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Transcode
Transcoding is the direct digital-to-digital conversion of one encoding to another, such as for video data files, audio files (e.g., MP3, WAV), or character encoding (e.g., UTF-8, ISO/IEC 8859). This is usually done in cases where a target device (or workflow) does not support the format or has limited storage capacity that mandates a reduced file size, "Advancements in Compression and Transcoding: 2008 and Beyond", Society of Motion Picture and Television Engineers (SMPTE), 2008, webpageSMPTE-spm or to convert incompatible or obsolete data to a better-supported or modern format. In the analog video world, transcoding can be performed just while files are being searched, as well as for presentation. For example, Cineon and DPX files have been widely used as a common format for digital cinema, but the data size of a two-hour movie is about 8 terabytes (TB). That large size can increase the cost and difficulty of handling movie files. However, transcoding into a JPEG20 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Data Compression
In information theory, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compression reduces bits by identifying and eliminating statistical redundancy. No information is lost in lossless compression. Lossy compression reduces bits by removing unnecessary or less important information. Typically, a device that performs data compression is referred to as an encoder, and one that performs the reversal of the process (decompression) as a decoder. The process of reducing the size of a data file is often referred to as data compression. In the context of data transmission, it is called source coding; encoding done at the source of the data before it is stored or transmitted. Source coding should not be confused with channel coding, for error detection and correction or line coding, the means for mapping data onto a signal. ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sample Rate Conversion
Sample-rate conversion, sampling-frequency conversion or resampling is the process of changing the sampling rate or sampling frequency of a discrete signal to obtain a new discrete representation of the underlying continuous signal. Application areas include image scaling and audio/visual systems, where different sampling rates may be used for engineering, economic, or historical reasons. For example, Compact Disc Digital Audio and Digital Audio Tape systems use different sampling rates, and American television, European television, and movies all use different frame rates. Sample-rate conversion prevents changes in speed and pitch that would otherwise occur when transferring recorded material between such systems. More specific types of resampling include: '' upsampling'' or ''upscaling''; '' downsampling'', ''downscaling'', or ''decimation''; and '' interpolation''. The term multi-rate digital signal processing is sometimes used to refer to systems that incorporate sample ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Audio Interchange File Format
Audio Interchange File Format (AIFF) is an audio file format standard used for storing sound data for personal computers and other electronic audio devices. The format was developed by Apple Inc. in 1988 based on Electronic Arts' Interchange File Format (IFF, widely used on Amiga systems) and is most commonly used on Apple Macintosh computer systems. The audio data in most AIFF files is uncompressed pulse-code modulation (PCM). This type of AIFF file uses much more disk space than lossy formats like MP3—about 10 MB for one minute of stereo audio at a sample rate of 44.1 kHz and a bit depth of 16 bits. There is also a compressed variant of AIFF known as AIFF-C or AIFC, with various defined compression codecs. In addition to audio data, AIFF can include loop point data and the musical note of a sample, for use by hardware samplers and musical applications. The file extension for the standard AIFF format is .aiff or .aif. For the compressed variants it is supposed to be .ai ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Pulse-code Modulation
Pulse-code modulation (PCM) is a method used to digitally represent sampled 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 regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. 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 per second that samples are taken; and the bit depth, which determines ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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 l ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
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 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. Compression ratios achieved by the TTA codec vary, depending on music type, but range from 30% to 70% of the original. The TTA lossless compressed audio format supports both ID3v1/ID3v2 information tags and APEv2. The TTA lossless audio codec a ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Apple Lossless
The Apple Lossless Audio Codec (ALAC), also known as Apple Lossless, or Apple Lossless Encoder (ALE), is an audio coding format, and its reference audio codec implementation, developed by Apple Inc. for lossless data compression of digital music. After initially keeping it proprietary from its inception in 2004, in late 2011 Apple made the codec available open source and royalty-free. Traditionally, Apple has referred to the codec as ''Apple Lossless'', though more recently it has begun to use the abbreviated term ''ALAC'' when referring to the codec. Codec ALAC supports up to 8 channels of audio at 16, 20, 24 and 32 bit depth with a maximum sample rate of 384 kHz. ALAC data is frequently stored within an MP4 container with the filename extension ''.m4a''. This extension is also used by Apple for lossy AAC audio data in an MP4 container (same container, different audio encoding). The codec can also be used by the .CAF file type container, though this is much less common. ... [...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] |
Compression Artifact
A compression artifact (or artefact) is a noticeable distortion of media (including images, audio, and video) caused by the application of lossy compression. Lossy data compression involves discarding some of the media's data so that it becomes small enough to be stored within the desired disk space or transmitted (''streamed'') within the available bandwidth (known as the data rate or bit rate). If the compressor cannot store enough data in the compressed version, the result is a loss of quality, or introduction of artifacts. The compression algorithm may not be intelligent enough to discriminate between distortions of little subjective importance and those objectionable to the user. The most common digital compression artifacts are DCT blocks, caused by the discrete cosine transform (DCT) compression algorithm used in many digital media standards, such as JPEG, MP3, and MPEG video file formats. These compression artifacts appear when heavy compression is applied, and occu ... [...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 ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Bitrate Peeling
Bitrate peeling is a technique used in Ogg Vorbis audio encoded streams, wherein a stream can be encoded at one bitrate but can be served at that or any lower bitrate. The purpose is to provide access to the clip for people with slower Internet connections, and yet still allow people with faster connections to enjoy the higher quality content. The server automatically chooses which stream to deliver to the user, depending on user's connection speed. , Ogg Vorbis bitrate peeling existed only as a concept as there was not yet an encoder capable of producing peelable datastreamBounties - XiphWiki Difference from other technologies The difference between SureStream and bitrate peeling is that SureStream is limited to only a handful of pre-defined bitrates, with significant difference between them, and SureStream encoded files are big because they contain all of the bitrates used, while bitrate peeling uses much smaller steps to change the available bitrate and quality, and only th ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
