Elementary Stream
An elementary stream (ES) as defined by the MPEG communication protocol is usually the output of an audio encoder or video encoder. An ES contains only one kind of data (e.g. audio, video, or closed caption). An elementary stream is often referred to as "elementary", "data", "audio", or "video" bitstreams or streams. The format of the elementary stream depends upon the codec or data carried in the stream, but will often carry a common header when packetized into a packetized elementary stream. Header for MPEG-2 video elementary stream General layout of MPEG-1 audio elementary stream The digitized sound signal is divided up into blocks of 384 samples in Layer I and 1152 samples in Layers II and III. The sound sample block is encoded within an audio frame: * header * error check * audio data * ancillary data The header of a frame contains general information such as the MPEG Layer, the sampling frequency, the number of channels, whether the frame is CRC protected, whether the ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MPEG
The Moving Picture Experts Group (MPEG) is an alliance of working groups established jointly by International Organization for Standardization, ISO and International Electrotechnical Commission, IEC that sets standards for media coding, including compression coding of audio compression (data), audio, video compression, video, graphics, and Compression of Genomic Sequencing Data, genomic data; and transmission and Container format (digital), file formats for various applications.John Watkinson, ''The MPEG Handbook'', p. 1 Together with Joint Photographic Experts Group, JPEG, MPEG is organized under ISO/IEC JTC 1/ISO/IEC JTC 1/SC 29, SC 29 – ''Coding of audio, picture, multimedia and hypermedia information'' (ISO/IEC Joint Technical Committee 1, Subcommittee 29). MPEG formats are used in various multimedia systems. The most well known older MPEG media formats typically use MPEG-1, MPEG-2, and MPEG-4 AVC media coding and MPEG-2 systems MPEG transport stream, transport streams an ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Quantization (image Processing)
Quantization, involved in image processing, is a lossy compression technique achieved by compressing a range of values to a single quantum (discrete) value. When the number of discrete symbols in a given stream is reduced, the stream becomes more compressible. For example, reducing the number of colors required to represent a digital image makes it possible to reduce its file size. Specific applications include DCT data quantization in JPEG and DWT data quantization in JPEG 2000. Color quantization Color quantization reduces the number of colors used in an image; this is important for displaying images on devices that support a limited number of colors and for efficiently compressing certain kinds of images. Most bitmap editors and many operating systems have built-in support for color quantization. Popular modern color quantization algorithms include the nearest color algorithm (for fixed palettes), the median cut algorithm, and an algorithm based on octrees. It is common ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MPEG Program Stream
Program stream (PS or MPEG-PS) is a container format for multiplexing digital audio, video and more. The PS format is specified in MPEG-1 Part 1 (ISO/IEC 11172-1) and MPEG-2 Part 1, Systems (ISO/IEC standard 13818-1/ITU-T H.222.0). The MPEG-2 Program Stream is analogous and similar to ISO/IEC 11172 Systems layer and it is forward compatible.ISO (2000-12-01ISO/IEC 13818-1 : 2000, Second editionPage X, Retrieved on 2009-07-25 Program streams are used on DVD-Video discs and HD DVD video discs, but with some restrictions and extensions.DVD - MPeg differences Retrieved on 2009-07-24 The filename extensions are VOB and EVO respectively. Coding structure ...
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Copyright
A copyright is a type of intellectual property that gives its owner the exclusive right to copy, distribute, adapt, display, and perform a creative work, usually for a limited time. The creative work may be in a literary, artistic, educational, or musical form. Copyright is intended to protect the original expression of an idea in the form of a creative work, but not the idea itself. A copyright is subject to limitations based on public interest considerations, such as the fair use doctrine in the United States. Some jurisdictions require "fixing" copyrighted works in a tangible form. It is often shared among multiple authors, each of whom holds a set of rights to use or license the work, and who are commonly referred to as rights holders. These rights frequently include reproduction, control over derivative works, distribution, public performance, and moral rights such as attribution. Copyrights can be granted by public law and are in that case considered "territorial righ ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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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 to 8 bit/s. Prefixes When quantifying large or small bit rates, SI prefixes (also known as metric prefixes or decimal prefixes) are used, thus: Binary prefixes are sometimes used for bit rates. The International Standard ( IEC 80000-13) specifies different a ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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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 (computing), 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 Dolby AC-3, 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 (Communications), USB packets begin with a sync field (8 bits long at low speed, 32 bits long at high speed) used to synchronize the r ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Cyclic Redundancy Check
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 ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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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 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 type of low-pass filter called 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 samp ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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MPEG-1 Audio Layer II
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II (MP2, sometimes incorrectly called Musicam or MUSICAM) is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting. History of development from MP2 to MP3 MUSICAM MPEG-1 Audio Layer 2 encoding was derived from the MUSICAM (''Masking pattern adapted Universal Subband Integrated Coding And Multiplexing'') audio codec, developed by Centre commun d'études de télévision et télécommunications (CCETT), Philips, and the Institut für Rundfunktechnik (IRT) in 1989 as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of a system for the broadcasting of audio and data to fixed, portable or mobile receivers (established in 1987). It began as the Digital Audio Broadcast (DAB) project manage ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Video Buffering Verifier
The Video Buffering Verifier (VBV) is a theoretical MPEG video buffer model, used to ensure that an encoded video stream can be correctly buffered, and played back at the decoder device. By definition, the VBV shall not overflow nor underflow when its input is a compliant stream, (except in the case of low_delay). It is therefore important when encoding such a stream that it comply with the VBV requirements. One way to think of the VBV is to consider both a maximum bitrate and a maximum buffer size. You'll need to know how quickly the video data is coming into the buffer. Keep in mind that video data is always changing the bitrate so there is no constant number to note how fast the data is arriving. The larger question is how long before the buffer overflows. A larger buffer size simply means that the decoder will tolerate high bitrates for longer periods of time, but no buffer is infinite, so eventually even a large buffer will overflow. Operation Modes There are two operational ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |
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Audio Encoder
An audio codec is a device or computer program capable of encoding or decoding a digital data stream (a codec) that encodes or decodes audio. In software, an audio codec is a computer program implementing an algorithm that compresses and decompresses digital audio data according to a given audio file or streaming media audio coding format. The objective of the algorithm is to represent the high-fidelity audio signal with minimum number of bits while retaining quality. This can effectively reduce the storage space and the bandwidth required for transmission of the stored audio file. Most software codecs are implemented as libraries which interface to one or more multimedia players. Most modern audio compression algorithms are based on modified discrete cosine transform (MDCT) coding and linear predictive coding (LPC). In hardware, audio codec refers to a single device that encodes analog audio as digital signals and decodes digital back into analog. In other words, it contains bot ... [...More Info...]       [...Related Items...]     OR:     [Wikipedia]   [Google]   [Baidu]   |