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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] |
MP3Gain
MP3Gain is an audio normalization software tool. The tool is available on multiple platforms and is free software. It analyzes the MP3 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. Origin ... and reversibly changes its volume. The volume can be adjusted for single files or as album where all files would have the same perceived loudness. It is an implementation of ReplayGain. In 2015 Debian and Ubuntu removed it from their repositories due to a lack of an active maintainer. MP3Gain Technical details MP3Gain first computes the desired gain (volume adjustment), either per track or per album, using the ReplayGain algorithm. It then modifies the overall volume scale factor in each MP3 frame, and writes undo information as a tag (in APEv2, or ID3v2 format) making this a reversible proc ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Normalization
Audio normalization is the application of a constant amount of gain to an audio recording to bring the amplitude to a target level (the norm). Because the same amount of gain is applied across the entire recording, the signal-to-noise ratio and relative dynamics are unchanged. Normalization is one of the functions commonly provided by a digital audio workstation. Two principal types of audio normalization exist. Peak normalization adjusts the recording based on the highest signal level present in the recording. Loudness normalization adjusts the recording based on perceived loudness. Normalization differs from dynamic range compression, which applies varying levels of gain over a recording to fit the level within a minimum and maximum range. Normalization adjusts the gain by a constant value across the entire recording. Peak normalization One type of normalization is peak normalization, wherein the gain is changed to bring the highest PCM sample value or analog signal peak ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Loudness
In acoustics, loudness is the subjective perception of sound pressure. More formally, it is defined as, "That attribute of auditory sensation in terms of which sounds can be ordered on a scale extending from quiet to loud". The relation of physical attributes of sound to perceived loudness consists of physical, physiological and psychological components. The study of apparent loudness is included in the topic of psychoacoustics and employs methods of psychophysics. In different industries, loudness may have different meanings and different measurement standards. Some definitions, such as ITU-R BS.1770 refer to the relative loudness of different segments of electronically reproduced sounds, such as for broadcasting and cinema. Others, such as ISO 532A (Stevens loudness, measured in sones), ISO 532B ( Zwicker loudness), DIN 45631 and ASA/ANSI S3.4, have a more general scope and are often used to characterize loudness of environmental noise. More modern standards, such as Nordt ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
FLAC
FLAC (; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, developed by the Xiph.Org Foundation, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to between 50 and 70 percent of its original size and decompresses to an identical copy of the original audio data. FLAC is an open format with royalty-free licensing and a reference implementation which is free software. FLAC has support for metadata tagging, album cover art, and fast seeking. History Development was started in 2000 by Josh Coalson. The bit-stream format was frozen when FLAC entered beta stage with the release of version 0.5 of the reference implementation on 15 January 2001. Version 1.0 was released on 20 July 2001. On 29 January 2003, the Xiph.Org Foundation and the FLAC project announced the ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Vorbis
Vorbis is a free and open-source software project headed by the Xiph.Org Foundation. The project produces an audio coding format and software reference encoder/decoder (codec) for lossy audio compression. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis. Vorbis is a continuation of audio compression development started in 1993 by Chris Montgomery. Intensive development began following a September 1998 letter from the Fraunhofer Society announcing plans to charge licensing fees for the MP3 audio format. The Vorbis project started as part of the Xiphophorus company's Ogg project (also known as OggSquish multimedia project). Chris Montgomery began work on the project and was assisted by a growing number of other developers. They continued refining the source code until the Vorbis file format was frozen for 1.0 in May 2000. Originally licensed as LGPL, in 2001 the Vorbis license was changed to the BSD ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Vorbis Comment
A Vorbis comment is a metadata container used in the Vorbis, FLAC, Theora, Speex and Opus file formats. It allows information such as the title, artist, album, track number or other information about the file to be added to the file itself. However, as the official Ogg Vorbis documentation notes, “ he comment headeris meant for short, text comments, not arbitrary metadata; arbitrary metadata belongs in a separate logical bitstream (usually an XML stream type) that provides greater structure and machine parseability.” Instead, the intended function of Vorbis comments is to approximate the kind of information that might be hand-inked onto a blank faced CD-R or CD-RW: a few lines of notes briefly detailing the content. Format A Vorbis tag is a list of fields in the format FieldName=Data. The field name can be composed of printable ASCII characters, 0x20 (''space'') through 0x7D (‘}’), with 0x3D (‘=’) and 0x7E (‘~’) excluded. It is case insensitive, so artist and ARTIST ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
ID3v2
ID3 is a metadata container most often used in conjunction with the MP3 audio file format. It allows information such as the title, artist, album, track number, and other information about the file to be stored in the file itself. There are two unrelated versions of ID3: ID3v1 and ID3v2. ID3v1 takes the form of a 128-byte segment at the end of an MP3 file containing a fixed set of data fields. ID3v1.1 is a slight modification which adds a "track number" field at the expense of a slight shortening of the "comment" field. ID3v2 is structurally very different from ID3v1, consisting of an extensible set of "frames" located at the start of the file, each with a frame identifier (a three- or four-byte string) and one piece of data. 83 types of frames are declared in the ID3v2.4 specification, and applications can also define their own types. There are standard frames for containing cover art, Beats per minute, BPM, copyright and license, lyrics, and arbitrary text and URL data, as well ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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] |
Windows Media Audio
Windows Media Audio (WMA) is a series of audio codecs and their corresponding audio coding formats developed by Microsoft. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as ''WMA'', was conceived as a competitor to the popular MP3 and RealAudio codecs. ''WMA Pro'', a newer and more advanced codec, supports multichannel and high resolution audio. A lossless codec, ''WMA Lossless'', compresses audio data without loss of audio fidelity (the regular WMA format is lossy). ''WMA Voice'', targeted at voice content, applies compression using a range of low bit rates. Microsoft has also developed a digital container format called Advanced Systems Format to store audio encoded by WMA. Development history The first WMA codec was based on earlier work by Henrique Malvar and his team which was transferred to the Windows Media team at Microsoft. Malvar was a senior researcher and ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Technical Standard
A technical standard is an established norm or requirement for a repeatable technical task which is applied to a common and repeated use of rules, conditions, guidelines or characteristics for products or related processes and production methods, and related management systems practices. A technical standard includes definition of terms; classification of components; delineation of procedures; specification of dimensions, materials, performance, designs, or operations; measurement of quality and quantity in describing materials, processes, products, systems, services, or practices; test methods and sampling procedures; or descriptions of fit and measurements of size or strength. It is usually a formal document that establishes uniform engineering or technical criteria, methods, processes, and practices. In contrast, a custom, convention, company product, corporate standard, and so forth that becomes generally accepted and dominant is often called a ''de facto'' standard. A techni ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
DBSPL
Sound pressure or acoustic pressure is the local pressure deviation from the ambient (average or equilibrium) atmospheric pressure, caused by a sound wave. In air, sound pressure can be measured using a microphone, and in water with a hydrophone. The SI unit of sound pressure is the pascal (Pa). Mathematical definition A sound wave in a transmission medium causes a deviation (sound pressure, a ''dynamic'' pressure) in the local ambient pressure, a ''static'' pressure. Sound pressure, denoted ''p'', is defined by p_\text = p_\text + p, where * ''p''total is the total pressure, * ''p''stat is the static pressure. Sound measurements Sound intensity In a sound wave, the complementary variable to sound pressure is the particle velocity. Together, they determine the sound intensity of the wave. ''Sound intensity'', denoted I and measured in W· m−2 in SI units, is defined by \mathbf I = p \mathbf v, where * ''p'' is the sound pressure, * v is the particle velocity. Acoust ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Tag (metadata)
In information systems, a tag is a keyword or term assigned to a piece of information (such as an Internet bookmark, multimedia, database record, or computer file). This kind of metadata helps describe an item and allows it to be found again by browsing or searching. Tags are generally chosen informally and personally by the item's creator or by its viewer, depending on the system, although they may also be chosen from a controlled vocabulary. Tagging was popularized by websites associated with Web 2.0 and is an important feature of many Web 2.0 services. It is now also part of other database systems, desktop applications, and operating systems. Overview People use tags to aid classification, mark ownership, note boundaries, and indicate online identity. Tags may take the form of words, images, or other identifying marks. An analogous example of tags in the physical world is museum object tagging. People were using textual keywords to classify information and objects ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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