AIMP
AIMP (Artem Izmaylov Media Player) is a freeware audio player for Windows and Android, originally developed by Russian developer Artem Izmaylov ( rus, Артём Измайлов, Artyom Izmajlov).AIMP FAQs Retrieved September 24, 2011. Development The first version of AIMP was released on August 8, 2006. AIMP was initially based on the BASS audio library. Version 3 added a new audio engine and full support for , and revamped the music library interface transparency effects.Features AIMP supports audio codecs, including[...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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ReplayGain
ReplayGain is a proposed technical standard published by David Robinson in 2001 to measure and normalize the perceived loudness of audio in computer audio formats such as MP3 and Ogg Vorbis. It allows media players to normalize loudness for individual tracks or albums. This avoids the common problem of having to manually adjust volume levels between tracks when playing audio files from albums that have been mastered at different loudness levels. Although this de facto standard is now formally known as ReplayGain, it was originally known as Replay Gain and is sometimes abbreviated RG. ReplayGain is supported in a large number of media software and portable devices. Operation ReplayGain works by first performing a psychoacoustic analysis of an entire audio track or album to measure peak level and perceived loudness. Equal-loudness contours are used to compensate for frequency effects and statistical analysis is used to accommodate for effects related to time. The difference bet ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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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. Opus combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed for maximal efficiency. Bitrate, audio bandwidth, complexity, and algorithm can all be adjusted seamlessly in each frame. Opus has the low algorithmic delay (26.5 ms by default) necessary for use as part of a real-time communication link, networke ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Audio Player (software)
Media player software is a type of application software for playing multimedia computer files 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 (⏩️), backforward (⏪), 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. For example, Windows comes with Windows Media Player, Microsoft Movies & TV and Groove Music, while macOS comes with QuickTime Player and Music. Linux distributions come with different media players, such as SMPlayer, Amarok, Audacious, Banshee, MPlayer, mpv, Rhythmbox, Totem, VLC media player, and xine. Android comes with Google Play Music for audio and Google Photos for video, ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MPEG-1 Audio Layer III
MP3 (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, 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. A third version, known as MPEG 2.5 — extended to better support lower bit rates — is commonly implemented, but is not a recognized standard. MP3 (or mp3) as a file format commonly designates files containing an elementary stream of MPEG-1 Audio or MPEG-2 Audio encoded data, without other complexities of the MP3 standard. With regard to audio compression (the aspect of the standard most apparent to end-users, and for which it is best known), MP3 uses lossy data-compression to encode data using inexact ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Advanced Audio Coding
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. Designed to be the successor of the MP3 format, AAC generally achieves higher sound quality than MP3 encoders at the same bit rate. AAC has been standardized by ISO and IEC as part of the MPEG-2 and MPEG-4 specifications.ISO (2006ISO/IEC 13818-7:2006 - Information technology -- Generic coding of moving pictures and associated audio information -- Part 7: Advanced Audio Coding (AAC), Retrieved on 2009-08-06ISO (2006, Retrieved on 2009-08-06 Part of AAC, HE-AAC ("AAC+"), is part of MPEG-4 Audio and is adopted into digital radio standards DAB+ and Digital Radio Mondiale, and mobile television standards DVB-H and ATSC-M/H. AAC supports inclusion of 48 full-bandwidth (up to 96 kHz) audio channels in one stream plus 16 low frequency effects ( LFE, limited to 120 Hz) channels, up to 16 "coupling" or dialog channels, and up to 16 data streams. The quality for stereo is satisf ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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TTA (codec)
True Audio (TTA) is a lossless compressor for multichannel 8, 16 and 24 bits audio data. .tta is the extension to filenames of audio files created by the True Audio codec. Codec overview True Audio compresses up to 30% of the original, broadly similar to FLAC and APE. It features a real-time encoding and decoding algorithm and hardware compression support. As with most other lossless codecs, plugins are available for most media players. TTA performs lossless compression on multichannel 8, 16 and 24 bit The bit is the most basic unit of information in computing and digital communications. The name is a portmanteau of binary digit. The bit represents a logical state with one of two possible values. These values are most commonly represente ... data of uncompressed wav input files. The term "lossless" refers to the fact that such compression results in no data or quality loss; when decompressed, the audio file data are bit-identical to those of their originals ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MPEG-1 Audio Layer II
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM) is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting. History of development from MP2 to MP3 MUSICAM MPEG-1 Audio Layer 2 encoding was derived from the MUSICAM (''Masking pattern adapted Universal Subband Integrated Coding And Multiplexing'') audio codec, developed by Centre commun d'études de télévision et télécommunications (CCETT), Philips, and the Institut für Rundfunktechnik (IRT) in 1989 as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of a system for the broadcasting of audio and data to fixed, portable or mobile receivers (established in 1987). It began as the Digital Audio Broadcast (DAB) project manage ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Musepack
Musepack or MPC is an open source lossy audio codec, specifically optimized for transparent compression of stereo audio at bitrates of 160–180 (manual set allows bitrates up to 320) kbit/s. It was formerly known as MPEGplus, MPEG+ or MP+. Development of MPC was initiated in 1997 by Andree Buschmann and later assumed by Frank Klemm, and as of 2004 is maintained by the Musepack Development Team (MDT) with assistance from Buschmann and Klemm. Encoders and decoders are available for Microsoft Windows, Linux and Mac OS X, and plugins for several third-party media players available from the Musepack website, licensed under the GNU Lesser General Public License (LGPL) or BSD licenses, and an extensive list of programs supporting the format. Technical details Musepack was developed using the MP2 codec as a starting point, but many features have since been added, including: * subband selectable M/S encoding (as in AAC) * Huffman coding (as in MP3 and AAC, but more efficient). S ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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OptimFROG
OptimFROG is a proprietary lossless audio data compression codec developed by Florin Ghido. OptimFROG is optimized for very high compression (small file sizes) at the expense of encoding and decoding speed, and consistently measures among the highest compressing lossless codecs. OptimFROG comes with three compressors: one lossless codec for integer LPCM format WAVE files, one for IEEE_754 floating-point WAVE files, and a ''DualStream'' format with a lossy part and a correction file for losslessness. OptimFROG DualStream OptimFROG DualStream is a lossy codec, aimed to fill the gap between perceptual coding and lossless coding as OptimFROG DualStream has an option to produce a correction file. This file can be used, in combination with the main lossy-encoded file, for lossless decoding, but not, unlike Wavpack hybrid for instance, for playback. This correction feature is also offered by MPEG-4 SLS and DTS-HD Master Audio. Technical details Metadata The OptimFROG file formats use ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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WavPack
WavPack is a free and open-source lossless audio compression format and application implementing the format. It is unique in the way that it supports hybrid audio compression alongside normal compression which is similar to how FLAC works. It also supports compressing a wide variety of lossless formats, including various variants of PCM and also DSD as used in SACDs, together with its support for surround audio. Features WavPack compression can compress (and restore) 8-, 16-, 24-, and 32-bit fixed-point, and 32-bit floating-point PCM audio files in the .WAV file format. It also supports surround sound streams and high sampling rates. Like other lossless compression schemes, the data reduction rate varies with the source, but it is generally between 30% and 70% for typical popular music and somewhat better than that for classical music and other sources with greater dynamic range. Hybrid mode WavPack also incorporates a "hybrid" mode, which still provides the features of ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MOD (file Format)
MOD is a computer file format used primarily to represent music, and was the first module file format. MOD files use the “.MOD” file extension, except on the Amiga which doesn't rely on filename extensions; instead, it reads a file's header to determine filetype. A MOD file contains a set of ''instruments'' in the form of samples, a number of ''patterns'' indicating how and when the samples are to be played, and a list of what patterns to play in what order. History The first version of the format was created by Karsten Obarski for use in the Ultimate Soundtracker, tracker software released for the Amiga computer in 1987. The format has since been supported by hundreds of playback programs and dozens of other trackers. The original version of the MOD format featured four channels of simultaneous audio playback, corresponding to the capabilities of the original Amiga chipset, and up to 15 instruments. Later variations of the format have extended this to up to 32 chann ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |