AAC - supports wide range of sampling rates (8 to 96kHz), 48 audio channels, 15 auxiliary low-frequency enhancement channel and up to 15 embedded data streams. The format supports bit-rates from 8kbps to 320kbps.
MPEG-2 AAC evolved out of a search for an efficient coding method for surround sound signals. a2b music and Liquid Audio are delivery systems based on AAC. Widespread use of AAC, similar to that of MP3, is not anticipated due to high licensing costs which make software development an expensive prospect.
Source – www.mp3machine.com/glossary
Bit - Unit. Binary digit. The unit of information; the amount of information obtained by asking a yes-or-no question; a computational quantity that can take on one of two values, such as false and true or 0 and 1; the smallest unit of storage - sufficient to hold one bit.
A bit is said to be "set" if its value is true or 1, and "reset" or "clear" if its value is false or 0. One speaks of setting and clearing bits. To toggle or "invert" a bit is to change it, either from 0 to 1 or from 1 to 0.
The term "bit" first appeared in print in the computer-science sense in 1949, and seems to have been coined by early computer scientist John Tukey. Tukey records that it evolved over a lunch table as a handier alternative to "bigit" or "binit."
Source – foldoc.org
Bit Rate - The resolution of the recording of the music file. MP3's are generally recorded at 128 kbps for the best compromise between speed and file size.
Source – www.mp3phonezone.com/glossary.htm
Bit rate is the amount of information (bits) transferred in a second ("bps" is the abbreviation of bits-per-second). In terms of MP3 audio files, the bit rate unit is more commonly referred to as 'kbps', which is thousand-bits-per-second. The higher the bit rate or 'kbps' of an MP3 file, the higher the sound quality. Most MP3 encoders support a range of bit rates from 24kbps up to 320kbps (or 320,000 bits per second). The most widely used 'standard' bit rate for MP3s is 128kbps, but below that is not especially enjoyable to listen to. At around 160kbps there is a noticeable improvement in the audible quality of an MP3 encoded audio file.
Source – www.mp3machine.com/glossary
Codec - A codec is a software component that translates video or audio between its uncompressed form and compressed form. A codec is used to compress digital audio files by discarding redundant data.
MP3 encoders and decoders are audio codecs. The compression of a WAV or AIFF file to MP3 format using a MPEG layer III audio codec is encoding, similarly, an MP3 file can be 'decompressed' or decoded to WAV or AIFF format using a separate codec. There are also codecs for WMA, QuickTime 4 Streaming Audio, RealAudio and other audio formats.
As a form of compression, MP3 is based on a psycho-acoustic model which recognizes that the human ear cannot hear all the audio frequencies on a recording. The human hearing range is between 20Hz to 20Khz and it is most sensitive between 2 to 4 KHz. When sound is compressed into an MP3 file, an attempt is made to get rid of the frequencies that can't be heard. As such, this is known as 'destructive' compression. After a file is compressed, the data that is eliminated in the creation of the MP3 cannot be replaced.
Source – www.mp3machine.com/glossary
CD (Compact Disc) - A 4.72 inch disc developed by Sony and Philips that can store, on the same disc, still and/or moving images in monochrome and/or color; stereo or two separate sound tracks integrated with and/or separate from the images; and digital program and information files.
The same fabrication process is used to make both audio CDs and CD-ROMs for storing computer data, the only difference is in the device used to read the CD (the player or drive).
Source – foldoc.org
CD Known by its abbreviation, CD, 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.
Another type of compact disc, called CD-Rs and CD-RWs, can have their data erased and overwritten by new data. Currently, erasable optical storage is too slow to be used as a computer's main storage facility, but as the speed improves and the cost comes down, optical storage devices are becoming a popular alternative to tape systems as a backup method.
Source – www.webopedia.com
Compression - Compression is the act of fitting something into a smaller space than it would normally fit into. In the case of MP3 files, they are generally compressed at a ratio of 12 to 1. This allows you to download a file from the internet much faster than was otherwise possible.
Source – www.mp3phonezone.com/glossary.htm
Constant Bit Rate (CBR) - encoding means that you encode a file at a fixed rate, such as 128 kpbs. For many people this is a common method of encoding MP3s. You can usually tell CBR files because they have consistent file sizes and sound quality.
Source – www.mp3machine.com/glossary
Constant Bit Rate (CBR)
- encoding means that you encode a file
at a fixed rate, such as 128 kpbs. For
many people this is a common method of
encoding MP3s. You can usually tell CBR
files because they have consistent file
sizes and sound quality.
Source – www.mp3machine.com/glossary
DAE - is also referred to as Digital Audio Extraction (DAE). Depending on your operating system (platform) CD ripping software will record the CD audio (.CDA) tracks to WAV format (e.g. Windows) or AIFF (e.g. Macintosh). It is usually as simple as inserting a CD into the CD-ROM drive of your computer and selecting the tracks to be recorded to hard drive.
Ripping can also refer to the recording
of vinyl records to digital audio. Several
ripping programs now incorporate the option
to encode recordings to MP3 and other
compressed audio formats.
Source – www.mp3machine.com/glossary
DRM (Digital Radio Mondiale) - A form of monaural digital broadcast using carrier frequencies below 30 MHz. DRM uses MPEG-4 AAC Main Profile and SBR at data rates of 16-25 kbps.
Source – foldoc.org
Encode - To convert data or some physical quantity into a given format.
Source – foldoc.org
FreeDb - FreeDb is a database to look up CD information using the internet. This is done by a client (a FreeDb aware application) which calculates a (nearly) unique disc ID for a CD in your CD-Rom and then queries the database. As a result, the client displays the artist, CD-title, track list and some additional info.
Source – www.freedb.org
ID3 -
The ID3 tag is attached to an MP3 audio
file to carry information relevant to
that MP3. The development of ID3 began
in 1996 when it was realized that by adding
a small chunk of extra data at the start
or end of the audio data the MP3 file
could hold information about the audio
and not just the audio data itself.
File management software and many audio/media players rely on ID3 tag data when working with MP3 files to allow better presentation, sorting and classification of files. Software is available that can edit the information contained in an MP3's ID3 tag, commonly referred to as 'tag editors'. The entire size of the ID3 tag is 128 bytes with a certain amount of bytes allocated to store the song title, artist, album, year, genre and comment/s.
ID3v2 now allows extensive comments,
picture, lyrics and other information;
far more information than the original
ID3 tag.
Source – www.mp3machine.com/glossary
Kilobyte
- Unit. (KB) 2^10 = 1024 bytes.
Source – foldoc.org
Lossless
Compression - A term describing
a data compression algorithm which retains
all the information in the data, allowing
it to be recovered perfectly by decompression.
Source – www.mp3machine.com/glossary
Lossy Compression - A
term describing a data compression algorithm
which actually reduces the amount of information
in the data, rather than just the number
of bits used to represent that information.
The lost information is usually removed
because it is subjectively less important
to the quality of the data (usually an
image or sound) or because it can be recovered
reasonably by interpolation from the remaining
data. MPEG and JPEG are examples of lossy
compression techniques.
Source – Dictionary.com
MP3 is a lossy format. Based on the compression
settings chosen by the user, some of the
audio data is thrown away or 'lost' to
decrease the actual compressed file size.
This is why the more an MP3 is compressed
(low bit rate and sample rate, mono), the
poorer the sound quality. As there are
some elements in any sound recording that
are inaudible to the human ear, MP3s still
manage to sound good even with the loss
of data.
Source – www.mp3machine.com/glossary
Lossy Compression - MP3 uses a compression
scheme that is considered "Lossy".
What this means it that it throws some
of the information of the original file
away to achieve a smaller file size.
Source – www.mp3phonezone.com/glossary.htm
Mono/ Monophonic - A system where all the audio signals are mixed together and routed through a single audio channel, or sound produced by a system in which one or more microphones feed a single signal-processing amplifier whose output is coupled to one or more loudspeakers.
A mono signal output through two or more
speakers will still be mono, as the same
content or audible information is being
produced by each speaker.
Source – www.mp3machine.com/glossary
MP3 (MPEG-1 Audio Layer 3) - A digital audio compression algorithm that achieves a compression factor of about twelve while preserving sound quality. It does this by optimizing the compression according to the range of sound that people can actually hear. MP3 is currently (July 1999) the most powerful algorithm in a series of audio encoding standards developed under the sponsorship of the Moving Picture Experts Group (MPEG) and formalized by the International Organization for Standardization (ISO).
MP3 is very different from Layer 2, using an additional MDCT layer to increase frequency resolution. Its scale factor groups are more optimized for the human ear, and it uses nonlinear sample quantization and Huffman coding.
MP3 files (filename extension ".mp3") can be downloaded from many World-Wide Web sites and can be played using software available for most operating systems (also downloadable), e.g. Winamp for PC, MacAmp for Macintosh, and mpeg123 for Unix.
MP3 files are usually downloaded completely
before playing but streaming MP3 is also
possible. A program called a "ripper"
can be used to copy a selection from a
music CD onto your hard disk and another
program called an encoder can convert
it to an MP3 file.
Source – foldoc.org
MP3 Is the file extension for MPEG, audio layer 3. Layer 3 is one of three coding schemes (layer 1, layer 2 and layer 3) for the compression of audio signals. Layer 3 uses perceptual audio coding and psychoacoustic compression to remove all superfluous information (more specifically, the redundant and irrelevant parts of a sound signal. The stuff the human ear doesn't hear anyway). It also adds a MDCT (Modified Discrete Cosine transform) that implements a filter bank, increasing the frequency resolution 18 times higher than that of layer 2.
The result in real terms is layer 3 shrinks
the original sound data from a CD (with
a *bit rate of 1411.2 kilobits per one
second of stereo music) by a factor of
12 (down to 112-128kbps) without sacrificing
sound quality.
*Bit rate denotes the average number of
bits that one second of audio data will
consume.
Because MP3 files are small, they can
easily be transferred across the Internet.
Controversy arises when copyrighted songs
are sold and distributed illegally off
of Web sites. On the other hand, musicians
may be able to use this technology to
distribute their own songs from their
own Web sites to their listeners, thus
eliminating the need for record companies.
Costs to the consumer would decrease,
and profits for the musicians would increase.
Source – www.webopedia.com
MP3 - The most common file format for
digital music. MP3 is popular because
it allows songs to be compressed to more
than 1/10 the original size with little
noticeable loss of quality. It is short
for MPEG 1, Audio Layer 3.
Source – www.mp3phonezone.com/glossary.htm
MPEG - Motion Picture
Expert Group - A set of standards for
compressing digital movie and sound files.
Source – www.mp3phonezone.com/glossary.htm
Ogg Vorbis - Ogg Vorbis is an audio compression format for high quality (44.1-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bit rates from 16 to 128 kbps/channel.
Ogg Vorbis supporters claim the format features better sound quality and a smaller file size.
As a lossy compression format, is similar to formats such as MP3, VQF and AAC. It is different from these other formats because it is free, open source, and un-patented.
Vorbis can also encode and decode as
a single pass, real-time stream. You can
recognize Ogg Vorbis files by the .ogg
extension.
Source – www.mp3machine.com/glossary
Red Book - It is said that the original CD standards document was bound in red covers, hence its name. Subsequent CD standards documents have followed suit with color names; Green Book, Orange Book, White Book, Yellow Book. Red Book specifications formed the basis for all later CD technologies and specifies the format of the CD-DA (Compact Disc - Digital Audio) such as tracks, sampling, coding, sector and block layout. The CD-DA is defined as a content medium for 16 bit, 44.1kHz digital audio.
The actual physical characters of the
disc are also defined in the Red Book,
including the material make-up and data
allocation within the disc.
Source – www.mp3machine.com/glossary
Ripper - A software program
that "grabs" digital audio from
a compact disc and transfers it to a computer's
hard drive. The integrity of the data
is preserved because the signal does not
pass through the computer's sound card
and does not need to be converted to an
analog format. The digital-to-digital
transfer creates a WAV file that can then
be converted into an MP3 file.
Source – www.webopedia.com
Sample
Rate - The sample rate of an audio
recording partially determines the overall
sound quality. In the recording process,
audio samples are saved to memory or disk;
the rate each sample of audio input is
recorded per second is the sample rate.
The sample rate is measured in Hertz (Hz
- cycles per second) and Kilohertz (kHz
- thousand cycles per second).
CD quality audio has a sample rate of
44,100Hz, 16-bit (resolution) and stereo
(channels). The most common sample rates
are 11, 22 and 44kHz, with most recording
software supporting sample rates from
6kHz up to 192kHz. Like early footage
filmed at a low frame rate looks flickered
and robotic, the quality of an audio recording
decreases as the sample rate is lowered.
For audio recordings destined to be encoded
to MP3, 22kHz is considered acceptable.
Source – www.mp3machine.com/glossary
Stereo/Stereophonic - Stereo sound systems have two or more separate audio signal channels where the signals have specific levels and phase relationships to each other. When reproduced through a suitable system there will be an apparent reproduction of the original sound source(s). Stereo can replicate the aural perspective and position of instruments within a band on stage. A listener's proximity to a speaker or speakers of a stereo system will often determine which instruments or tones they will hear.
Stereophonic - Of or used in a sound-reproduction system that uses two or more separate channels to give a more natural distribution of sound.
Stereo audio uses about twice the bandwidth
of mono audio because of the two separate
channels, with much of the information
duplicated on both channels. MPEG audio
can use conservative methods to maintain
audio quality but reduce bandwidth by
retaining only the audio information perceived
as important to the stereo image.
Source – Dictionary.com
Variable
Bit Rate (VBR) - Compresses music
according to how much data there is. In
theory, VBR can result in better sounding,
smaller files. Some MP3 players cannot
properly play files encoded with VBR.
Source – www.mp3phonezone.com/glossary.htm
Variable Bit Rate encoding is a method that ensures high audio quality bit-allocation decisions during encoding. The encoder allocates an appropriate amount of data per second, depending on the complexity of the audio file.
If there are very complex parts in a song it will use a quite high bit rate and a lower bit rate for something such as silence. The average bit rate may not be as high as the bit rate of an MP3 of the same quality with constant bit rate.
You should use VBR encoding when consistent
audio quality is the top priority.
Source – www.mp3machine.com/glossary
WAV - A sound format developed by Microsoft and used extensively in Microsoft Windows. Conversion tools are available to allow most other operating systems to play .wav files.
.wav files are also used as the sound
source in wave table synthesis, e.g. in
E-mu's SoundFont. In addition, .wav files
are also supported by some MIDI sequencers
as add-on audio. That is, pre-recorded
.wav files are played back by control
commands written in the sequence script.
Source – foldoc.org
WMA - A file with the
.wma file extension is an audio recording
encoded with Microsoft's Windows Media
Audio Codec, and shows that it is an audio-only
file that can either be downloaded or
streamed. Using speech audio recorded
at 8khz sample rate or high-quality stereo
music files recorded at sample rates up
to 48kHz, Windows Audio Media files can
be encoded at bit rates as low as 5 kbps
up to Windows Media Players own 'near
CD quality' bit rate of 192 kbps. It is
claimed that Windows Media Audio provides
CD quality encoding at half the bit rate
of MP3.
Source – www.mp3machine.com/glossary
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