1.
Movie
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A film, also called a movie, motion picture, theatrical film or photoplay, is a series of still images which, when shown on a screen, creates the illusion of moving images due to the phi phenomenon. This optical illusion causes the audience to perceive continuous motion between separate objects viewed rapidly in succession, the process of filmmaking is both an art and an industry. The word cinema, short for cinematography, is used to refer to the industry of films. Films were originally recorded onto plastic film through a photochemical process, the adoption of CGI-based special effects led to the use of digital intermediates. Most contemporary films are now fully digital through the process of production, distribution. Films recorded in a form traditionally included an analogous optical soundtrack. It runs along a portion of the film exclusively reserved for it and is not projected, Films are cultural artifacts created by specific cultures. They reflect those cultures, and, in turn, affect them, Film is considered to be an important art form, a source of popular entertainment, and a powerful medium for educating—or indoctrinating—citizens. The visual basis of film gives it a power of communication. Some films have become popular worldwide attractions by using dubbing or subtitles to translate the dialog into the language of the viewer, some have criticized the film industrys glorification of violence and its potentially negative treatment of women. The individual images that make up a film are called frames, the perception of motion is due to a psychological effect called phi phenomenon. The name film originates from the fact that film has historically been the medium for recording and displaying motion pictures. Many other terms exist for a motion picture, including picture, picture show, moving picture, photoplay. The most common term in the United States is movie, while in Europe film is preferred. Terms for the field, in general, include the big screen, the screen, the movies, and cinema. In early years, the sheet was sometimes used instead of screen. Preceding film in origin by thousands of years, early plays and dances had elements common to film, scripts, sets, costumes, production, direction, actors, audiences, storyboards, much terminology later used in film theory and criticism apply, such as mise en scène. Owing to the lack of any technology for doing so, the moving images, the magic lantern, probably created by Christiaan Huygens in the 1650s, could be used to project animation, which was achieved by various types of mechanical slides
2.
PAL
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Phase Alternating Line is a colour encoding system for analogue television used in broadcast television systems in most countries broadcasting at 625-line /50 field per second. Other common colour encoding systems are NTSC and SECAM, all the countries using PAL are currently in process of conversion or have already converted standards to DVB, ISDB or DTMB. This page primarily discusses the PAL colour encoding system, the articles on broadcast television systems and analogue television further describe frame rates, image resolution and audio modulation. To overcome NTSCs shortcomings, alternative standards were devised, resulting in the development of the PAL, the goal was to provide a colour TV standard for the European picture frequency of 50 fields per second, and finding a way to eliminate the problems with NTSC. PAL was developed by Walter Bruch at Telefunken in Hannover, Germany, with important input from Dr. Kruse, the format was patented by Telefunken in 1962, citing Bruch as inventor, and unveiled to members of the European Broadcasting Union on 3 January 1963. When asked, why the system was named PAL and not Bruch the inventor answered that a Bruch system would not have sold very well. The first broadcasts began in the United Kingdom in June 1967, the one BBC channel initially using the broadcast standard was BBC2, which had been the first UK TV service to introduce 625-lines in 1964. Telefunken PALcolor 708T was the first PAL commercial TV set and it was followed by Loewe-Farbfernseher S920 & F900. Telefunken was later bought by the French electronics manufacturer Thomson, Thomson also bought the Compagnie Générale de Télévision where Henri de France developed SECAM, the first European Standard for colour television. The term PAL was often used informally and somewhat imprecisely to refer to the 625-line/50 Hz television system in general, accordingly, DVDs were labelled as PAL or NTSC even though technically the discs do not carry either PAL or NTSC composite signal. CCIR 625/50 and EIA 525/60 are the names for these standards, PAL. Both the PAL and the NTSC system use a quadrature amplitude modulated subcarrier carrying the chrominance information added to the video signal to form a composite video baseband signal. The frequency of this subcarrier is 4.43361875 MHz for PAL and NTSC4.43, the SECAM system, on the other hand, uses a frequency modulation scheme on its two line alternate colour subcarriers 4.25000 and 4.40625 MHz. Early PAL receivers relied on the eye to do that cancelling, however. The effect is that phase errors result in changes, which are less objectionable than the equivalent hue changes of NTSC. In any case, NTSC, PAL, and SECAM all have chrominance bandwidth reduced greatly compared to the luminance signal. The 4.43361875 MHz frequency of the carrier is a result of 283.75 colour clock cycles per line plus a 25 Hz offset to avoid interferences. Since the line frequency is 15625 Hz, the carrier frequency calculates as follows,4.43361875 MHz =283.75 ×15625 Hz +25 Hz
3.
SECAM
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SECAM, also written SÉCAM, is an analogue colour television system first used in France. It was one of three major television standards, the others being the European PAL and North American NTSC. Development of SECAM began in 1956 by a led by Henri de France working at Compagnie Française de Télévision. The first SECAM broadcast was made in France in 1967, making it the first such standard to go live in Europe, the system was also selected as the standard for colour in the Soviet Union, who began broadcasts shortly after the French. The standard spread from these two countries to many client states and former colonies, SECAM remained a major standard into the 2000s. It is in the process of being phased out and replaced by DVB, work on SECAM began in 1956. The technology was ready by the end of the 1950s, a version of SECAM for the French 819-line television standard was devised and tested, but not introduced. The first proposed system was called SECAM I in 1961, followed by studies to improve compatibility. These improvements were called SECAM II and SECAM III, with the latter being presented at the 1965 CCIR General Assembly in Vienna, further improvements were SECAM III A followed by SECAM III B, the adopted system for general use in 1967. Soviet technicians were involved in the development of the standard, and created their own incompatible variant called NIR or SECAM IV, the team was working in Moscows Telecentrum under the direction of Professor Shmakov. The NIR designation comes from the name of the Nautchno-Issledovatelskiy Institut Radio, two standards were developed, Non-linear NIR, in which a process analogous to gamma correction is used, and Linear NIR or SECAM IV that omits this process. SECAM was inaugurated in France on 1 October 1967, on la deuxième chaîne, a group of four suited men—a presenter and three contributors to the systems development—were shown standing in a studio. Following a count from 10, at 2,15 pm the black-and-white image switched to color, in 1967, CLT of Lebanon became the third television station in the world, after the Soviet Union and France, to broadcast in color utilizing the French SECAM technology. The first color television sets cost 5000 Francs, color TV was not very popular initially, only about 1500 people watched the inaugural program in color. A year later, only 200,000 sets had been sold of an expected million and this pattern was similar to the earlier slow build-up of color television popularity in the US. SECAM was later adopted by former French and Belgian colonies, Greece, the Soviet Union and Eastern bloc countries, and Middle Eastern countries. However, with the fall of communism, and following a period when multi-standard TV sets became a commodity, other countries, notably the United Kingdom and Italy, briefly experimented with SECAM before opting for PAL. Since late 2000s, SECAM is in the process of being phased out, some have argued that the primary motivation for the development of SECAM in France was to protect French television equipment manufacturers
4.
NTSC
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The first NTSC standard was developed in 1941 and had no provision for color. In 1953 a second NTSC standard was adopted, which allowed for television broadcasting which was compatible with the existing stock of black-and-white receivers. NTSC was the first widely adopted broadcast color system and remained dominant until 1997, North America, parts of Central America, and South Korea are adopting or have adopted the ATSC standards, while other countries are adopting or have adopted other standards instead of ATSC. After nearly 70 years, the majority of over-the-air NTSC transmissions in the United States ceased on January 1,2010, the majority of NTSC transmissions ended in Japan on July 24,2011, with the Japanese prefectures of Iwate, Miyagi, and Fukushima ending the next year. In March 1941, the committee issued a standard for black-and-white television that built upon a 1936 recommendation made by the Radio Manufacturers Association. Technical advancements of the side band technique allowed for the opportunity to increase the image resolution. The NTSC selected 525 scan lines as a compromise between RCAs 441-scan line standard and Philcos and DuMonts desire to increase the number of lines to between 605 and 800. The standard recommended a frame rate of 30 frames per second, other standards in the final recommendation were an aspect ratio of 4,3, and frequency modulation for the sound signal. In January 1950, the committee was reconstituted to standardize color television, in December 1953, it unanimously approved what is now called the NTSC color television standard. The compatible color standard retained full backward compatibility with existing black-and-white television sets, Color information was added to the black-and-white image by introducing a color subcarrier of precisely 315/88 MHz. These changes amounted to 0.1 percent and were tolerated by existing television receivers. The FCC had briefly approved a different color standard, starting in October 1950. However, this standard was incompatible with black-and-white broadcasts and it used a rotating color wheel, reduced the number of scan lines from 525 to 405, and increased the field rate from 60 to 144, but had an effective frame rate of only 24 frames per second. CBS rescinded its system in March 1953, and the FCC replaced it on December 17,1953, with the NTSC color standard, later that year, the improved TK-41 became the standard camera used throughout much of the 1960s. The NTSC standard has been adopted by countries, including most of the Americas. With the advent of television, analog broadcasts are being phased out. Most US NTSC broadcasters were required by the FCC to shut down their analog transmitters in 2009, low-power stations, Class A stations and translators were required to shut down by 2015. NTSC color encoding is used with the System M television signal, each frame is composed of two fields, each consisting of 262.5 scan lines, for a total of 525 scan lines
5.
MP3
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Compared to CD quality digital audio, MP3 compression commonly achieves 75 to 95% reduction in size. MP3 files are thus 1/4 to 1/20 the size of the digital audio stream. This is important for both transmission and storage concerns, the basis for such comparison is the CD digital audio format which requires 1411200 bit/s. A commonly used MP3 encoding setting is CBR128 kbit/s resulting in file size 1/11 of the original CD-quality file, the MP3 lossy compression works by reducing the accuracy of certain parts of a continuous sound that are considered to be beyond the auditory resolution ability of most people. This method is referred to as perceptual coding or psychoacoustics. It uses psychoacoustic models to discard or reduce the precision of less audible to human hearing. MP3 was designed by the Moving Picture Experts Group as part of its MPEG-1 standard, the first subgroup for audio was formed by several teams of engineers at Fraunhofer IIS, University of Hanover, AT&T-Bell Labs, Thomson-Brandt, CCETT, and others. MPEG-1 Audio, which included MPEG-1 Audio Layer I, II and III was approved as a draft of ISO/IEC standard in 1991, finalised in 1992. A backwards compatible MPEG-2 Audio extension with lower sample and bit rates was published in 1995, MP3 is a streaming or broadcast format meaning that individual frames can be lost without affecting the ability to decode successfully delivered frames. Storing an MP3 stream in a file enables time-shifted playback, the MP3 lossy audio data compression algorithm takes advantage of a perceptual limitation of human hearing called auditory masking. In 1894, the American physicist Alfred M. Mayer reported that a tone could be rendered inaudible by another tone of lower frequency, in 1959, Richard Ehmer described a complete set of auditory curves regarding this phenomenon. Ernst Terhardt et al. created an algorithm describing auditory masking with high accuracy and this work added to a variety of reports from authors dating back to Fletcher, and to the work that initially determined critical ratios and critical bandwidths. A wide variety of compression algorithms were reported in IEEEs refereed Journal on Selected Areas in Communications. The genesis of the MP3 technology is described in a paper from Professor Hans Musmann who chaired the ISO MPEG Audio group for several years. The immediate predecessors of MP3 were Optimum Coding in the Frequency Domain, the first practical implementation of an audio perceptual coder in hardware, was an implementation of a psychoacoustic transform coder based on Motorola 56000 DSP chips. Another predecessor of the MP3 format and technology is to be found in the perceptual codec MUSICAM based on an integer arithmetics 32 sub-bands filterbank, driven by a psychoacoustic model. It was primarily designed for Digital Audio Broadcasting and digital TV and this codec incorporated into a broadcasting system using COFDM modulation was demonstrated on air and on the field together with Radio Canada and CRC Canada during the NAB show in 1991. F. As a doctoral student at Germanys University of Erlangen-Nuremberg, Karlheinz Brandenburg began working on music compression in the early 1980s
6.
UTF-8
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UTF-8 is a character encoding capable of encoding all possible characters, or code points, defined by Unicode and originally designed by Ken Thompson and Rob Pike. The encoding is variable-length and uses 8-bit code units and it was designed for backward compatibility with ASCII and to avoid the complications of endianness and byte order marks in the alternative UTF-16 and UTF-32 encodings. The name is derived from Unicode Transformation Format – 8-bit, UTF-8 is the dominant character encoding for the World Wide Web, accounting for 88. 9% of all Web pages in April 2017. The Internet Mail Consortium recommended that all programs be able to display and create mail using UTF-8. UTF-8 encodes each of the 1,112,064 valid code points in Unicode using one to four 8-bit bytes, code points with lower numerical values, which tend to occur more frequently, are encoded using fewer bytes. The following table shows the structure of the encoding, the x characters are replaced by the bits of the code point. If the number of significant bits is no more than 7, the first line applies, if no more 11 bits, the second line applies, the first 128 characters need one byte. Three bytes are needed for characters in the rest of the Basic Multilingual Plane, four bytes are needed for characters in the other planes of Unicode, which include less common CJK characters, various historic scripts, mathematical symbols, and emoji. The salient features of this scheme are as follows, Backward compatibility, One-byte codes are used for the ASCII values 0 through 127, clear indication of byte sequence length, The first byte indicates the number of bytes in the sequence. The length of multi-byte sequences is determined as it is simply the number of high-order 1s in the leading byte. Self-synchronization, The leading bytes and the continuation bytes do not share values and this means a search will not accidentally find the sequence for one character starting in the middle of another character. It also means the start of a character can be found from a position by backing up at most 3 bytes to find the leading byte. Consider the encoding of the Euro sign, €, the Unicode code point for € is U+20AC. According to the table above, this will take three bytes to encode, since it is between U+0800 and U+FFFF. Hexadecimal 20AC is binary 0010000010101100, the two leading zeros are added because, as the scheme table shows, a three-byte encoding needs exactly sixteen bits from the code point. All continuation bytes contain exactly six bits from the code point, so the next six bits of the code point are stored in the low order six bits of the next byte, and 10 is stored in the high order two bits to mark it as a continuation byte. Finally the last six bits of the point are stored in the low order six bits of the final byte. The three bytes 111000101000001010101100 can be concisely written in hexadecimal, as E282 AC
7.
Digital cinema
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Digital cinema refers to the use of digital technology to distribute or project motion pictures as opposed to the historical use of reels of motion picture film, such as 35 mm film. Digital movies are projected using a digital projector instead of a film projector. Digital cinema is distinct from high-definition television and is not dependent on using television or high-definition video standards, aspect ratios, in digital cinema, resolutions are represented by the horizontal pixel count, usually 2K or 4K. As digital cinema technology improved in the early 2010s, most of the theaters across the world converted to digital, Digital media playback of high-resolution 2K files has at least a 20-year history. Early video data storage units fed custom frame buffer systems with large memories, in early digital video units, content was usually restricted to several minutes of material. Transfer of content between remote locations was slow and had limited capacity and it was not until the late 1990s that feature-length films could be sent over the wire. In conjunction with Texas Instruments, the movie was publicly demonstrated in five theaters across the United States. In Europe, on February 2,2000, Texas Instruments DLP Cinema projector technology was demonstrated, by Philippe Binant. On January 19,2000, the Society of Motion Picture and Television Engineers, in the United States, by December 2000, there were 15 digital cinema screens in the United States and Canada,11 in Western Europe,4 in Asia, and 1 in South America. Digital Cinema Initiatives was formed in March 2002 as a joint project of many motion picture studios to develop a specification for digital cinema. In April 2004, in cooperation with the American Society of Cinematographers, DCI created standard evaluation material for testing of 2K and 4K playback, DCI selected JPEG2000 as the basis for the compression in the system the same year. In China, in June 2005, an E-Cinema System called dMs was established and was used in over 15,000 screens spread across Chinas 30 provinces, DMS estimated that the system would expand to 40,000 screens in 2009. In 2005 the UK Film Council Digital Screen Network launched in the UK by Arts Alliance Media creating a chain of 250 2K digital cinema systems, the roll out was completed in 2006. This was the first mass roll out in Europe, axis IT/Christie Digital also started a roll out in the United States and Canada. By mid 2006, about 400 theaters were equipped with 2K digital projectors with the number increasing every month, several digital 3D films surfaced in 2006 and several prominent filmmakers committed to making their next productions in stereo 3D. VUE West End was one of the first 3D Digital Cinemas along with Odeon Printworks Manchester and this was done by Emil and Eric Digital Films, a company based at Thrissur using the end-to-end digital cinema system developed by Singapore-based DG2L Technologies. This film was mastered at Real Image Media Technologies in India. In June 2007, Arts Alliance Media announced the first European commercial digital cinema Virtual Print Fee agreements, by June 2010, there were close to 16,000 digital cinema screens, with over 5000 of them being stereoscopic setups
8.
JPEG 2000
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JPEG2000 is an image compression standard and coding system. The standardized filename extension is. jp2 for ISO/IEC 15444-1 conforming files and. jpx for the extended part-2 specifications, the registered MIME types are defined in RFC3745. For ISO/IEC 15444-1 it is image/jp2, JPEG2000 code streams are regions of interest that offer several mechanisms to support spatial random access or region of interest access at varying degrees of granularity. It is possible to store different parts of the picture using different quality. As of 2017, there are very few digital cameras that shoot photos in the JPEG2000 format, while there is a modest increase in compression performance of JPEG2000 compared to JPEG, the main advantage offered by JPEG2000 is the significant flexibility of the codestream. By ordering the codestream in various ways, applications can achieve significant performance increases, however, as a consequence of this flexibility, JPEG2000 requires encoders/decoders that are complex and computationally demanding. JPEG2000 has been published as an ISO standard, ISO/IEC15444, as of 2013, JPEG2000 is not widely supported in web browsers, and hence is not generally used on the Internet. At high bit rates, artifacts become nearly imperceptible, JPEG2000 has a small machine-measured fidelity advantage over JPEG, at lower bit rates, JPEG2000 has a significant advantage over certain modes of JPEG, artifacts are less visible and there is almost no blocking. The compression gains over JPEG are attributed to the use of DWT, JPEG2000 decomposes the image into a multiple resolution representation in the course of its compression process. This pyramid representation can be put to use for other image presentation purposes beyond compression and these features are more commonly known as progressive decoding and signal-to-noise ratio scalability. JPEG2000 provides efficient code-stream organizations which are progressive by pixel accuracy and this way, after a smaller part of the whole file has been received, the viewer can see a lower quality version of the final picture. The quality then improves progressively through downloading more data bits from the source, like the Lossless JPEG standard, the JPEG2000 standard provides both lossless and lossy compression in a single compression architecture. Lossless compression is provided by the use of a reversible wavelet transform in JPEG2000. Like JPEG1992, JPEG2000 is robust to bit errors introduced by noisy communication channels, the JP2 and JPX file formats allow for handling of color-space information, metadata, and for interactivity in networked applications as developed in the JPEG Part 9 JPIP protocol. JPEG2000 supports any bit depth, such as 16- and 32-bit floating point pixel images, full support for transparency and alpha planes. JPEG 2000s improvement in performance relative to the original JPEG standard is actually rather modest. Very low and very high rates are supported in JPEG2000. The ability of the design to handle a large range of effective bit rates is one of the strengths of JPEG2000
9.
Lossy compression
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In information technology, lossy compression or irreversible compression is the class of data encoding methods that uses inexact approximations and partial data discarding to represent the content. These techniques are used to reduce data size for storage, handling, different versions of the photo of the cat above show how higher degrees of approximation create coarser images as more details are removed. This is opposed to data compression which does not degrade the data. The amount of data reduction possible using lossy compression is often higher than through lossless techniques. Well-designed lossy compression technology often reduces file sizes significantly before degradation is noticed by the end-user, even when noticeable by the user, further data reduction may be desirable. Lossy compression is most commonly used to compress multimedia data, especially in such as streaming media. By contrast, lossless compression is required for text and data files, such as bank records. A picture, for example, is converted to a file by considering it to be an array of dots and specifying the color. If the picture contains an area of the color, it can be compressed without loss by saying 200 red dots instead of red dot. The original data contains an amount of information, and there is a lower limit to the size of file that can carry all the information. Basic information theory says there is an absolute limit in reducing the size of this data. When data is compressed, its entropy increases, and it cannot increase indefinitely, as an intuitive example, most people know that a compressed ZIP file is smaller than the original file, but repeatedly compressing the same file will not reduce the size to nothing. Most compression algorithms can recognize when further compression would be pointless, in many cases, files or data streams contain more information than is needed for a particular purpose. Developing lossy compression techniques as closely matched to human perception as possible is a complex task, the terms irreversible and reversible are preferred over lossy and lossless respectively for some applications, such as medical image compression, to circumvent the negative implications of loss. The type and amount of loss can affect the utility of the images, artifacts or undesirable effects of compression may be clearly discernible yet the result still useful for the intended purpose. Or lossy compressed images may be visually lossless, or in the case of medical images and this is because uncompressed audio can only reduce file size by lowering bit rate or depth, whereas compressing audio can reduce size while maintaining bit rate and depth. This compression becomes a loss of the least significant data. From this point of view, perceptual encoding is not essentially about discarding data, green, and 50 pixels of blue vs. red, which are proportional to human sensitivity to each component
10.
Generation loss
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Generation loss is the loss of quality between subsequent copies or transcodes of data. Anything that reduces the quality of the representation when copying, and would cause further reduction in quality on making a copy of the copy, can be considered a form of generation loss. File size increases are a result of generation loss, as the introduction of artifacts may actually increase the entropy of the data through each generation. In analog systems, generation loss is due to noise and bandwidth issues in cables, amplifiers, mixers, recording equipment. Poorly adjusted distribution amplifiers and mismatched impedances can make these problems even worse, repeated conversion between analog and digital can also cause loss. Careful planning was required to minimize loss, and the resulting noise. According to ATIS, Generation loss is limited to analog recording because digital recording, used correctly, digital technology can eliminate generation loss. Copying a digital file gives a copy if the equipment is operating properly. This trait of digital technology has given rise to awareness of the risk of unauthorized copying, before digital technology was widespread, a record label, for example, could be confident knowing that unauthorized copies of their music tracks were never as good as the originals. Processing a lossily compressed file rather than an original usually results in loss of quality than generating the same output from an uncompressed original. For example, a digital image for a web page is better if generated from an uncompressed raw image than from an already-compressed JPEG file of higher quality. In digital systems, several techniques, used because of advantages, may introduce generation loss. However, copying a digital file itself incurs no generation loss--the copied file is identical to the original, some digital transforms are reversible, while some are not. Lossless compression is, by definition, fully reversible, while lossy compression throws away some data which cannot be restored, similarly, many DSP processes are not reversible. Thus careful planning of an audio or video signal chain from beginning to end, similarly, when using lossy compression, it will ideally only be done once, at the end of the workflow involving the file, after all required changes have been made. Repeated applications of lossy compression and decompression can cause generation loss, some lossy compression algorithms are much worse than others in this regard, being neither idempotent nor scalable, and introducing further degradation if parameters are changed. For example, with JPEG, changing the quality setting will cause different quantization constants to be used, causing additional loss. Further, as JPEG is divided into 16×16 blocks, cropping that does not fall on an 8×8 boundary shifts the encoding blocks and this can be avoided by the use of jpegtran or similar tools for cropping
11.
Pulse-code modulation
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Pulse-code modulation is a method used to digitally represent sampled analog signals. It is the form of digital audio in computers, compact discs, digital telephony. In a PCM stream, the amplitude of the signal is sampled regularly at uniform intervals. Linear pulse-code modulation is a type of PCM where the quantization levels are linearly uniform. This is in contrast to PCM encodings where quantization levels vary as a function of amplitude, though PCM is a more general term, it is often used to describe data encoded as LPCM. Early electrical communications started to sample signals in order to interlace samples from multiple telegraphy sources, the American inventor Moses G. Farmer conveyed telegraph time-division multiplexing as early as 1853. Electrical engineer W. M. Miner, in 1903, used an electro-mechanical commutator for time-division multiplexing multiple telegraph signals and he obtained intelligible speech from channels sampled at a rate above 3500–4300 Hz, lower rates proved unsatisfactory. This was TDM, but pulse-amplitude modulation rather than PCM, in 1920 the Bartlane cable picture transmission system, named after its inventors Harry G. Bartholomew and Maynard D. In 1926, Paul M. Rainey of Western Electric patented a machine which transmitted its signal using 5-bit PCM. The machine did not go into production, british engineer Alec Reeves, unaware of previous work, conceived the use of PCM for voice communication in 1937 while working for International Telephone and Telegraph in France. He described the theory and advantages, but no practical application resulted, Reeves filed for a French patent in 1938, and his US patent was granted in 1943. By this time Reeves had started working at the Telecommunications Research Establishment, the first transmission of speech by digital techniques, the SIGSALY encryption equipment, conveyed high-level Allied communications during World War II. In 1943 the Bell Labs researchers who designed the SIGSALY system became aware of the use of PCM binary coding as already proposed by Alec Reeves. In 1949 for the Canadian Navys DATAR system, Ferranti Canada built a working PCM radio system that was able to transmit digitized radar data over long distances, PCM in the late 1940s and early 1950s used a cathode-ray coding tube with a plate electrode having encoding perforations. The plate collected or passed the beam, producing current variations in binary code, rather than natural binary, the grid of Goodalls later tube was perforated to produce a glitch-free Gray code, and produced all bits simultaneously by using a fan beam instead of a scanning beam. Patent 2,801,281 filed in 1946 and 1952, another patent by the same title was filed by John R. Pierce in 1945, and issued in 1948, U. S. The three of them published The Philosophy of PCM in 1948, PCM is the method of encoding generally used for uncompressed audio, although there are other methods such as pulse-density modulation. The 4ESS switch introduced time-division switching into the US telephone system in 1976, LPCM is used for the lossless encoding of audio data in the Compact disc Red Book standard, introduced in 1982
12.
YUV
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YUV is a color space typically used as part of a color image pipeline. The scope of the terms Y′UV, YUV, YCbCr, YPbPr, etc. is sometimes ambiguous, today, the term YUV is commonly used in the computer industry to describe file-formats that are encoded using YCbCr. The Y′UV model defines a space in terms of one luma. The Y′UV color model is used in the PAL and SECAM composite color video standards, previous black-and-white systems used only luma information. The YPbPr color model used in analog component video and its digital version YCbCr used in video are more or less derived from it. The Y′IQ color space used in the analog NTSC television broadcasting system is related to it, as for etymology, Y, Y′, U, and V are not abbreviations. The use of the letter Y for luminance can be traced back to the choice of X Y Z primaries and this lends itself naturally to the usage of the same letter in luma, which approximates a perceptually uniform correlate of luminance. Likewise, U and V were chosen to differentiate the U and V axes from those in other spaces, see the equations below or compare the historical development of the math. Y′UV was invented when engineers wanted color television in a black-and-white infrastructure and they needed a signal transmission method that was compatible with black-and-white TV while being able to add color. The luma component already existed as the black and white signal, the UV representation of chrominance was chosen over straight R and B signals because U and V are color difference signals. This meant that in a black and white scene the U and V signals would be zero, if R and B were to have been used, these would have non-zero values even in a B&W scene, requiring all three data-carrying signals. In addition, black and white receivers could take the Y′ signal and ignore the signals, making Y′UV backward-compatible with all existing black-and-white equipment. It was necessary to assign a narrower bandwidth to the channel because there was no additional bandwidth available. If some of the information arrived via the chrominance channel. Y′UV signals are created from RGB source. Weighted values of R, G, and B are summed to produce Y′, U and V are computed as scaled differences between Y′ and the B and R values. BT.601 defines the following constants, W R =0.299, W G =1 − W R − W B =0.587, W B =0.114, U max =0.436, V max =0.615. The resulting ranges of Y′, U, and V respectively are, equivalently, substituting values for the constants and expressing them as matrices gives these formulas for BT.601, =, =
13.
JPEG
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JPEG is a commonly used method of lossy compression for digital images, particularly for those images produced by digital photography. The degree of compression can be adjusted, allowing a selectable tradeoff between size and image quality. JPEG typically achieves 10,1 compression with little loss in image quality. JPEG compression is used in a number of file formats. These format variations are not distinguished, and are simply called JPEG. The term JPEG is an initialism/acronym for the Joint Photographic Experts Group, the MIME media type for JPEG is image/jpeg, except in older Internet Explorer versions, which provides a MIME type of image/pjpeg when uploading JPEG images. JPEG files usually have an extension of. jpg or. jpeg. JPEG/JFIF supports a maximum size of 65, 535×65,535 pixels. JPEG stands for Joint Photographic Experts Group, the name of the committee created the JPEG standard. The Joint stood for ISO TC97 WG8 and CCITT SGVIII, in 1987 ISO TC97 became ISO/IEC JTC1 and in 1992 CCITT became ITU-T. Currently on the JTC1 side JPEG is one of two sub-groups of ISO/IEC Joint Technical Committee 1, Subcommittee 29, Working Group 1 – titled as Coding of still pictures, on the ITU-T side ITU-T SG16 is the respective body. The original JPEG group was organized in 1986, issuing the first JPEG standard in 1992, which was approved in September 1992 as ITU-T Recommendation T.81 and in 1994 as ISO/IEC 10918-1. The JPEG standard specifies the codec, which defines how an image is compressed into a stream of bytes and decompressed back into an image, the Exif and JFIF standards define the commonly used file formats for interchange of JPEG-compressed images. JPEG standards are formally named as Information technology – Digital compression, ISO/IEC10918 consists of the following parts, Ecma International TR/98 specifies the JPEG File Interchange Format, the first edition was published in June 2009. The JPEG compression algorithm is at its best on photographs and paintings of scenes with smooth variations of tone. For web usage, where the amount of used for an image is important. JPEG/Exif is also the most common format saved by digital cameras, on the other hand, JPEG may not be as well suited for line drawings and other textual or iconic graphics, where the sharp contrasts between adjacent pixels can cause noticeable artifacts. Such images may be saved in a lossless graphics format such as TIFF, GIF, PNG
14.
DVD
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DVD is a digital optical disc storage format invented and developed by Philips, Sony, Toshiba, and Panasonic in 1995. The medium can store any kind of data and is widely used for software. DVDs offer higher capacity than compact discs while having the same dimensions. Pre-recorded DVDs are mass-produced using molding machines that physically stamp data onto the DVD, such discs are a form of DVD-ROM because data can only be read and not written or erased. Blank recordable DVD discs can be recorded using a DVD recorder. Rewritable DVDs can be recorded and erased many times, DVDs containing other types of information may be referred to as DVD data discs. The OED also states that in 1995, The companies said the name of the format will simply be DVD. Toshiba had been using the name ‘digital video disk’, but that was switched to ‘digital versatile disk’ after computer companies complained that it left out their applications, Digital versatile disc is the explanation provided in a DVD Forum Primer from 2000 and in the DVD Forums mission statement. There were several formats developed for recording video on optical discs before the DVD, Optical recording technology was invented by David Paul Gregg and James Russell in 1958 and first patented in 1961. A consumer optical disc data format known as LaserDisc was developed in the United States and it used much larger discs than the later formats. CD Video used analog video encoding on optical discs matching the established standard 120 mm size of audio CDs, Video CD became one of the first formats for distributing digitally encoded films in this format, in 1993. In the same year, two new optical disc formats were being developed. By the time of the launches for both formats in January 1995, the MMCD nomenclature had been dropped, and Philips and Sony were referring to their format as Digital Video Disc. Representatives from the SD camp asked IBM for advice on the system to use for their disc. Alan E. Bell, a researcher from IBMs Almaden Research Center, got that request and this group was referred to as the Technical Working Group, or TWG. On August 14,1995, an ad hoc group formed from five computer companies issued a release stating that they would only accept a single format. The TWG voted to both formats unless the two camps agreed on a single, converged standard. They recruited Lou Gerstner, president of IBM, to pressure the executives of the warring factions, as a result, the DVD specification provided a storage capacity of 4.7 GB for a single-layered, single-sided disc and 8.5 GB for a dual-layered, single-sided disc
15.
Video CD
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Video CD is a home video format and the first format for distributing films on standard 120 mm optical discs. The format was adopted in Southeast Asia instead of VHS. The format is a digital format for storing video on a compact disc. VCDs are playable in dedicated VCD players, most DVD and Blu-ray Disc players, personal computers, the Video CD standard was created in 1993 by Sony, Philips, Matsushita, and JVC and is referred to as the White Book standard. Although they have been superseded by other media, VCDs continue to be retailed as a low-cost video format, in 1979, Philips introduced the optical LaserDisc, which was about 30 cm in diameter. This disc could hold an hour of analog video along with audio on each side. The Laserdisc provided picture quality nearly double that of VHS tape, Philips later teamed up with Sony to develop a new type of disc, the compact disc or CD. Introduced in 1982 in Japan, the CD is about 120 mm in diameter, the format was initially designed to store digitized sound and proved to be a success in the music industry. A few years later, Philips decided to give CDs the ability to produce video and this led to the creation of CD Video in 1987. However, the small size significantly impeded the ability to store analog video. Therefore, CD-V distribution was limited to featuring music videos, by the early 1990s engineers were able to digitize and compress video signals, greatly improving storage efficiency. Because this new format could hold 83 minutes of audio and video, extra capacity was obtained by sacrificing the error correction. This format was named Video CD or VCD, VCD enjoyed a brief period of success, with a few major feature films being released in the format. However, VCDs are still being released in countries in Asia. The development of sophisticated, higher capacity optical disc formats yielded the DVD format. DVD players use lasers that are of shorter wavelength than those used on CDs, allowing the recorded pits to be smaller, the DVD was so successful that it eventually pushed VHS out of the video market once suitable recorders became widely available. Nevertheless, VCDs made considerable inroads into developing nations, where they are still in use today, Video CDs comply with the CD-i Bridge format, and are authored using tracks in CD-ROM XA mode. The first track of a VCD is in CD-ROM XA Mode 2 Form 1 and this track may also contain other non-essential files, and is shown by operating systems when loading the disc
16.
Image scaling
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In computer graphics and digital imaging, image scaling refers to the resizing of a digital image. In video technology, the magnification of digital material is known as upscaling or resolution enhancement, when scaling a vector graphic image, the graphic primitives that make up the image can be scaled using geometric transformations, with no loss of image quality. When scaling a raster image, a new image with a higher or lower number of pixels must be generated. In the case of decreasing the pixel number this usually results in a quality loss. Image scaling can be interpreted as a form of image resampling FIX or image reconstruction from the view of the Nyquist sampling theorem, the image is reduced to the information that can be carried by the smaller image. In the case of up sampling, a reconstruction filter takes the place of the anti-aliasing filter. A more sophisticated approach to up scaling treats the problem as a problem, solving the question of generating a plausible image which. A variety of techniques have been applied for this, including optimization techniques with regularization terms, an image size can be changed in several ways. Diagonal lines, for example, show a stairway shape, although this is desirable for continuous-tone images, this algorithm reduces contrast in a way that may be undesirable for line art. Bicubic interpolation yields substantially better results, with only an increase in computational complexity. Sinc and Lanczos resampling Sinc resampling in theory provides the best possible reconstruction for a perfectly bandlimited signal, in practice, the assumptions behind sinc resampling are not completely met by real-world digital images. Lanczos resampling, an approximation to the method, yields better results. Bicubic interpolation can be regarded as an efficient approximation to Lanczos resampling. Box sampling One weakness of bilinear, bicubic and related algorithms is that sample a specific number of pixels. The trivial solution to this issue is box sampling, which is to consider the target pixel a box on the original image and this ensures that all input pixels contribute to the output. The major weakness of this algorithm is that it is hard to optimize, mipmap Another solution to the downscale problem of bi-sampling scaling are mipmaps. A mipmap is a set of downscale copies. When downscaling the nearest larger mipmap is used as the origin and this is algorithm is fast, and easy to optimize
17.
Decimation (signal processing)
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In digital signal processing, decimation is the process of reducing the sampling rate of a signal. Complementary to upsampling and interpolation, which increases sampling rate, it is a case of sample rate conversion in a multi-rate digital signal processing system. Decimation utilizes filtering to mitigate aliasing distortion, which can occur when simply downsampling a signal, a system component that performs decimation is called a decimator. Decimation reduces the rate or the size of the data. The decimation factor is usually an integer or a rational fraction greater than one and this factor multiplies the sampling time or, equivalently, divides the sampling rate. For example, if 16-bit compact disc audio is decimated to 22,050 Hz, the bit rate is also reduced in half, from 705,600 bit/s to 352,800 bit/s, assuming that each sample retains its bit depth of 16 bits. Downsample the filtered signal by M, that is, keep only every Mth sample, downsampling alone causes high-frequency signal components to be misinterpreted by subsequent users of the data, which is a form of distortion called aliasing. The first step, if necessary, is to suppress aliasing to an acceptable level, in this application, the filter is called an anti-aliasing filter, and its design is discussed below. Also see undersampling for information about downsampling bandpass functions and signals, when the anti-aliasing filter is an IIR design, it relies on feedback from output to input, prior to the downsampling step. With FIR filtering, it is a matter to compute only every Mth output. X represents the sequence being downsampled. In a general purpose processor, after computing y, the easiest way to compute y is to advance the starting index in the x array by M, and recompute the dot product. In the case M=2, h can be designed as a half-band filter, impulse response coefficients taken at intervals of M form a subsequence, and there are M such subsequences multiplexed together. The dot product is the sum of the dot products of each subsequence with the samples of the x sequence. Furthermore, because of downsampling by M, the stream of x samples involved in any one of the M dot products is never involved in the dot products. Thus M low-order FIR filters are each filtering one of M multiplexed phases of the stream. This viewpoint offers a different implementation that might be advantageous in a multi-processor architecture, in other words, the input stream is demultiplexed and sent through a bank of M filters whose outputs are summed. When implemented that way, it is called a polyphase filter, the equivalence of this inefficient method and the implementation described above is known as the first Noble identity
18.
Vorbis
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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 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 companys Ogg 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 BSD license to encourage adoption with endorsement of Richard Stallman. A stable version of the software was released on July 19,2002. The Xiph. Org Foundation maintains an implementation, libvorbis. There are also some fine-tuned forks, most notably aoTuV, that offer better audio quality and these improvements are periodically merged back into the reference codebase. Vorbis is named after a Discworld character, Exquisitor Vorbis in Small Gods by Terry Pratchett, the Ogg format, however, is not named after Nanny Ogg, another Discworld character, the name is in fact derived from ogging, jargon that arose in the computer game Netrek. The Vorbis format has proven popular among supporters of free software and they argue that its higher fidelity and completely free nature, unencumbered by patents, make it a well-suited replacement for patented and restricted formats like MP3. Vorbis has different uses for consumer products, many video game titles store in-game audio as Vorbis, including Amnesia, The Dark Descent, Grand Theft Auto, San Andreas, Halo, Combat Evolved, Minecraft, and World of Warcraft, among others. Popular software players support Vorbis playback either natively or through an external plugin, a number of websites, including Wikipedia, use it. Others include Jamendo and Mindawn, as well as national radio stations like JazzRadio, Absolute Radio, NPR, Radio New Zealand. The Spotify audio streaming service uses Vorbis for its audio streams, also, the French music site Qobuz offers its customers the possibility to download their purchased songs in Vorbis format, as does the American music site Bandcamp. However, by 2014, not many further significant tests had been made, listening tests have attempted to find the best quality lossy audio codecs at certain bitrates. Mid to low bitrates, private tests in 2005 at 80 kbit/s and 96 kbit/s showed that aoTuV Vorbis had a better quality than other lossy audio formats, high bitrates, most people do not hear significant differences. However, trained listeners can often hear significant differences between codecs at identical bitrates, and aoTuV Vorbis performed better than LC-AAC, MP3, due to the ever-evolving nature of audio codecs, the results of many of these tests have become outdated. Listening tests are carried out as ABX tests, i. e. the listener has to identify an unknown sample X as being A or B
19.
Compression artifact
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A compression artifact is a noticeable distortion of media caused by the application of lossy compression. Lossy data compression involves discarding some of the data so that it becomes simplified enough to be stored within the desired disk space or be transmitted within the bandwidth limitations. If the compressor could not reproduce enough data in the version to reproduce the original. Alternatively, the algorithm may not be intelligent enough to discriminate between distortions of little subjective importance and those objectionable to the viewer. Compression artifacts occur in common media such as DVDs, common computer file formats such as JPEG, MP3, or MPEG files. Uncompressed media or losslessly compressed media do not suffer from compression artifacts, the minimization of perceivable artifacts is a key goal in implementing a lossy compression algorithm. However, artifacts are occasionally intentionally produced for artistic purposes, a known as glitch art or datamoshing. Technically speaking, an artifact is a particular class of data error that is usually the consequence of quantization in lossy data compression. Where transform coding is used, they assume the form of one of the basis functions of the coders transform space. When performing block-based coding for quantization, as in JPEG-compressed images, for example, the numbers 6 and 8 may get replaced. This has been observed to happen with JBIG2 in certain photocopier machines, at low bit rates, any lossy block-based coding scheme introduces visible artifacts in pixel blocks and at block boundaries. These boundaries can be transform block boundaries, prediction block boundaries, the term macroblocking is commonly used regardless of the artifacts cause. Other names include tiling, mosaicing, pixelating, quilting, block-artifacts are a result of the very principle of block transform coding. The transform is applied to a block of pixels, and to achieve lossy compression, the lower the bit rate, the more coarsely the coefficients are represented and the more coefficients are quantized to zero. Statistically, images have more low-frequency than high-frequency content, so it is the content that remains after quantization. Because this quantization process is applied individually in each block, neighboring blocks quantize coefficients differently and this leads to discontinuities at the block boundaries. These are most visible in areas, where there is little detail to mask the effect. Many photo editing programs, for instance, have proprietary JPEG artifact reduction algorithms built-in, consumer equipment often calls this post-processing MPEG Noise Reduction
20.
Audio editing software
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Audio editing software is software which allows editing and generating of audio data. Audio editing software can be implemented completely or partly as library, as computer application, wave Editors are digital audio editors and there are many sources of software available to perform this function. Most can edit music, apply effects and filters, adjust stereo channels etc, Editors designed for use with music typically allow the user to do the following, The ability to import and export various audio file formats for editing. Audio editors may process the audio data non-destructively in real-time, or destructively as a process, or a hybrid with some real-time effects. Destructive editing modifies the data of the audio file, as opposed to just editing its playback parameters. Destructive editors are also known as sample editors, destructive editing applies edits and processing directly to the audio data, changing the data immediately. If, for example, part of a track is deleted, real-time editing does not apply changes immediately, but applies edits and processing on the fly during playback. If, for example, part of a track is deleted, the audio data is not actually removed from the track. In graphical editors, all changes to the audio is usually visible immediately as the waveform is updated to match the audio data. The number of effects that may be applied is virtually unlimited, editing is usually precise down to exact sample intervals. Effects may be applied to a precisely specified selected region, mixing down or exporting the edited audio is usually relatively quick as little additional processing is required. Once an effect has been applied, it cannot usually be changed and this is usually mitigated by the ability to undo the last performed action. Typically a destructive audio editor will maintain many levels of history so that multiple actions may be undone in the reverse order that they were applied. Edits can only be undone in the order that they were applied. Effects can usually be adjusted during playback, or at any other time, edits may be undone or adjusted at any time in any order. Multiple effects and edits may be stacked so that they are applied to the audio as an effect chain, a stack of effects may be changed so that effects are applied in a different order, or effects inserted or removed from the chain. Some real-time editors support effect automation so that changes to effect parameters may be programmed to occur at specified times during audio playback, the waveform does not usually show the effect of processing until the audio has been mixed-down or bounced to another track. The number of effects that may be applied is limited by the processing power of the computer or editing hardware
21.
Data compression
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In signal processing, data compression, source coding, or bit-rate reduction involves encoding information using fewer bits than the original representation. Compression can be 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, the process of reducing the size of a data file is referred to as data compression. In the context of data transmission, it is called coding in opposition to channel coding. Compression is useful because it reduces resources required to store and transmit data, computational resources are consumed in the compression process and, usually, in the reversal of the process. Data compression is subject to a space–time complexity trade-off, Lossless data compression algorithms usually exploit statistical redundancy to represent data without losing any information, so that the process is reversible. Lossless compression is possible because most real-world data exhibits statistical redundancy, for example, an image may have areas of color that do not change over several pixels, instead of coding red pixel, red pixel. The data may be encoded as 279 red pixels and this is a basic example of run-length encoding, there are many schemes to reduce file size by eliminating redundancy. The Lempel–Ziv compression methods are among the most popular algorithms for lossless storage, DEFLATE is a variation on LZ optimized for decompression speed and compression ratio, but compression can be slow. DEFLATE is used in PKZIP, Gzip, and PNG, LZW is used in GIF images. LZ methods use a table-based compression model where table entries are substituted for repeated strings of data, for most LZ methods, this table is generated dynamically from earlier data in the input. The table itself is often Huffman encoded, current LZ-based coding schemes that perform well are Brotli and LZX. LZX is used in Microsofts CAB format, the best modern lossless compressors use probabilistic models, such as prediction by partial matching. The Burrows–Wheeler transform can also be viewed as a form of statistical modelling. The basic task of grammar-based codes is constructing a context-free grammar deriving a single string, sequitur and Re-Pair are practical grammar compression algorithms for which software is publicly available. In a further refinement of the use of probabilistic modelling. Arithmetic coding is a more modern coding technique that uses the mathematical calculations of a machine to produce a string of encoded bits from a series of input data symbols
22.
FLAC
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FLAC is an audio coding format for lossless compression of digital audio, and is also the name of the reference codec implementation. Digital audio compressed by FLACs algorithm can typically be reduced to 50–60% of its original size, FLAC is an open format with royalty-free licensing and a reference implementation which is free software. FLAC has support for metadata tagging, album art. 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 incorporation of FLAC under the Xiph. org banner. Xiph. org is behind other free compression formats such as Vorbis, Theora, Speex and Opus. Version 1.3.0 was released on 26 May 2013, at which point development was moved to the Xiph. org git repository. flac files, the reference implementation is free software. The source code for libFLAC and libFLAC++ is available under the BSD license, and the sources for flac, metaflac, in its stated goals, the FLAC project encourages its developers not to implement copy prevention features of any kind. FLAC supports only fixed-point samples, not floating-point and it can handle any PCM bit resolution from 4 to 32 bits per sample, any sampling rate from 1 Hz to 655,350 Hz in 1 Hz increments, and any number of channels from 1 to 8. Channels can be grouped in some cases, for stereo and 5.1 channel surround. FLAC uses CRC checksums for identifying corrupted frames when used in a streaming protocol, FLAC allows for a Rice parameter between 0 and 16. FLAC uses linear prediction to convert the audio samples, there are two steps, the predictor and the error coding. The predictor can be one of four types, the difference between the predictor and the actual sample data is calculated and is known as the residual. The residual is stored efficiently using Golomb-Rice coding and it also uses run-length encoding for blocks of identical samples, such as silent passages. For tagging, FLAC uses the system as Vorbis comments. The libFLAC API is organized into streams, seekable streams, most FLAC applications will generally restrict themselves to encoding/decoding using libFLAC at the file level interface. LibFLAC uses a compression level parameter that varies from 0 to 8, the compressed files are always perfect, lossless representations of the original data. Although the compression process involves a tradeoff between speed and size, the process is always quite fast and not very dependent on the level of compression
23.
Image editing
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Image editing encompasses the processes of altering images, whether they are digital photographes, traditional photochemical photographs, or illustrations. Traditional analog image editing is known as photo retouching, using such as an airbrush to modify photographs. Many image editing programs are used to render or create computer art from scratch. Raster images are stored in a computer in the form of a grid of picture elements and these pixels contain the images color and brightness information. Image editors can change the pixels to enhance the image in many ways, the pixels can be changed as a group, or individually, by the sophisticated algorithms within the image editors. This article mostly refers to bitmap graphics editors, which are used to alter photographs. It is easier to rasterize a vector image than to vectorize a raster image, vector images can be modified more easily, because they contain descriptions of the shapes for easy rearrangement. They are also scalable, being rasterizable at any resolution and these are called automatic because generally they happen without user interaction or are offered with one click of a button or mouse button or by selecting an option from a menu. Additionally, some automatic editing features offer a combination of editing actions with little or no user interaction, many image file formats use data compression to reduce file size and save storage space. Digital compression of images may take place in the camera, or can be done in the computer with the image editor, when images are stored in JPEG format, compression has already taken place. Both cameras and computer programs allow the user to set the level of compression, some compression algorithms, such as those used in PNG file format, are lossless, which means no information is lost when the file is saved. JPEG uses knowledge of the way the brain and eyes perceive color to make this loss of detail less noticeable. Listed below are some of the most used capabilities of the better graphic manipulation programs, the list is by no means all inclusive. There are a myriad of choices associated with the application of most of these features, one of the prerequisites for many of the applications mentioned below is a method of selecting part of an image, thus applying a change selectively without affecting the entire picture. The border of an area in an image is often animated with the marching ants effect to help the user to distinguish the selection border from the image background. Image editors can resize images in a process often called image scaling, making them larger, High image resolution cameras can produce large images which are often reduced in size for Internet use. Image editor programs use a process called resampling to calculate new pixel values whose spacing is larger or smaller than the original pixel values. Images for Internet use are kept small, say 640 x 480 pixels which would equal 0.3 megapixels, digital editors are used to crop images
24.
Video editing
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The term video editing can refer to, The process of manipulating video images. Once the province of expensive machines called video editors, video editing software is now available for personal computers, video editing includes cutting segments, re-sequencing clips, and adding transitions and other Special Effects. Linear video editing, using tape and is edited in a very linear way. Several video clips from different tapes are recorded to one single tape in the order that they will appear, non-linear editing system, This is edited on computers with specialised software. These are non destructive to the video being edited and use such as Adobe Premiere Pro, Final Cut Pro. Offline editing is the process in which raw footage is copied from an original source, once the editing has been completely edited, the original media is then re-assembled in the online editing stage. Online editing is the process of reassembling the edit to full resolution video after an offline edit has been performed and is done in the stage of a video production. Vision mixing, when working within live television and video production environments, a vision mixer is used to cut live feed coming from several cameras in real time. Video editing is the process of editing segments of video production footage, special effects. Using video, a director can communicate non-fictional and fictional events, the goals of editing is to manipulate these events to bring the communication closer to the original goal or target. The two pieces of tape to be joined were painted with a solution of fine iron filings suspended in carbon tetrachloride. This developed the magnetic tracks, making them visible when viewed through a microscope so that they could be aligned in a designed for this task. If a scene closer to the beginning of the video tape needed to be changed in length, in addition, sources could be played back simultaneously through a vision mixer to create more complex transitions between scenes. A popular 1970-80s system for doing that was the U-matic equipment and that system used two tape players and one tape recorder, and edits were done by automatically having the machines back up, then speed up together in synchrony, so the edit didnt roll or glitch. Later, 1980-90s came the smaller beta equipment, and more complex controllers, content is ingested and recorded natively with the appropriate codec which will be used by video editing software to manipulate the captured footage. High-definition video is becoming popular and can be readily edited using the same video editing software along with related motion graphics programs. Video clips are arranged on a timeline, music tracks, titles, digital on-screen graphics are added, special effects can be created, and the finished program is rendered into a finished video. The video may then be distributed in a variety of ways including DVD, web streaming, QuickTime Movies, iPod, CD-ROM, like many other technologies, the cost of video editing has declined by an order of magnitude or more
25.
Mobile phone
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A mobile phone is a portable telephone that can make and receive calls over a radio frequency link while the user is moving within a telephone service area. The radio frequency link establishes a connection to the systems of a mobile phone operator. Most modern mobile telephone services use a network architecture, and, therefore. Mobile phones which offer these and more general computing capabilities are referred to as smartphones, the first handheld mobile phone was demonstrated by John F. Mitchell and Martin Cooper of Motorola in 1973, using a handset weighing c.4.4 lbs. In 1983, the DynaTAC 8000x was the first commercially available mobile phone. From 1983 to 2014, worldwide mobile phone subscriptions grew to seven billion, penetrating 100% of the global population. In first quarter of 2016, the top smartphone manufacturers were Samsung, Apple, a handheld mobile radio telephone service was envisioned in the early stages of radio engineering. In 1917, Finnish inventor Eric Tigerstedt filed a patent for a pocket-size folding telephone with a thin carbon microphone. Early predecessors of cellular phones included analog radio communications from ships, the race to create truly portable telephone devices began after World War II, with developments taking place in many countries. These 0G systems were not cellular, supported few simultaneous calls, the first handheld mobile phone was demonstrated by John F. Mitchell and Martin Cooper of Motorola in 1973, using a handset weighing c.4.4 lbs. The first commercial automated cellular network was launched in Japan by Nippon Telegraph and this was followed in 1981 by the simultaneous launch of the Nordic Mobile Telephone system in Denmark, Finland, Norway, and Sweden. Several other countries followed in the early to mid-1980s. These first-generation systems could support far more simultaneous calls but still used analog cellular technology, in 1983, the DynaTAC 8000x was the first commercially available handheld mobile phone. In 1991, the digital cellular technology was launched in Finland by Radiolinja on the GSM standard. This sparked competition in the sector as the new operators challenged the incumbent 1G network operators, ten years later, in 2001, the third generation was launched in Japan by NTT DoCoMo on the WCDMA standard. This was followed by 3. 5G, 3G+ or turbo 3G enhancements based on the high-speed packet access family, allowing UMTS networks to have data transfer speeds. By 2009, it had become clear that, at point, 3G networks would be overwhelmed by the growth of bandwidth-intensive applications. Consequently, the industry began looking to data-optimized fourth-generation technologies, with the promise of speed improvements up to ten-fold over existing 3G technologies
26.
Digital cameras
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A digital camera or digicam is a camera that produces digital images that can be stored in a computer, displayed on a screen and printed. Most cameras sold today are digital, and digital cameras are incorporated into many devices ranging from PDAs, Digital and movie cameras share an optical system, typically using a lens with a variable diaphragm to focus light onto an image pickup device. The diaphragm and shutter admit the correct amount of light to the imager, just as with film, however, unlike film cameras, digital cameras can display images on a screen immediately after being recorded, and store and delete images from memory. Many digital cameras can also record moving videos with sound, some digital cameras can crop and stitch pictures and perform other elementary image editing. The history of the camera began with Eugene F. Lally of the Jet Propulsion Laboratory. His 1961 idea was to take pictures of the planets and stars while travelling through space to give information about the astronauts position, unfortunately, as with Texas Instruments employee Willis Adcocks filmless camera in 1972, the technology had yet to catch up with the concept. Steven Sasson as an engineer at Eastman Kodak invented and built the first electronic camera using a charge-coupled device image sensor in 1975, earlier ones used a camera tube, later ones digitized the signal. Early uses were military and scientific, followed by medical. In the mid to late 1990s digital cameras became common among consumers, by the mid-2000s digital cameras had largely replaced film cameras, and higher-end cell phones had an integrated digital camera. By the beginning of the 2010s, almost all smartphones had a digital camera. The two major types of image sensor are CCD and CMOS. A CCD sensor has one amplifier for all the pixels, while each pixel in a CMOS active-pixel sensor has its own amplifier, compared to CCDs, CMOS sensors use less power. Cameras with a small sensor use a back-side-illuminated CMOS sensor, overall final image quality is more dependent on the image processing capability of the camera, than on sensor type. The resolution of a camera is often limited by the image sensor that turns light into that discrete signals. The brighter the image at a point on the sensor. Depending on the structure of the sensor, a color filter array may be used. The number of pixels in the sensor determines the cameras pixel count, in a typical sensor, the pixel count is the product of the number of rows and the number of columns. For example, a 1,000 by 1,000 pixel sensor would have 1,000,000 pixels, final quality of an image depends on all optical transformations in the chain of producing the image
27.
Multimedia Messaging Service
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Multimedia Messaging Service is a standard way to send messages that include multimedia content to and from a mobile phone over a cellular network. Users and providers may refer to such a message as a PXT, the MMS standard extends the core SMS capability, allowing the exchange of text messages greater than 160 characters in length. Unlike text-only SMS, MMS can deliver a variety of media, including up to forty seconds of video, one image, the most common use involves sending photographs from camera-equipped handsets. The 3GPP and WAP groups fostered the development of the MMS standard, early MMS deployments were plagued by technical issues and frequent consumer disappointments. In recent years, MMS deployment by major companies have solved many of the early challenges through handset detection, content optimization. China was one of the markets to make MMS a major commercial success, partly as the penetration rate of personal computers was modest. The chairman and CEO of China Mobile said at the GSM Association Mobile Asia Congress in 2009 that MMS in China was now a service on par with SMS text messaging. Europes most advanced MMS market has been Norway, and in 2008, Norwegian mobile subscribers sent on average one MMS per week. Between 2010 and 2013, MMS traffic in the U. S. increased by 70% from 57 billion to 96 billion messages sent and this is due in part to the wide adoption of smartphones. MMS messages are delivered in a different way from SMS, the first step is for the sending device to encode the multimedia content in a fashion similar to sending a MIME message. The message is forwarded to the carriers MMS store and forward server. If the receiver is on a different from the sender, then the MMSC acts as a relay. Once the recipients MMSC has received a message, it first determines whether the handset is MMS capable. If so, the content is extracted and sent to a storage server with an HTTP front-end. An SMS control message containing the URL of the content is sent to the recipients handset to trigger the receivers WAP browser to open. Several other messages are exchanged to indicate the status of the delivery attempt, before delivering content, some MMSCs also include a conversion service that will attempt to modify the multimedia content into a format suitable for the receiver. This is known as content adaptation, if the receivers handset is not MMS capable, the message is usually delivered to a web-based service from where the content can be viewed from a normal internet browser. The URL for the content is sent to the receivers phone in a normal text message
28.
Pixel
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The address of a pixel corresponds to its physical coordinates. LCD pixels are manufactured in a grid, and are often represented using dots or squares. Each pixel is a sample of an image, more samples typically provide more accurate representations of the original. The intensity of each pixel is variable, in color imaging systems, a color is typically represented by three or four component intensities such as red, green, and blue, or cyan, magenta, yellow, and black. The word pixel is based on a contraction of pix and el, the word pixel was first published in 1965 by Frederic C. Billingsley of JPL, to describe the elements of video images from space probes to the Moon. Billingsley had learned the word from Keith E. McFarland, at the Link Division of General Precision in Palo Alto, McFarland said simply it was in use at the time. The word is a combination of pix, for picture, the word pix appeared in Variety magazine headlines in 1932, as an abbreviation for the word pictures, in reference to movies. By 1938, pix was being used in reference to pictures by photojournalists. The concept of a picture element dates to the earliest days of television, some authors explain pixel as picture cell, as early as 1972. In graphics and in image and video processing, pel is often used instead of pixel, for example, IBM used it in their Technical Reference for the original PC. A pixel is generally thought of as the smallest single component of a digital image, however, the definition is highly context-sensitive. For example, there can be printed pixels in a page, or pixels carried by electronic signals, or represented by digital values, or pixels on a display device, or pixels in a digital camera. This list is not exhaustive and, depending on context, synonyms include pel, sample, byte, bit, dot, Pixels can be used as a unit of measure such as,2400 pixels per inch,640 pixels per line, or spaced 10 pixels apart. For example, a high-quality photographic image may be printed with 600 ppi on a 1200 dpi inkjet printer, even higher dpi numbers, such as the 4800 dpi quoted by printer manufacturers since 2002, do not mean much in terms of achievable resolution. The more pixels used to represent an image, the closer the result can resemble the original, the number of pixels in an image is sometimes called the resolution, though resolution has a more specific definition.3 megapixels. The pixels, or color samples, that form an image may or may not be in one-to-one correspondence with screen pixels. In computing, a composed of pixels is known as a bitmapped image or a raster image
