|
TwinVQ
TwinVQ (transform-domain weighted interleave vector quantization) is an audio compression technique developed by Nippon Telegraph and Telephone Corporation (NTT) Human Interface Laboratories (now Cyber Space Laboratories) in 1994. The compression technique has been used in both standardized and proprietary designs. TwinVQ in MPEG-4 In the context of the MPEG-4 Audio (MPEG-4 Part 3), TwinVQ is an audio codec optimized for audio coding at ultra low bitrates around 8 kbit/s. TwinVQ is one of the object types defined in MPEG-4 Audio, published as subpart 4 of ISO/IEC 14496-3 (for the first time in 1999 - a.k.a. MPEG-4 Audio version 1). This object type is based on a general audio transform coding scheme which is integrated with the AAC coding frame work, a spectral flattening module, and a weighted interleave vector quantization module. This scheme reportedly has high coding gain for low bit rate and potential robustness against channel errors and packet loss, since it does no ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
MPEG-4 Part 3
MPEG-4 Part 3 or MPEG-4 Audio (formally ISO/ IEC 14496-3) is the third part of the ISO/ IEC MPEG-4 international standard developed by Moving Picture Experts Group. It specifies audio coding methods. The first version of ISO/IEC 14496-3 was published in 1999. The MPEG-4 Part 3 consists of a variety of audio coding technologies – from lossy speech coding ( HVXC, CELP), general audio coding ( AAC, TwinVQ, BSAC), lossless audio compression (MPEG-4 SLS, Audio Lossless Coding, MPEG-4 DST), a Text-To-Speech Interface (TTSI), Structured Audio (using SAOL, SASL, MIDI) and many additional audio synthesis and coding techniques. MPEG-4 Audio does not target a single application such as real-time telephony or high-quality audio compression. It applies to every application which requires the use of advanced sound compression, synthesis, manipulation, or playback. MPEG-4 Audio is a new type of audio standard that integrates numerous different types of audio coding: natural sound and s ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Vector Quantization
Vector quantization (VQ) is a classical quantization technique from signal processing that allows the modeling of probability density functions by the distribution of prototype vectors. Developed in the early 1980s by Robert M. Gray, it was originally used for data compression. It works by dividing a large set of points (vectors) into groups having approximately the same number of points closest to them. Each group is represented by its centroid point, as in k-means and some other clustering algorithms. In simpler terms, vector quantization chooses a set of points to represent a larger set of points. The density matching property of vector quantization is powerful, especially for identifying the density of large and high-dimensional data. Since data points are represented by the index of their closest centroid, commonly occurring data have low error, and rare data high error. This is why VQ is suitable for lossy data compression. It can also be used for lossy data correction ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Comparison Of Audio Formats
The following tables compare general and technical information for a variety of audio coding formats. For listening tests comparing the perceived audio quality of audio formats and codecs, see the article Codec listening test. General information Notes # The 'Music' category is merely a guideline on commercialized uses of a particular format, not a technical assessment of its capabilities. For example, MP3 and AAC dominate the personal audio market in terms of market share, though many other formats are comparably well suited to fill this role from a purely technical standpoint. # First public release date is first of either specification publishing or source releasing, or in the case of closed-specification, closed-source codecs, is the date of first binary releasing. Many developing codecs have pre-releases consisting of pre-1.0 versions and perhaps 1.0 release candidates (RCs), although 1.0 may not necessarily be the release version. # Latest stable version is that of spe ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Libavcodec
libavcodec is a free and open-source library of codecs for encoding and decoding video and audio data. libavcodec is an integral part of many open-source multimedia applications and frameworks. The popular MPV, xine and VLC media players use it as their main, built-in decoding engine that enables playback of many audio and video formats on all supported platforms. It is also used by the ffdshow tryouts decoder as its primary decoding library. libavcodec is also used in video editing and transcoding applications like Avidemux, MEncoder or Kdenlive for both decoding and encoding. libavcodec contains decoder and sometimes encoder implementations of several proprietary formats, including ones for which no public specification has been released. As such, a significant reverse engineering effort is part of libavcodec development. Having such codecs available within the standard libavcodec framework gives a number of benefits over using the original codecs, most notably increased ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
FFmpeg
FFmpeg is a free and open-source software project consisting of a suite of libraries and programs for handling video, audio, and other multimedia files and streams. At its core is the command-line ffmpeg tool itself, designed for processing video and audio files. It is widely used for format transcoding, basic editing (trimming and concatenation), video scaling, video post-production effects, and standards compliance ( SMPTE, ITU). FFmpeg also includes other tools: ffplay, a simple media player, and ffprobe, a command-line tool to display media information. Among included libraries are libavcodec, an audio/video codec library used by many commercial and free software products, libavformat (Lavf), an audio/video container mux and demux library, and libavfilter, a library for enhancing and editing filters through a GStreamer-like filtergraph. FFmpeg is part of the workflow of many other software projects, and its libraries are a core part of software media players such as V ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Constant Bit Rate
Constant bitrate (CBR) is a term used in telecommunications, relating to the quality of service. Compare with variable bitrate. When referring to codecs, constant bit rate encoding means that the rate at which a codec's output data should be consumed is constant. CBR is useful for streaming multimedia content on limited capacity channels since it is the maximum bit rate that matters, not the average, so CBR would be used to take advantage of all of the capacity. CBR is not optimal for storing data as it may not allocate enough data for complex sections (resulting in degraded quality); and if it maximizes quality for complex sections, it will waste data on simple sections. The problem of not allocating enough data for complex sections could be solved by choosing a high bitrate to ensure that there will be enough bits for the entire encoding process, though the size of the file at the end would be proportionally larger. Most coding schemes such as Huffman coding or run-length enco ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Nero Burning ROM
Nero Burning ROM, commonly called Nero, is an optical disc authoring program from Nero AG. The software is part of the Nero Multimedia Suite but is also available as a stand-alone product. It is used for burning and copying optical media such as CDs, DVDs, and Blu-ray disks. The program also supports the label printing technologies LightScribe and LabelFlash, and can be used to convert audio files into other audio formats. Name Nero Burning ROM is a pun in reference to Roman Emperor Nero, who was best known for his association in the Great Fire of Rome. The emperor allegedly fiddled while the city of Rome burned. Also, Rome in German is spelled Rom. The software's logo features a burning Colosseum, although this is an anachronism as it was not built until after Nero's death. Features Nero Burning ROM is only available for Microsoft Windows. A Linux-compatible version was available from 2005 to 2012, but it has since been discontinued. In newer versions, media can be ad ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
VLC Media Player
VLC media player (previously the VideoLAN Client) is a free and open-source software, free and open-source, software portability, portable, cross-platform media player software and streaming media Server (computing), server developed by the VideoLAN project. VLC is available for desktop operating systems and mobile platforms, such as Android (operating system), Android, iOS and iPadOS. VLC is also available on digital distribution platforms such as Apple Inc., Apple's App Store (iOS), App Store, Google Play, and Microsoft Store (digital), Microsoft Store. VLC supports many data compression, audio- and video-compression methods and file formats, including DVD-Video, Video CD, and streaming-communications protocol, protocols. It is able to stream media over computer networks and can transcode multimedia files. The default distribution of VLC includes many free decoding and encoding libraries, avoiding the need for finding/calibrating proprietary plugins. The libavcodec library fro ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Bit Rate
In telecommunications and computing, bit rate (bitrate or as a variable ''R'') is the number of bits that are conveyed or processed per unit of time. The bit rate is expressed in the unit bit per second (symbol: bit/s), often in conjunction with an SI prefix such as kilo (1 kbit/s = 1,000 bit/s), mega (1 Mbit/s = 1,000 kbit/s), giga (1 Gbit/s = 1,000 Mbit/s) or tera (1 Tbit/s = 1,000 Gbit/s). The non-standard abbreviation bps is often used to replace the standard symbol bit/s, so that, for example, 1 Mbps is used to mean one million bits per second. In most computing and digital communication environments, one byte per second (symbol: B/s) corresponds roughly to 8 bit/s. However if stop bits, start bits, and parity bits need to be factored in, a higher number of bits per second will be required to achieve a throughput of the same number of bytes. Prefixes When quantifying large or small bit rates, SI ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Sampling Frequency
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 th ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Speech Encoding
Speech coding is an application of data compression to digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream. Common applications of speech coding are mobile telephony and voice over IP (VoIP). The most widely used speech coding technique in mobile telephony is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques. The techniques employed in speech coding are similar to those used in audio data compression and audio coding where appreciation of psychoacoustics is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in the frequency band 400 to 3500 Hz is transmitted but ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
Code
In communications and information processing, code is a system of rules to convert information—such as a letter, word, sound, image, or gesture—into another form, sometimes shortened or secret, for communication through a communication channel or storage in a storage medium. An early example is an invention of language, which enabled a person, through speech, to communicate what they thought, saw, heard, or felt to others. But speech limits the range of communication to the distance a voice can carry and limits the audience to those present when the speech is uttered. The invention of writing, which converted spoken language into visual symbols, extended the range of communication across space and time. The process of encoding converts information from a source into symbols for communication or storage. Decoding is the reverse process, converting code symbols back into a form that the recipient understands, such as English, Spanish, etc. One reason for coding is ... [...More Info...] [...Related Items...] OR: [Wikipedia] [Google] [Baidu] |
