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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 supports metadata tagging, album cover art, and fast seeking. History Development was started in 2000 by Josh Coalson. The bitstream format was frozen with the release of version 0.9 of the reference implementation on 31 March 2001. Version 1.0 was released on 20 July 2001. On 29 January 2003, the Xiph.Org Foundation and the FLAC project announced the incorporation of FLAC under the Xiph.org banner. ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
FLAC 1
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 supports metadata tagging, album cover art, and fast seeking. History Development was started in 2000 by Josh Coalson. The bitstream format was frozen with the release of version 0.9 of the reference implementation on 31 March 2001. Version 1.0 was released on 20 July 2001. On 29 January 2003, the Xiph.Org Foundation and the FLAC project announced the incorporation of FLAC under the Xiph.org ba ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Metadata
Metadata (or metainformation) is "data that provides information about other data", but not the content of the data itself, such as the text of a message or the image itself. There are many distinct types of metadata, including: * Descriptive metadata – the descriptive information about a resource. It is used for discovery and identification. It includes elements such as title, abstract, author, and keywords. * Structural metadata – metadata about containers of data and indicates how compound objects are put together, for example, how pages are ordered to form chapters. It describes the types, versions, relationships, and other characteristics of digital materials. * Administrative metadata – the information to help manage a resource, like resource type, and permissions, and when and how it was created. * Reference metadata – the information about the contents and quality of Statistical data type, statistical data. * Statistical metadata – also called process data, may ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Vorbis Comment
A Vorbis comment is a metadata container used in the Ogg file format (with Vorbis, FLAC, Theora, Speex and Opus codecs). 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, ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Bit Depth
In digital audio using pulse-code modulation (PCM), bit depth is the number of bits of information in each sample, and it directly corresponds to the resolution of each sample. Examples of bit depth include Compact Disc Digital Audio, which uses 16 bits per sample, and DVD-Audio and Blu-ray Disc, which can support up to 24 bits per sample. In basic implementations, variations in bit depth primarily affect the noise level from quantization error—thus the signal-to-noise ratio (SNR) and dynamic range. However, techniques such as dithering, noise shaping, and oversampling can mitigate these effects without changing the bit depth. Bit depth also affects bit rate and file size. Bit depth is useful for describing PCM digital signals. Non-PCM formats, such as those using lossy compression, do not have associated bit depths. Binary representation A PCM signal is a sequence of digital audio samples containing the data providing the necessary information to reconst ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sample Rate
In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a sound wave to a sequence of "samples". A sample is a value of the signal at a point in time and/or space; this definition differs from the term's usage in statistics, which refers to a set of such values. A sampler is a subsystem or operation that extracts samples from a continuous signal. A theoretical ideal sampler produces samples equivalent to the instantaneous value of the continuous signal at the desired points. The original signal can be reconstructed from a sequence of samples, up to the Nyquist limit, by passing the sequence of samples through a reconstruction filter. Theory Functions of space, time, or any other dimension can be sampled, and similarly in two or more dimensions. For functions that vary with time, let s(t) be a continuous function (or "signal") to be sampled, and let sampling be performed by measuring ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sync Word
In computer networks, a syncword, sync character, sync sequence or preamble is used to synchronize a data transmission by indicating the end of header information and the start of data. The syncword is a known sequence of data used to identify the start of a frame, and is also called ''reference signal'' or ''midamble'' in wireless communications. Prefix codes allow unambiguous identification of synchronization sequences and may serve as self-synchronizing code. Examples In an audio receiver receiving a bit stream of data, an example of a syncword is 0x0B77 for an AC-3 encoded stream. An Ethernet packet with the Ethernet preamble, 56 bits of alternating 1 and 0 bits, allowing the receiver to synchronize its clock to the transmitter, followed by a one-octet start frame delimiter byte and then the header. All USB packets begin with a sync field (8 bits long at low speed, 32 bits long at high speed) used to synchronize the receiver's clock to the transmitter's clock. A ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
CRC16
A cyclic redundancy check (CRC) is an error-detecting code commonly used in digital networks and storage devices to detect accidental changes to digital data. Blocks of data entering these systems get a short ''check value'' attached, based on the remainder of a polynomial division of their contents. On retrieval, the calculation is repeated and, in the event the check values do not match, corrective action can be taken against data corruption. CRCs can be used for error correction (see bitfilters). CRCs are so called because the ''check'' (data verification) value is a ''redundancy'' (it expands the message without adding information) and the algorithm is based on ''cyclic'' codes. CRCs are popular because they are simple to implement in binary hardware, easy to analyze mathematically, and particularly good at detecting common errors caused by noise in transmission channels. Because the check value has a fixed length, the function that generates it is occasionally used as a h ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
File Format
A file format is a Computer standard, standard way that information is encoded for storage in a computer file. It specifies how bits are used to encode information in a digital storage medium. File formats may be either proprietary format, proprietary or open format, open. Some file formats are designed for very particular types of data: Portable Network Graphics, PNG files, for example, store Raster graphics, bitmapped Graphics file format, images using lossless data compression. Other file formats, however, are designed for storage of several different types of data: the Ogg format can act as a container format (digital), container for different types of multimedia including any combination of sound, audio and video, with or without text (such as subtitles), and metadata. A text file can contain any stream of characters, including possible control characters, and is encoded in one of various Character encoding, character encoding schemes. Some file formats, such as HTML, sca ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Linear Pulse-code Modulation
Pulse-code modulation (PCM) is a method used to digitally represent analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Alec Reeves, Claude Shannon, Barney Oliver and John R. Pierce are credited with its invention. Linear pulse-code modulation (LPCM) is a specific type of PCM in which the quantization levels are linearly uniform. This is in contrast to PCM encodings in which quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm). Though ''PCM'' is a more general term, it is often used to describe data encoded as LPCM. A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of times ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Opus (codec)
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. 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, networked music performances, and live lip sync; by trading off quality or bitrate, the delay can be reduced down to 5 ms. Its delay is exceptionally low compared to competing codecs, which ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speex
{{More citations needed, date=May 2025 The Speex project is an attempt to create a free software speech codec, unencumbered by patent restrictions. Speex is licensed under the BSD License and is used with the Xiph.org Foundation's Ogg container format. The Speex coder uses the Ogg bitstream format, and the Speex designers see their project as complementary to the Vorbis general-purpose audio compression project. Description Unlike many other speech codecs, Speex is not targeted at cell phones but rather at voice over IP (VoIP) and file-based compression. The design goals have been to make a codec that would allow both very good quality speech and low bit rate, which led to the development of a codec with multiple bit rates. Very good quality also meant the support of wideband (16 kHz sampling rate) in addition to narrowband (telephone quality, 8 kHz sampling rate). Designing for VoIP instead of cell phone use means that Speex must be robust to lost packets, but not to cor ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Theora
Theora is a free lossy video compression format. It was developed by the Xiph.Org Foundation and distributed without licensing fees alongside their other free and open media projects, including the Vorbis audio format and the Ogg container. The libtheora video codec is the reference implementation of the Theora video compression format developed by the Xiph.Org Foundation. Theora was derived from the formerly proprietary VP3 codec, released into the public domain by On2 Technologies. It is broadly comparable in design and bitrate efficiency to MPEG-4 Part 2, early versions of Windows Media Video, and RealVideo while it lacked some of the features present in some of these other codecs. It is comparable in open standards philosophy to the BBC's Dirac codec. Theora was named after Theora Jones, Edison Carter's Controller on the '' Max Headroom'' television program. Technical details Theora is a variable-bitrate, DCT-based video compression scheme. Like most ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
