|
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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Direct Stream Digital
Direct Stream Digital (DSD) is a trademark used by Sony and Philips for their system for digitally encoding audio signals for the Super Audio CD (SACD). DSD uses pulse-density modulation encoding - a technology to store audio signals on digital storage media which are used for the SACD. The signal is stored as delta-sigma modulated digital audio, which is a sequence of single-bit values at a sampling rate of 2.8224 MHz (64 times the CD audio sampling rate of 44.1 kHz, but only at \tfrac of its 16-bit resolution). Noise shaping occurs by use of the 64-times oversampled signal to reduce noise and distortion caused by the inaccuracy of quantization of the audio signal to a single bit. Therefore, it is a topic of discussion whether it is possible to eliminate distortion in one-bit delta-sigma conversion. Development DSD is a method of storing a delta-sigma signal before applying a decimation process that converts the signal to a PCM signal. Delta-sigma conversi ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Fixed-point Arithmetic
In computing, fixed-point is a method of representing fractional (non-integer) numbers by storing a fixed number of digits of their fractional part. Dollar amounts, for example, are often stored with exactly two fractional digits, representing the cents (1/100 of dollar). More generally, the term may refer to representing fractional values as integer multiples of some fixed small unit, e.g. a fractional amount of hours as an integer multiple of ten-minute intervals. Fixed-point number representation is often contrasted to the more complicated and computationally demanding floating-point representation. In the fixed-point representation, the fraction is often expressed in the same number base as the integer part, but using negative powers of the base ''b''. The most common variants are decimal (base 10) and binary (base 2). The latter is commonly known also as binary scaling. Thus, if ''n'' fraction digits are stored, the value will always be an integer multiple of ''b ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Rockbox
Rockbox is a free and open-source software replacement for the OEM firmware in various forms of digital audio players (DAPs) with an original kernel. It offers an alternative to the player's operating system, in many cases without removing the original firmware, which provides a plug-in architecture for adding various enhancements and functions. Enhancements include personal digital assistant (PDA) functions, applications, utilities, and games. Rockbox can also retrofit video playback functions on players first released in mid-2000. Rockbox includes a voice-driven user-interface suitable for operation by visually impaired users. Rockbox runs on a wide variety of devices with very different hardware abilities: from early Archos players with 1-bit character cell-based displays, to modern players with high resolution color displays, digital optical audio hardware and advanced recording abilities. History The Rockbox project began in late 2001 and was first implemented on the ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio File Format
An audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data (excluding metadata) is called the audio coding format and can be uncompressed, or compressed to reduce the file size, often using lossy compression. The data can be a raw bitstream in an audio coding format, but it is usually embedded in a container format or an audio data format with defined storage layer. Format types It is important to distinguish between the audio coding format, the container containing the raw audio data, and an audio codec. A codec performs the encoding and decoding of the raw audio data while this encoded data is (usually) stored in a container file. Although most audio file formats support only one type of audio coding data (created with an audio coder), a multimedia container format (as Matroska or AVI) may support multiple types of audio and video data. There are three major groups of audio file formats: * Uncompressed au ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Lossless Compression
Lossless compression is a class of data compression that allows the original data to be perfectly reconstructed from the compressed data with no loss of information. Lossless compression is possible because most real-world data exhibits statistical redundancy. By contrast, lossy compression permits reconstruction only of an approximation of the original data, though usually with greatly improved compression rates (and therefore reduced media sizes). By operation of the pigeonhole principle, no lossless compression algorithm can efficiently compress all possible data. For this reason, many different algorithms exist that are designed either with a specific type of input data in mind or with specific assumptions about what kinds of redundancy the uncompressed data are likely to contain. Therefore, compression ratios tend to be stronger on human- and machine-readable documents and code in comparison to entropic binary data (random bytes). Lossless data compression is used in ma ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
OptimFROG DualStream
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 In information technology, lossy compression or irreversible compression is the class of data compression methods that uses inexact approximations and partial data discarding to represent the content. These techniques are used to reduce data size ... 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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
DeaDBeeF
DeaDBeeF is an audio player software available for Linux, Android and other Unix-like operating systems. DeaDBeeF is free and open-source software, except on Android. History The player was first published in August 2009. Its author cited dissatisfaction with existing music players under Linux as the main reason for writing DeaDBeeF. The name is a reference to magic number 0xDEADBEEF. Characteristics Among DeaDBeeF's functionalities are included: * Support for formats MP3, FLAC, APE, TTA, Vorbis, WavPack, Musepack, AAC, ALAC, WMA, WAV, DTS, audio CD, many forms of module files and music from game consoles. TAK and Opus are supported via ffmpeg/libav. * Cuesheet support, both in built-in format and external files. iso.wv support. * Character encodings Windows-1251 and ISO 8859-1 are supported in addition to UTF-8. * The program doesn't have any dependencies on GNOME, KDE or gstreamer. * Plug-in architecture. * Gapless playback. * Customizable systemd notifications ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Foobar2000
foobar2000 (often abbreviated as fb2k or f2k) is a freeware audio player for Microsoft Windows, iOS and Android developed by Peter Pawłowski. It has a modular design, which provides user flexibility in configuration and customization. Standard "skin" elements can be individually augmented or replaced with different dials and buttons, as well as visualizers such as waveform, oscilloscope, spectrum, spectrogram (waterfall), peak and smoothed VU meters. foobar2000 offers third-party user interface modifications through a software development kit (SDK). foobar2000 supports many audio file formats, has many features for organizing metadata, files, and folders, and has a converter interface for use with command line encoders. To maximize audio fidelity in cases where resampling or downscaling in bit depth is required, it provides noise shaping and dithering. There are a number of official and third-party components which add many additional features. The core is closed so ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-4 SLS
MPEG-4 SLS, or MPEG-4 Scalable to Lossless as per ISO/ IEC 14496-3:2005/Amd 3:2006 (Scalable Lossless Coding), is an extension to the MPEG-4 Part 3 (MPEG-4 Audio) standard to allow lossless audio compression scalable to lossy MPEG-4 General Audio coding methods (e.g., variations of AAC). It was developed jointly by the Institute for Infocomm Research (I2R) and Fraunhofer, which commercializes its implementation of a limited subset of the standard under the name of HD-AAC. Standardization of the HD-AAC profile for MPEG-4 Audio is under development (as of September 2009). MPEG-4 SLS allows having both a lossy layer and a lossless correction layer similar to Wavpack Hybrid, OptimFROG DualStream and DTS-HD Master Audio, providing backwards compatibility to MPEG AAC-compliant bitstreams. MPEG-4 SLS can also work without a lossy layer (a.k.a. "SLS Non-Core"), in which case it will not be backwards compatible, Lossy compression of files is necessary for files that need to be stre ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Psychoacoustics
Psychoacoustics is the branch of psychophysics involving the scientific study of sound perception and audiology—how humans perceive various sounds. More specifically, it is the branch of science studying the psychological responses associated with sound (including noise, speech, and music). Psychoacoustics is an interdisciplinary field of many areas, including psychology, acoustics, electronic engineering, physics, biology, physiology, and computer science. Background Hearing is not a purely mechanical phenomenon of wave propagation, but is also a sensory and perceptual event; in other words, when a person hears something, that something arrives at the ear as a mechanical sound wave traveling through the air, but within the ear it is transformed into neural action potentials. The outer hair cells (OHC) of a mammalian cochlea give rise to enhanced sensitivity and better frequency resolution of the mechanical response of the cochlear partition. These nerve pulses then trave ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Cue Sheet (music Software)
A cue sheet, or cue file, is a metadata file which describes how the tracks of a CD or DVD are laid out. Cue sheets are stored as plain text files and commonly have a filename extension. CDRWIN first introduced cue sheets, which are now supported by many optical disc authoring applications and media players. Overview Cue sheets can describe many types of audio and data CDs. The main data (including audio) for a CD described by a cue sheet is stored in one or more files referenced by the cue sheet. Cue sheets also specify track lengths and CD-Text including track and disc titles and performers. They are especially useful when dividing audio stored in a single file into multiple songs or tracks. The data files referred to by the cue sheet may be audio files (commonly in MP3 or WAV format), or plain disc images, usually with a extension. When used for disc images, the format is usually called CUE/BIN, indicating that it stores a disc image composed of one cue sheet file an ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
