ADC
Analog to Digital Converter. A process that changes a wave form (analog audio) into a series of ones and zeros (digital audio). The computer needs to do this so that it can 'understand' and manipulate audio. See also DAC, Analog Audio, and Digital Audio.
Administrator
ALSA
Advanced Linux Sound Architecture. The part of the operating system that enables sound under Linux.
Audio Source
The source of an audio signal. For example if we plug a CD player into the sound card, then the CD player is referred to as the audio source. However source can also refer to the source code of a computer program.
Bitrate
Bit rate (sometimes written bitrate) is the frequency at which bits are passing a given (physical or metaphorical) "point". It is quantified using the bit per second (bit/s) unit. In the process of sampling digital audio, the bitrate influences the quality of the sample. Common bitrates used for digital audio are:
- 4 kbit/s - minimum necessary for recognizable speech (using special-purpose speech codecs)
- 8 kbit/s - telephone quality
- 32 kbit/s - MW quality
- 96 kbit/s - FM quality
- 128 kbit/s - CD quality
- 196 kbit/s - so-called "transparence", i.e. the quality at which most listeners can't tell the difference between compressed and uncompressed audio
Buffer
In computing, a buffer is a portion of memory set aside to store data, often before it is sent to an external device or as it is received from an external device. Streaming audio players store several seconds worth of data in a buffer before beginning playback. The buffer absorbs the bursts of data as they are delivered by the Internet and releases it at a constant rate for smooth playback.
Client
In audio streaming when we talk about a client we refer to the computer that is listening to a stream. If I have 50 computers listening to my stream I can also say there are "50 clients". Note: a client is a computer which is different from a user (which usually refers to a person).
Codec
Short for Compress DECompress. The codec is an algorithm that manages the reduction of audio and video data so it can be sent (streamed) over the Internet.
Most codecs are implemented as libraries which interface to one or more multimedia players, such as XMMS or Winamp. Codecs can perform lossy or lossless data compression. In the first case, psychoacustic models are used to eliminate from the original sound those passages and frequencies which the human ear isn't able to perceive; in the second case, all the sound data is preserved while still attaining a certain level of compression. Some popular lossy audio compression codecs are: MP3 (MPEG audio layer-3), WMA (Windows Media Audio), Vorbis and Speex. It's important to distinguish file format and codec; even though most audio file formats support only one audio codec, a file format may support multiple codecs, like AVI and OGG do. Examples of compressed audio file formats are MP3, WMA, and OGG.
Compression
Audio data compression is a process in which the amount of data in a recorded waveform is reduced for transmission. The purpose of audio compression is to reduce the size of audio data files which are stored on a physical medium, or the size of a stream which is played back over a network connection. Audio compression algorithms are typically referred to as audio codecs.
DAC
Digital Analog Converter. A piece of hardware or software that converts digital audio to analog audio.
Decode
The process of converting encoded information (data) into a human-readable format. In streaming this usually refers to the player software decoding the streaming audio or video so that the content can be seen or heard.
The transfer speed (usually that of an Internet connection) by which data can be sent from the server to the client. The process by which downstream is utilized is known as downloading.
Encode
In streaming this usually refers to the process of converting human readable information into data for transportation (streaming) over the Internet. The software that does this is usually called the encoder.
Encoder
The software that encodes audio or video for sending (streaming) over the Internet.
A firewall is a piece of hardware and/or software which functions in a network, with own rules to prevent unwanted communication protocols.
Host
The computer that is running the software that you need to use. If the computer is streaming server located on the Internet then when asked for the host or hostname you must supply an IP Number or URL of the computer.
Hostname
The IP Number or URL of the host machine.
Icecast
An audio streaming media server software. Icecast2 is the version of the Icecast server that supports MP3 AND OGG Vorbis streams.
IP
Internet Protocol
Localhost
Local Network
A localised network of computers, usually not accessible by computers outside its own network.
A file format for audio.
An open source file format for audio.
OSS
Open Sound System. The original unified sound processing component of UNIX and linux systems.
Real Audio. A file format for audio.
Sampling
To be treated by computers and digital machines in general, audio data has to be digitized, i.e. the analog sound waves have to be converted into a digital format understood by computers. This is done by sampling, i.e. the process where the instantaneous amplitude of an analog signal is measured many times per second (exactly how many times is specified by the sampling rate) and converted into a digital stream of bits.
Sampling Rate
In digital audio, the sampling rate is the Number of snapshots (samples) taken of a particular signal in one second during the process of sampling. Higher sampling rates result in more snapshots of the original signal. If a sufficient number of pieces is taken, they "meld" together to form a very close approximation of the original signal. The sampling rate is usually measured in Hertz, i.e. number of samples taken per second.
Common sampling rates for digital audio are:
- 8,000 Hz - telephone, adequate for human speech
- 22,050 Hz - radio
- 32,000 Hz - miniDV digital video camcorder
- 44,100 Hz (44.1 kHz) - compact disc, also most commonly used with MPEG-1 audio (VCD, SVCD, MP3)
- 47,250 Hz - world's first commercial PCM sound recorder by Nippon Columbia (Denon)
- 48,000 Hz - digital sound used for digital TV, DVD, films and professional audio
- 96,000 or 192,400 Hz - DVD-Audio, some LPCM DVD audio tracks, BD-ROM (Blu-ray Disc) audio tracks, and HD-DVD (High-Definition DVD) audio tracks
The number of bits used to represent each sample (often called "Bits per sample") also influences the quality of digital audio; more bits allow for a wider range of samples, which produces a thinner resolution in the representation of the original sound. Usually 8, 16 or 32 bits per sample are used.
Server
A combination of hardware and software that allows for the delivery over the Internet of a specific kind of content (service).
Stream
The audio or video content that is being delivered over the Internet.
Streaming Media
The whole universe of technologies that enables the delivery of audio and video over the Internet.
Superuser
The generic name for the user that has complete control over your linux/UNIX system. Sometimes also referred to as the administrator.
Terminal
The application you use to execute command under UNIX or linux.
The transfer speed (usually that of an Internet connection) by which data can be sent from the client to the server. The process by which upstream is utilized is known as uploading.
User
The account on a computer that you use to access the computer.
An audio file format.
An audio file format.