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m developed in 1984 it is used in the gif format introduced in 1987 38 deflate a lossless compression algorithm specified in 1996 is used in the portable network graphics png format 39 wavelet compression the use of wavelets in image compression began after the development of dct coding 40 the jpeg 2000 standard was introduced in 2000 41 in contrast to the dct algorithm used by the original jpeg format jpeg 2000 instead uses discrete wavelet transform dwt algorithms 42 43 44 jpeg 2000 technology which includes the motion jpeg 2000 extension was selected as the video coding standard for digital cinema in 2004 45 audio edit see also audio coding format and audio codec audio data compression not to be confused with dynamic range compression has the potential to reduce the transmission bandwidth and storage requirements of audio data audio compression formats compression algorithms are implemented in software as audio codecs in both lossy and lossless compression information redundancy is reduced using methods such as coding quantization dct and linear prediction to reduce the amount of information used to represent the uncompressed data lossy audio compression algorithms provide higher compression and are used in numerous audio applications including vorbis and mp3 these algorithms almost all rely on psychoacoustics to eliminate or reduce fidelity of less audible sounds thereby reducing the space required to store or transmit them 2 46 the acceptable trade off between loss of audio quality and transmission or storage size depends upon the application for example one 640 mb compact disc cd holds approximately one hour of uncompressed high fidelity music less than 2 hours of music compressed losslessly or 7 hours of music compressed in the mp3 format at a medium bit rate a digital sound recorder can typically store around 200 hours of clearly intelligible speech in 640 mb 47 lossless audio compression produces a representation of digital data that can be decoded to an exact digital duplicate of the original compression ratios are around 50 60 of the original size 48 which is similar to those for generic lossless data compression lossless codecs use curve fitting or linear prediction as a basis for estimating the signal parameters describing the estimation and the difference between the estimation and the actual signal are coded separately 49 a number of lossless audio compression formats exist see list of lossless codecs for a listing some formats are associated with a distinct system such as direct stream transfer used in super audio cd and meridian lossless packing used in dvd audio dolby truehd blu ray and hd dvd some audio file formats feature a combination of a lossy format and a lossless correction this allows stripping the correction to easily obtain a lossy file such formats include mpeg 4 sls scalable to lossless wavpack and optimfrog dualstream when audio files are to be processed either by further compression or for editing it is desirable to work from an unchanged original uncompressed or losslessly compressed processing of a lossily compressed file for some purpose usually produces a final result inferior to the creation of the same compressed file from an uncompressed original in addition to sound editing or mixing lossless audio compression is often used for archival storage or as master copies lossy audio compression edit comparison of spectrograms of audio in an uncompressed format and several lossy formats the lossy spectrograms show bandlimiting of higher frequencies a common technique associated with lossy audio compression lossy audio compression is used in a wide range of applications in addition to standalone audio only applications of file playback in mp3 players or computers digitally compressed audio streams are used in most video dvds digital television streaming media on the internet satellite and cable radio and increasingly in terrestrial radio broadcasts lossy compression typically achieves far greater compression than lossless compression by discarding less critical data based on psychoacoustic optimizations 50 psychoacoustics recognizes that not all data in an audio stream can be perceived by the human auditory system most lossy compression reduces redundancy by first identifying perceptually irrelevant sounds that is sounds that are very hard to hear typical examples include high frequencies or sounds that occur at the same time as louder sounds those irrelevant sounds are coded with decreased accuracy or not at all due to the nature of lossy algorithms audio quality suffers a digital generation loss when a file is decompressed and recompressed this makes lossy compression unsuitable for storing the intermediate results in professional audio engineering applications such as sound editing and multitrack recording however lossy formats such as mp3 are very popular with end users as the file size is reduced to 5 20 of the original size and a megabyte can store about a minute s worth of music at adequate quality several proprietary lossy compression algorithms have been developed that provide higher quality audio performance by using a combination of lossless and lossy algorithms with adaptive bit rates and lower compression ratios examples include aptx ldac lhdc mqa and scl6 coding methods edit to determine what information in an audio signal is perceptually irrelevant most lossy compression algorithms use transforms such as the modified discrete cosine transform mdct to convert time domain sampled waveforms into a transform domain typically the frequency domain once transformed component frequencies can be prioritized according to how audible they are audibility of spectral components is assessed using the absolute threshold of hearing and the principles of simultaneous masking the phenomenon wherein a signal is masked by another signal separated by frequency and in some cases temporal masking where a signal is masked by another signal separated by time equal loudness contours may also be used to weigh the perceptual importance of components models of the human ear brain combination incorporating such effects are often called psychoacoustic models 51 other types of lossy compressors such as the linear predictive coding lpc used with speech are source based coders lpc uses a model of the human vocal tract to analyze speech sounds and infer the parameters used by the model to produce them moment to moment these changing parameters are transmitted or stored and used to drive another model in the decoder which reproduces the sound lossy formats are often used for the distribution of streaming audio or interactive communication such as in cell phone networks in such applications the data must be decompressed as the data flows rather than after the entire data stream has been transmitted not all audio codecs can be used for streaming applications 50 latency is introduced by the methods used to encode and decode the data some codecs will analyze a longer segment called a frame of the data to optimize efficiency and then code it in a manner that requires a larger segment of data at one time to decode the inherent latency of the coding algorithm can be critical for example when there is a two way transmission of data such as with a telephone conversation significant delays may seriously degrade the perceived quality in contrast to the speed of compression which is proportional to the number of operations required by the algorithm here latency refers to the number of samples that must be analyzed before a block of audio is processed in the minimum case latency is zero samples e g if the coder decoder simply reduces the number of bits used to quantize the signal time domain algorithms such as lpc also often have low latencies hence their popularity in speech coding for telephony in algorithms such as mp3 however a large number of samples have to be analyzed to implement a psychoacoustic model in the frequency domain and latency is on the order of 23 ms speech encoding edit speech encoding is an important category of audio data compression the perceptual models used to estimate what aspects of speech a human ear can hear are generally somewhat different from those used for music the range of frequencies needed to convey the sounds of a human voice is normally far narrower than that needed for music and the sound is normally less complex as a result speech can be encoded at high quality using a relatively low bit rate this is accomplished in general by some combination of two approaches only encoding sounds that could be made by a single human voice throwing away more of the data in the signal keeping just enough to reconstruct an intelligible voice rather than the full frequency range of human hearing the earliest algorithms used in speech encoding and audio data compression in general were the a law algorithm and the μ law algorithm history edit solidyne 922 the world s first commercial audio bit compression sound card for pc 1990 early audio research was conducted at bell labs there in 1950 c chapin cutler filed the patent on differential pulse code modulation dpcm 52 in 1973 adaptive dpcm adpcm was introduced by p cummiskey nikil s jayant and james l flanagan 53 54 perceptual coding was first used for speech coding compression with linear predictive coding lpc 55 initial concepts for lpc date back to the work of fumitada itakura nagoya university and shuzo saito nippon telegraph and telephone in 1966 56 during the 1970s bishnu s atal and manfred r schroeder at bell labs developed a form of lpc called adaptive predictive coding apc a perceptual coding algorithm that exploited the masking properties of the human ear followed in the early 1980s with the code excited linear prediction celp algorithm which achieved a significant compression ratio for its time 55 perceptual coding is used by modern audio compression formats such as mp3 55 and aac discrete cosine transform dct developed by nasir ahmed t natarajan and k r rao in 1974 16 provided the basis for the modified discrete cosine transform mdct used by modern audio compression formats such as mp3 57 dolby digital 58 59 and aac 60 mdct was proposed by j p princen a w johnson and a b bradley in 1987 61 following earlier work by princen and bradley in 1986 62 the world s first commercial broadcast automation audio compression system was developed by oscar bonello an engineering professor at the university of buenos aires 63 this broadcast automation system was launched in 1987 under the name audicom 64 65 a literature compendium for a large variety of audio coding systems was published in the ieee s journal on selected areas in communications jsac in february 1988 while there were some papers from before that time this collection documented an entire variety of finished working audio coders nearly all of them using perceptual techniques and some kind of frequency analysis and back end noiseless coding 66 video edit see also video coding format and video codec uncompressed video requires a very high data rate although lossless video compression codecs perform at a compression factor of 5 to 12 a typical h 264 lossy compression video has a compression factor between 20 and 200 67 the two key video compression techniques used in video coding standards are the dct and motion compensation mc most video coding standards such as the h 26x and mpeg formats typically use motion compensated dct video coding block motion compensation 68 69 most video codecs are used alongside audio compression techniques to store the separate but complementary data streams as one combined package using so called container formats 70 encoding theory edit video data may be represented as a series of still image frames such data usually contains abundant amounts of spatial and temporal redundancy video compression algorithms attempt to reduce redundancy and store information more compactly most video compression formats and codecs exploit both spatial and temporal redundancy e g through difference coding with motion compensation similarities can be encoded by only storing differences between e g temporally adjacent frames inter frame coding or spatially adjacent pixels intra frame coding inter frame compression a temporal delta encoding re uses data from one or more earlier or later frames in a sequence to describe the current frame intra frame coding on the other hand uses only data from within the current frame effectively being still image compression 51 the intra frame video coding formats used in camcorders and video editing employ simpler compression that uses only intra frame prediction this simplifies video editing software as it prevents a situation in which a compressed frame refers to data that the editor has deleted usually video compression additionally employs lossy compression techniques like quantization that reduce aspects of the source data that are more or less irrelevant to the human visual perception by exploiting perceptual features of human vision for example small differences in color are more difficult to perceive than are changes in brightness compression algorithms can average a color across these similar areas in a manner similar to those used in jpeg image compression 9 as in all lossy compression there is a trade off between video quality and bit rate cost of processing the compression and decompression and system requirements highly compressed video may present visible or distracting artifacts other methods other than the prevalent dct based transform formats such as fractal compression matching pursuit and the use of a discrete wavelet transform dwt have been the subject of some research but are typically not used in practical products wavelet compression is used in still image coders and video coders without motion compensation interest in fractal compression seems to be waning due to recent theoretical analysis showing a comparative lack of effectiveness of such methods 51 inter frame coding edit main article inter frame further information motion compensation in inter frame coding individual frames of a video sequence are compared from one frame to the next and the video compression codec records the differences to the reference frame if the frame contains areas where nothing has moved the system can simply issue a short command that copies that part of the previous frame into the next one if sections of the frame move in a simple manner the compressor can emit a slightly longer command that tells the decompressor to shift rotate lighten or darken the copy this longer command still remains much shorter than data generated by intra frame compression usually the encoder will also transmit a residue signal which describes the remaining more subtle differences to the reference imagery using entropy coding these residue signals have a more compact representation than the full signal in areas of video with more motion the compression must encode mo...
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