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(Average Bit Rate)

Similar to variable bit rate encoding, rather than encode every portion of the audio with the same, constant bit rate, the bits are concentrated where they are most needed. The complexity of the information content of audio varies over time. The perceived quality of the audio is increased by using more bits where the complexity is higher and fewer bits where it is lower. For average bit rate encoding, this variation is done within the limit of the specified average. Therefore, the size of the resulting file can be calculated before it is encoded. Average bit rate encoding is sometimes used as synonymous with variable bit rate encoding, but the two, while similar, have differences. ABR is a later method used in MP3 files.

It is useful for improving the perceived audio quality where maintaining limited, known file size is needed.

See also CBR, VBR, and bit rate.


(Analog to Digital Converter)

A device which turns an analog signal into one that is digital. This process is called digitizing.

Equipment which records sound (from the air) and stores it in digital form has an analog to digital converter. Examples are the sound card on a computer (when used to capture an analog input) and hand-held digital recorders.

See also DAC.


File format for uncompressed PCM audio data. Originally used on Macs.


A process by which the loudness of a recording is changed. Generally indicates an increase in loudness, but many software applications allow negative amplification.


Amplitude is the power of a signal. The stronger the amplitude, the stronger the energy carried. Amplitude, in this application, is relational to the strength or volume of a signal, usually measured in decibels.

See more technical jargon at Wikipedia


Having values which vary continuously across a range, as opposed to varying discretely. The electric clocks in use before the digital age qualify as analog, for their hands sweep in a continuous manner. Sound waves in the air vary continuously in amplitude and are analog. So is the music on vinyl records and standard music cassette tapes.

The opposite of analog is digital.

Analog to Digital Converter

See ADC.

Audio Book

An audio book is a recording of the contents of a book read aloud. It is usually distributed on compact discs (CDs), cassette tapes, or digital formats (such as MP3 or Audible audiobook). The term "audio book" has been synonymous with "books on tape" for roughly 20 years. Cassette tape sales still comprise roughly 40% of the audio book market, with CDs the other dominant format type. (Definition courtesy of Wikipedia.)

Average Bit Rate

See ABR.



Bandwidth is almost a generic term that fits various technologies. In regards to Internet bandwidth, it is the amount of data that can be sent through a network connection and is measured in bits (or baud) per second (bps) - call it transmission speed. Transmission speeds are rated in bytes per second - individual digits within the binary numerical system are called bits. A collection of bits are bytes. A user's Internet connection may be rated as a certain speed of bps in the range of kilo-, mega- (million), giga- (billion) bps. The term bandwidth is deceptive and when experiencing a lag in an Internet connection it is important to know that transmission speed is affected for multiple reasons: The user's computer; How long does it take to get to a site?; What is the response time of the site? All factor in. So a user's Internet connection can be slow for different reasons. High bandwidth allows fast transmission, but a when talking to a low bandwidth site/device the user gets a speed equal to the difference of the speeds.

For more information, see this article


(Book Coordinator)

A LibriVox volunteer who starts a collaborative project, inviting other readers to read sections of a book. The Book Coordinator allocates sections to volunteers 'signing up' for them, answers questions, and collects the recordings, making sure that the files adhere to the standards required for cataloging. The files are then handed over to the project's MC.

Bit Rate

The number of bits per unit of time used to represent audio information, as stored in a particular file format, usually expressed as Kilobits per second (Kbps). It functions as a measure of the resolution, or quality, of the audio. It is used with MP3 files.

Originally, the MP3 format use a constant bit rate (aka, CBR) and then was enhanced with average ( ABR) and variable ( VBR) bit rates methods. Older software can have problems with MP3 files encoded in VBR.

Book Coordinator

See BC.



(Constant Bit Rate)

Every portion of the audio is encoded with the same, constant, bit rate. That is, the same number of bits is used to represent each fragment of the audio regardless of its complexity. How many bits are used is determined by the bit rate set when the file is encoded. The size of the resulting file, before encoding, can be calculated from the bit rate and the length of the the audio. CBR is the first method used in MP3 files.

It is useful for creating files intended for streaming, especially low resolution files for low bandwidth connections. It is also faster to encode than other bit rate methods.

See also ABR, VBR, and bit rate.

Collaborative Project

A project in which two or more volunteers read sections of a book to share the work. These group projects are 'run' by BCs.


Compressors reduce the dynamic range of an audio signal. In the real world, it is generally an amplifier with two gain levels: the gain is unity for input signal levels below a certain threshold, and less than unity for signals with levels above the threshold. For example, compressors can be used to eliminate the variations in the peaks of an electric bass signal by clamping them to a constant level (thus providing an even, solid bass line.) Compressors can also be useful in compensating for the wide variations in the level of a signal produced by a singer who moves frequently or has an erratic dynamic range.

A compressor reduces the difference between the loudest and quietest parts of a performance. It works on a threshold system where signals exceeding the threshold are processed and those falling below it pass through unchanged. When a signal exceeds the threshold the compressor automatically reduces the gain. How much gain reduction is applied depends on the 'compression ratio' which on most compressors is variable: the higher the ratio, the stronger the compression. Very high ratios cause the compressor to act as a limiter where the input signal is prevented from ever exceeding the threshold.

Compressors are the most commonly used processor and are particularly popular for maintaining constant vocal and bass guitar levels live and in the studio. This is because, out of all instruments, singers tend to vary their levels the most. Compressors help to achieve the much sought-after tight, "punchy" sound. (Written from the perspective of operating a sound mixing board.)

A device designed to control or reduce the dynamic range of an audio signal. (Written from the perspective of operating a sound mixing board.)

Constant Bit Rate

See CBR.


Copyright gives the copyright holder exclusive rights on a text for a limited period of time. This means that no one else can reproduce the text or make derivative works (such as audio recordings) while the copyright is in force. Eventually, though, copyright expires, and the text enters the public domain. This means that anyone can use the text however they wish.

For more information, see Copyright and Public Domain.



(Digital to Analog Converter)

A device which turns a digital signal into one that is analog.

Equipment which plays digital audio through speakers has a digital to analog converter. Examples are the sound card on a computer (the part that creates sound output) and hand-held digital audio players.

See also ADC.


(Digital Audio Player)

A device that stores, organizes, and plays digital music files. It is more commonly referred to as an MP3 player (because of that format's ubiquity), but DAPs often play many additional file formats. (Definition (Wikipedia)).


The decibel [dB] describes a logarithmic ratio of two values. More details at Wikipedia. The interpretation of a value is dependent on the use. Note that a Value of 0dB means: The ratio is exactly one.

While recording the ratio described is usually the highest possible value divided by the actual value. This implies that on that scale the highest result is 0dB. The lowest value will be around -120dB.

While playing back the ratio usually is the Sound Pressure Level (SPL) of the actual sound divided by the a reference SPL. It is a common measure of acoustic volume. A level of 0dB should just be audible to very healthy ears. 100dB will damage your hearing if exposed continuously, anything above is way too high. There is a nice comparison chart on the internet.

Decode, Decoder

The process of converting data in a compressed to an uncompressed form.

All software and hardware which play MP3 files use a decoder.

The opposite process is encoding.

Dedicated Proof Listener

Dedicated Proof Listeners (DPLs) are volunteers who have agreed to proof-listen all the files in a particular project. They are able to access the Magic Window in order to make notes about the recordings. To sign up for listening to all files in an ongoing project, simply post in the project thread.


Having values which vary discretely across a range, as opposed to varying continuously. That is, the values vary in distinct steps, without intermediate values. Digital clocks and watches are an example. Audio on audio CDs and on computers is stored in digital form.

The opposite of digital is analog

Digital to Analog Converter

See DAC.

Digital Audio Player

See DAP.


Computer term; both noun and verb. Downloading is the process of transferring a file from another computer to your computer, usually by means of a network connection. The file is called a download. The opposite term is upload.


See Dedicated Proof Listener.

Dynamics Processing

(This term used by Cool Edit and Audition.) The dynamics processor varies the output level of a waveform, based on its input level. This lets you limit or compress the dynamic range of a sample so that the perceived loudness is kept below a defined limit, or so that the waveform's overall dynamic range is kept at roughly the same level. You can also expand or gate the signal so that low-level signals are reduced in level, thereby increasing the perceived dynamic range, or so that signals that fall below a certain threshold (i.e., noise) are eliminated.


Encode, Encoder

The process of converting data in one form to another, usually compressed, form. Used to refer to the process of creating MP3 files. Programs which do the encoding are called encoders. An MP3 encoder converts audio, typically in either #wave Wave or #AIFF AIFF format, by encoding it into MP3 format.

The opposite process is decoding.


Expanders are used to expand the dynamic range of an audio signal (basically the opposite of a compressor.) An expander can also be considered an amplifier with two gain levels: the gain is unity for input signal levels above a certain threshold, and less than unity for signals with levels below the threshold. An expander boosts the high-level signals and attenuates low-level signals.

Expanders accomplish much the same task as gates, though they are more like compressors in reverse. Compressors affect the gain of signals exceeding the threshold, while expanders act on signals falling below the threshold. A gate will close completely when the signal falls below its threshold, but an expander works like an automatic mixing engineer who pulls down the signal when the signal falls below the threshold; the more it falls below the threshold - the more he pulls down the fader.

Expanders are most often use in Studio recording to provide the best mix signal to noise ratio when producing final masters. (Written from the perspective of operating a sound mixing board.)




Gain is the factor of how much the level of a signal is increased or amplified. Normally expressed in decibels.


Noise Gates are a special type of expander that can be used to reduce or eliminate noise below a threshold level. It does this by heavily attenuating signals with levels that fall below the threshold. It's often used to totally cut off the signal level during a musical pause so as not to pass background noise. It can also be used to silence the pauses in speech.

A gate is designed to shut down the audio signal path when the input signal falls below a threshold set by the user. It may be used to clean-up any signal that has pauses in it. For example gates are widely used to prevent 'spill' between adjacent mics on a multi-mic'd drum kit where, say, a tomtom mic may pick up the snare drum. (Written from the perspective of operating a sound mixing board.)

A user-adjustable electronic device that switches off the signal path when the signal falls below a certain predetermined level or threshold. Typically used to ensure silence between pauses in the signal during vocal passages or to prevent ‘spill’ between the close-proximity, multiple mics on a drum kit. (Written from the perspective of operating a sound mixing board.)



ID3 tag

The MP3 file format provides for human-readable, descriptive information about the audio content of the file. This includes items like the artist or author, the title, the album or book, and the date. This information is displayed to the listener by computer-based playback software or hand-held audio players.

For more information, see ID3 tags.





LibriVox is the sort of name you invent when you don't know Latin at all. "Libri" means book, and "Vox" means "voice"... so it means: "BookVoice." But it's possible Latin scholars would cringe at some error in the melding together of the two words. Still, it sounded pretty good. Another Latin word for book is "Liber", which also means: "child, offspring;" and "free, independent, unrestricted." So we like to think LibriVox might be interpreted as "child of the voice," and "free voice." Finally, the other link we like is library you could imagine it to mean Library of Voice, which sounds cool too. But all this is the result of using online Latin dictionaries with no formal training.

For information on what LibriVox is, as opposed to what it means, see LibriVox.


Limiters are compressors with a compression ratio of 10:1. This has the effect of reducing or "limiting" input signals that exceed a specified threshold level, so that the output does not increase in gain beyond that point. In other words, a limiter only allows the dynamic range at its input to increase up to a certain point (determined by the threshold setting). Beyond this level, as the input continues to increase in gain, the output level remains relatively constant and does not increase in volume.


Magic Window

Also known as the Magic Box. This is an interface between the forum and the database. Information is sorted here about the particular project in progress, including which chapters have been assigned, to whom, and links to any completed files. The BC and DPL is able to access and edit the window for their own projects. MCs and admins are able to access and edit all windows.


(Meta Coordinator)

A LibriVox volunteer who looks after cataloging. Every recording project is assigned one MC. The MC (also sometimes called 'meta') answers any questions the BC or solo reader might have, puts the project in the 'To Come' section on the catalog and, upon project completion, takes over all files from the BC or solo reader. A new catalog page is then created and the recordings are made available for download so listeners can enjoy them.

Meta Coordinator

See MC.


(MPEG-1 Audio Layer 3)

A data and file format for storing audio information in a compact way.

The MP3 format is based on psycho-acoustical research, which deals with how the human ear and mind work together to perceive sound. The goal is to compress the audio data with the least deterioration in audio quality. In other words, small file size and big sound. Clearly, these two factors conflict, for the more you "squeeze" the audio in compressing it, the worse it will sound. Fortunately, for most ears, the effects of compression don't become audible until the level of compression becomes very high.

When uncompressed audio (say, in Wave or AIFF format) is encoded to MP3 format, the user can choose the level of compression that is used, which is expressed as a bit rate. As mentioned above, there's a trade off between file size and (retained) audio quality. How much compression can be applied without the effects becoming noticeable depends entirely on the nature of the audio being compressed, so there are no fixed rules for choosing. Especially for music, a compression level that works well for one song may create poor results for different song.

MP3 is known as a lossy format. This means that, at most compression bit rates, information content is lost. Therefore, if the MP3 file is decoded back to, say, Wave format, the original and reconstituted Wave files are not identical in terms of their audio quality (even though they will be essentially the same size). That's because compression destroys ("loses") some information.

See also ID3 tag.
For more information, see

Wikipedia's article on psycho-acoustics


Noise Gate

(See Gate)

Normalization, Normalize

Amplifies the signal to within the specified percentage of the maximum level. Use normalization to achieve the greatest amount of amplification that will not result in clipping (when set to 0dB or 100%).

Most programs that normalize sound files do so by adjusting all the samples so that the loudest single sample is at some specified value.


Ogg Vorbis

"A patent- and royalty-free, lossy audio compression technology from the Xiph.Org Foundation (, which is dedicated to open source multimedia standards for the Internet. Ogg is the project name and container format. Vorbis is an audio codec that generates 16 bit samples at 16KHz to 48KHz, providing variable bit rates from 16 to 128 Kbps per channel. Ogg Vorbis is considered comparable to AAC and better than MP3 in sound quality as well as providing lower bit rates and smaller file sizes." Courtesy of PC Magazine.



(Pulse Code Modulation)

PCM is a way of converting analogue signals to digital data in a computer. Uncompressed audio is typically PCM data packed in a container such as audio CDs, Wave or AIFF files.


(Project Gutenberg)
(See Project Gutenberg)


Plosive is a speech sound that is produced when there is complete closure of the air passage, followed by a burst of released air, often in the sound of (d) or (p). Plosive is termed as a stop consonant - read more about this at Wikipedia.


also plug-in

Computer term. A program or module designed to add to or change the functionality of another program. An example is a plugin to add the ability to play a specific file format to a media player program.

For more information, see the Wikipedia.

Project Gutenberg


Project Gutenberg is the oldest producer of free ebooks on the Internet. Its collection was produced by hundreds of volunteers.

For more information, see the Project Gutenberg Web site and the LibriVox page on

Copyright and Public Domain.

Public Domain

Works in the public domain may be freely used by anyone for any purpose. This includes selling them and making derivative works, such as audio recordings. See also Copyright in this glossary.

For more information, see Copyright and Public Domain.

Pulse Code Modulation

(See PCM)



[ Supply definition ]


Replay Gain

An alternative to standard peak normalization, which is based on perceived loudness. It is the default method used by MP3Gain, a program for adjusting the volume level in MP3 files.

For more information, see the Replay Gain home page.


Computer term. It means to click the right mouse button while the mouse pointer is positioned over something. On MS Windows, this usually opens a pop-up menu.

An example use is to download audio files from !LibriVox if you use an MS Windows-based PC.


Sample Frequency

supply definition


The electrical representation of an input such as sound.


Computer term, often used in the plural. A skin changes the way a program looks, often radically. The program is created to allow any person, including users, to develop new user interfaces -- how the parts, controls, and displays of the program look and are arranged, that is, new skins. The Web sites for such programs usually make these skins available to other users. After you download skins to your computer, you can switch between them using the program's menu for that purpose.

For more information, see the Wikipedia.

Solo Reader

A LibriVox volunteer who reads a whole text by him/herself, as opposed to joining a Collaborative Project.


A method often referred to as "streaming media" that allows a large media file ( MP3, Ogg Vorbis, WAV, AIFF) to be continuously downloaded while being viewed or heard before the completion of downloading.




Variable Bit Rate

See VBR.


(Variable Bit Rate)

Similar to average bit rate encoding, rather than encode every portion of the audio with the same, constant bit rate, the bits are concentrated where they are most needed. The complexity of the information content of audio varies over time. The perceived quality of the audio is increased by using more bits where the complexity is higher and fewer bits where it is lower. For variable bit rate encoding, this variation is done within broad limits. Therefore, it is not possible to know the size of the resulting file with precision before it is encoded. Variable bit rate encoding is sometimes used as synonymous with average bit rate encoding, but the two, while similar, have differences. VBR is a later method used in MP3 files.

It produces the highest perceived audio quality at the cost of larger, somewhat unpredictable file size.

Older software can have problems with MP3 files encoded in VBR.

See also CBR, ABR, and bit rate.



A file format originally developed for MS Windows. It is the format required by PC sound cards. The most common data format stored in Wave files is PCM. Wave files containing audio in PCM format are what most Windows-based audio editing program operate on. The equivalent format on the Mac is AIFF.

On MS Windows, when one extracts tracks from an audio CD, the resulting data is in Wave format, even if the application is automatically converting it to MP3 format.

Strictly speaking, Wave is a file format, as distinct from a data format. This means a Wave file is really container, that has a certain structure, for holding data which can be in a variety of formats. The most common data format in Wave files is PCM. Because of this, in general practice, Wave is thought of as a data format and is, in this case, synonymous with PCM.