Variable bitrateVariable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding. As opposed to constant bitrate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bitrate (and therefore more storage space) to be allocated to the more complex segments of media files while less space is allocated to less complex segments. The average of these rates can be calculated to produce an average bitrate for the file.
Portable media playerA portable media player (PMP) (also including the related digital audio player (DAP)) is a portable consumer electronics device capable of storing and playing digital media such as audio, images, and video files. The data is typically stored on a compact disc (CD), Digital Versatile Disc (DVD), Blu-ray Disc (BD), flash memory, microdrive, SD cards or hard drive; most earlier PMPs used physical media, but modern players mostly use flash memory.
Media player softwareMedia player software is a type of application software for playing multimedia s like audio and video files. Media players commonly display standard media control icons known from physical devices such as tape recorders and CD players, such as play ( ), pause ( ), fastforward (⏩️), rewind (⏪), and stop ( ) buttons. In addition, they generally have progress bars (or "playback bars"), which are sliders to locate the current position in the duration of the media file. Mainstream operating systems have at least one default media player.
MP3MP3 (formally MPEG-1 Audio Layer III or MPEG-2 Audio Layer III) is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg, with support from other digital scientists in the United States and elsewhere. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended — defining additional bit-rates and support for more audio channels — as the third audio format of the subsequent MPEG-2 standard.
QuickTimeQuickTime is a discontinued extensible multimedia architecture created by Apple, which supports playing, streaming, encoding, and transcoding a variety of digital media formats. The term QuickTime also refers to the QuickTime Player front-end media player application, which is built-into macOS, and was available for download on Windows until 2016. QuickTime was created in 1991, when the concept of playing digital video directly on computers was "groundbreaking.
Opus (audio format)Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.
WinampWinamp is a media player for Microsoft Windows originally developed by Justin Frankel and Dmitry Boldyrev by their company Nullsoft, which they later sold to AOL in 1999 for $80 million. It was then acquired by Radionomy in 2014. Since version 2 it has been sold as freemium and supports extensibility with plug-ins and skins, and features music visualization, playlist and a media library, supported by a large online community. Version 1 of Winamp was released in 1997, and quickly grew popular with over 3 million downloads, paralleling the developing trend of MP3 (music) .
Digital audioDigital audio is a representation of sound recorded in, or converted into, digital form. In digital audio, the sound wave of the audio signal is typically encoded as numerical samples in a continuous sequence. For example, in CD audio, samples are taken 44,100 times per second, each with 16-bit sample depth. Digital audio is also the name for the entire technology of sound recording and reproduction using audio signals that have been encoded in digital form.
WAVWaveform Audio File Format (WAVE, or WAV due to its ; pronounced "wave" or "wæv" ) is an standard, developed by IBM and Microsoft, for storing an audio bitstream on personal computers. It is the main format used on Microsoft Windows systems for uncompressed audio. The usual bitstream encoding is the linear pulse-code modulation (LPCM) format. WAV is an application of the (RIFF) bitstream format method for storing data in chunks, and thus is similar to the 8SVX and the (AIFF) format used on Amiga and Macintosh computers, respectively.
MPEG-1MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s (26:1 and 6:1 compression ratios respectively) without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical. Today, MPEG-1 has become the most widely compatible lossy audio/video format in the world, and is used in a large number of products and technologies.
MPEG-4MPEG-4 is a group of international standards for the compression of digital audio and visual data, multimedia systems, and file storage formats. It was originally introduced in late 1998 as a group of audio and video coding formats and related technology agreed upon by the ISO/IEC Moving Picture Experts Group (MPEG) (ISO/IEC JTC 1/SC29/WG11) under the formal standard ISO/IEC 14496 – Coding of audio-visual objects. Uses of MPEG-4 include compression of audiovisual data for Internet video and CD distribution, voice (telephone, videophone) and broadcast television applications.
Audio coding formatAn audio coding format (or sometimes audio compression format) is a content representation format for storage or transmission of digital audio (such as in digital television, digital radio and in audio and video files). Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.
FFmpegFFmpeg is a free and open-source software project consisting of a suite of libraries and programs for handling video, audio, and other multimedia files and streams. At its core is the command-line ffmpeg tool itself, designed for processing of video and audio files. It is widely used for format transcoding, basic editing (trimming and concatenation), video scaling, video post-production effects and standards compliance (SMPTE, ITU). FFmpeg also includes other tools: ffplay, a simple media player and ffprobe, a command-line tool to display media information.
MPEG-2MPEG-2 (a.k.a. H.222/H.262 as was defined by the ITU) is a standard for "the generic coding of moving pictures and associated audio information". It describes a combination of lossy video compression and lossy audio data compression methods, which permit storage and transmission of movies using currently available storage media and transmission bandwidth. While MPEG-2 is not as efficient as newer standards such as H.264/AVC and H.
Windows Media PlayerWindows Media Player (WMP) is the first media player and media library application that Microsoft developed to play audio and video on personal computers. It has been a component of the Microsoft Windows operating system, including Windows 9x, Windows NT, Pocket PC, and Windows Mobile. Microsoft also released editions of Windows Media Player for classic Mac OS, Mac OS X, and Solaris, but has since discontinued them.
Data compression ratioData compression ratio, also known as compression power, is a measurement of the relative reduction in size of data representation produced by a data compression algorithm. It is typically expressed as the division of uncompressed size by compressed size. Data compression ratio is defined as the ratio between the uncompressed size and compressed size: Thus, a representation that compresses a file's storage size from 10 MB to 2 MB has a compression ratio of 10/2 = 5, often notated as an explicit ratio, 5:1 (read "five" to "one"), or as an implicit ratio, 5/1.
DVBDigital Video Broadcasting (DVB) is a set of international open standards for digital television. DVB standards are maintained by the DVB Project, an international industry consortium, and are published by a Joint Technical Committee (JTC) of the European Telecommunications Standards Institute (ETSI), European Committee for Electrotechnical Standardization (CENELEC) and European Broadcasting Union (EBU).
Digital Audio BroadcastingDigital Audio Broadcasting (DAB) is a digital radio standard for broadcasting digital audio radio services in many countries around the world, defined, supported, marketed and promoted by the WorldDAB organisation. The standard is dominant in Europe and is also used in Australia, and in parts of Africa and Asia; as of 2022, 55 countries are actively running DAB broadcasts. DAB was the result of a European research project and first publicly rolled out in 1995, with consumer-grade DAB receivers appearing at the start of this millennium.
Streaming mediaStreaming media is multimedia that is delivered and consumed in a continuous manner from a source, with little or no intermediate storage in network elements. Streaming refers to the delivery method of content, rather than the content itself. Distinguishing delivery method from the media applies specifically to telecommunications networks, as most of the traditional media delivery systems are either inherently streaming (e.g. radio, television) or inherently non-streaming (e.g. books, videotapes, audio CDs).
Transform codingTransform coding is a type of data compression for "natural" data like audio signals or photographic s. The transformation is typically lossless (perfectly reversible) on its own but is used to enable better (more targeted) quantization, which then results in a lower quality copy of the original input (lossy compression). In transform coding, knowledge of the application is used to choose information to discard, thereby lowering its bandwidth. The remaining information can then be compressed via a variety of methods.
