

Temporary file
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 computer crashed.

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 modus.

Touch 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, 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.

Track ID
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 compression occurs.

Cutting down a sound file to the desired size or shape.

Tube emulation
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 Plug-Ins.

U-Music Software Glossary-U

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 supported.

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 sources.

Virtual 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.

Waveform demonstrated 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
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 stand-alone systems.

(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 instruments.

X-Music Software Glossary-X

XMP Metadata
Metadata is information about the file, such as the author’s 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