1.
Mobile telephony
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Mobile telephony is the provision of telephone services to phones which may move around freely rather than stay fixed in one location. Mobile phones connect to a cellular network of base stations. Both networks are interconnected to the switched telephone network to allow any phone in the world to be dialed. In 2010 there were estimated to be five billion mobile cellular subscriptions in the world. S, public mobile phone systems were first introduced in the years after the Second World War and made use of technology developed before and during the conflict. The first system opened in St Louis, Missouri, USA in 1946 whilst other countries followed in the succeeding decades, the UK introduced its System 1 manual radiotelephone service as the South Lancashire Radiophone Service in 1958. Calls were made via an operator using handsets identical to ordinary phone handsets, the phone itself was a large box located in the boot of the vehicle containing valves and other early electronic components. Although an uprated manual service was extended to cover most of the UK and these cellular systems were based on US Advanced Mobile Phone Service technology, the modified technology being named Total Access Communication System. In 1947 Bell Labs was the first to propose a cellular telephone network. In 1956 the MTA system was launched in Sweden, both problems were solved by Bell Labs employee Amos Joel who, in 1970 applied for a patent for a mobile communications system. However, a consulting firm calculated the entire U. S. As a consequence, Bell Labs concluded that the invention was of little or no consequence, the invention earned Joel induction into the National Inventors Hall of Fame in 2008. The first call on a mobile phone was made on April 3,1973 by Martin Cooper. Bell Labs went on to install the first trial cellular network in Chicago in 1978. This trial system was licensed by the FCC to ATT for commercial use in 1982 and, as part of the arrangements for the breakup of ATT. The first commercial system opened in Chicago in October 1983, a system designed by Motorola also operated in the Washington D. C. /Baltimore area from summer 1982 and became a full public service later the following year. Japans first commercial service was launched by NTT in 1978. The first fully automatic first generation system was the Nordic Mobile Telephone system, simultaneously launched in 1981 in Denmark, Finland, Norway. NMT was the first mobile phone network featuring international roaming, the Swedish electrical engineer Östen Mäkitalo started to work on this vision in 1966, and is considered as the father of the NMT system and some consider him also the father of the cellular phone
2.
Vocoder
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A vocoder is a category of voice codec that analyzes and synthesizes the human voice signal for audio data compression, multiplexing, voice encryption, voice transformation, etc. The decoder applies these control signals to corresponding filters for re-synthesis, since these control signals change only slowly compared to the original speech waveform, the bandwidth required to transmit speech can be reduced. This allows more speech channels to share a single communication channel, by encrypting the control signals, voice transmission can be secured against interception. Its primary use in fashion is for secure radio communication. The advantage of this method of encryption is that none of the signal is sent. The receiving unit needs to be set up in the same filter configuration to re-synthesize a version of the signal spectrum. The vocoder has also used extensively as an electronic musical instrument. The decoder portion of the vocoder, called a voder, can be used independently for speech synthesis, the human voice consists of sounds generated by the opening and closing of the glottis by the vocal cords, which produces a periodic waveform with many harmonics. This basic sound is filtered by the nose and throat to produce differences in harmonic content in a controlled way. There is another set of sounds, known as the unvoiced and plosive sounds, the vocoder examines speech by measuring how its spectral characteristics change over time. This results in a series of signals representing these modified frequencies at any time as the user speaks. In simple terms, the signal is split into a number of frequency bands, thus, the vocoder dramatically reduces the amount of information needed to store speech, from a complete recording to a series of numbers. Information about the frequency of the original voice signal is discarded. It is this aspect of the vocoding process that has made it useful in creating special voice effects in popular music. Analog vocoders typically analyze an incoming signal by splitting the signal into a number of tuned frequency bands or ranges, a modulator and carrier signal are sent through a series of these tuned bandpass filters. In the example of a robot voice, the modulator is a microphone. There are usually between eight and 20 bands, the amplitude of the modulator for each of the individual analysis bands generates a voltage that is used to control amplifiers for each of the corresponding carrier bands. The result is that components of the modulating signal are mapped onto the carrier signal as discrete amplitude changes in each of the frequency bands
3.
Videotelephony
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Videotelephony comprises the technologies for the reception and transmission of audio-video signals by users at different locations, for communication between people in real-time. A videophone is a telephone with a display, capable of simultaneous video. Videoconferencing implies the use of technology for a group or organizational meeting rather than for individuals. Telepresence may refer either to a high-quality videotelephony system or to meetup technology which goes beyond video into robotics, Videoconferencing has also been called visual collaboration and is a type of groupware. It is also used in commercial and corporate settings to facilitate meetings and conferences, Simple analog videophone communication could be established as early as the invention of the television. Such an antecedent usually consisted of two closed-circuit television systems connected via cable or radio. An example of that was the German Reich Postzentralamt video telephone network serving Berlin, the development of the crucial video technology first started in the latter half of the 1920s in the United Kingdom and the United States, spurred notably by John Logie Baird and AT&Ts Bell Labs. This occurred in part, at least with AT&T, to serve as an adjunct supplementing the use of the telephone, a number of organizations believed that videotelephony would be superior to plain voice communications. However video technology was to be deployed in analog television broadcasting long before it could become practical—or popular—for videophones, during the first manned space flights, NASA used two radio-frequency video links, one in each direction. TV channels routinely use this type of videotelephony when reporting from distant locations, the news media were to become regular users of mobile links to satellites using specially equipped trucks, and much later via special satellite videophones in a briefcase. This technique was very expensive, though, and could not be used for such as telemedicine, distance education. Videotelephony developed in parallel with conventional voice telephone systems from the mid-to-late 20th century, Only in the late 20th century with the advent of powerful video codecs combined with high-speed Internet broadband and ISDN service did videotelephony become a practical technology for regular use. In the 1980s, digital telephony transmission networks became possible, such as with ISDN networks, assuring a bit rate for compressed video. During this time, there was research into other forms of digital video. Many of these technologies, such as the Media space, are not as used today as videoconferencing but were still an important area of research. The first dedicated systems started to appear in the market as ISDN networks were expanding throughout the world, one of the first commercial videoconferencing systems sold to companies came from PictureTel Corp. which had an Initial Public Offering in November,1984. The company also secured a patent for a codec for full-motion videoconferencing, in 1992 CU-SeeMe was developed at Cornell by Tim Dorcey et al. In 1995 the first public videoconference between North America and Africa took place, linking a technofair in San Francisco with a techno-rave and cyberdeli in Cape Town
4.
Free software
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The right to study and modify software entails availability of the software source code to its users. This right is conditional on the person actually having a copy of the software. Richard Stallman used the existing term free software when he launched the GNU Project—a collaborative effort to create a freedom-respecting operating system—and the Free Software Foundation. The FSFs Free Software Definition states that users of software are free because they do not need to ask for permission to use the software. Free software thus differs from proprietary software, such as Microsoft Office, Google Docs, Sheets, and Slides or iWork from Apple, which users cannot study, change, freeware, which is a category of freedom-restricting proprietary software that does not require payment for use. For computer programs that are covered by law, software freedom is achieved with a software license. Software that is not covered by law, such as software in the public domain, is free if the source code is in the public domain. Proprietary software, including freeware, use restrictive software licences or EULAs, Users are thus prevented from changing the software, and this results in the user relying on the publisher to provide updates, help, and support. This situation is called vendor lock-in, Users often may not reverse engineer, modify, or redistribute proprietary software. Other legal and technical aspects, such as patents and digital rights management may restrict users in exercising their rights. Free software may be developed collaboratively by volunteer computer programmers or by corporations, as part of a commercial, from the 1950s up until the early 1970s, it was normal for computer users to have the software freedoms associated with free software, which was typically public domain software. Software was commonly shared by individuals who used computers and by manufacturers who welcomed the fact that people were making software that made their hardware useful. Organizations of users and suppliers, for example, SHARE, were formed to exchange of software. As software was written in an interpreted language such as BASIC. Software was also shared and distributed as printed source code in computer magazines and books, in United States vs. IBM, filed January 17,1969, the government charged that bundled software was anti-competitive. While some software might always be free, there would henceforth be an amount of software produced primarily for sale. In the 1970s and early 1980s, the industry began using technical measures to prevent computer users from being able to study or adapt the software as they saw fit. In 1980, copyright law was extended to computer programs, Software development for the GNU operating system began in January 1984, and the Free Software Foundation was founded in October 1985
5.
Speech synthesis
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Speech synthesis is the artificial production of human speech. A computer system used for purpose is called a speech computer or speech synthesizer. A text-to-speech system converts normal language text into speech, other systems render symbolic linguistic representations like phonetic transcriptions into speech, synthesized speech can be created by concatenating pieces of recorded speech that are stored in a database. Systems differ in the size of the stored speech units, a system that stores phones or diphones provides the largest output range, for specific usage domains, the storage of entire words or sentences allows for high-quality output. Alternatively, a synthesizer can incorporate a model of the vocal tract, the quality of a speech synthesizer is judged by its similarity to the human voice and by its ability to be understood clearly. An intelligible text-to-speech program allows people with visual impairments or reading disabilities to listen to words on a home computer. Many computer operating systems have included speech synthesizers since the early 1990s, a text-to-speech system is composed of two parts, a front-end and a back-end. The front-end has two major tasks, first, it converts raw text containing symbols like numbers and abbreviations into the equivalent of written-out words. This process is called text normalization, pre-processing, or tokenization. The front-end then assigns phonetic transcriptions to each word, and divides and marks the text into prosodic units, like phrases, clauses, the process of assigning phonetic transcriptions to words is called text-to-phoneme or grapheme-to-phoneme conversion. Phonetic transcriptions and prosody information together make up the symbolic representation that is output by the front-end. The back-end—often referred to as the converts the symbolic linguistic representation into sound. In certain systems, this includes the computation of the target prosody. Long before the invention of electronic signal processing, some tried to build machines to emulate human speech. Some early legends of the existence of Brazen Heads involved Pope Silvester II, Albertus Magnus, there followed the bellows-operated acoustic-mechanical speech machine of Wolfgang von Kempelen of Pressburg, Hungary, described in a 1791 paper. This machine added models of the tongue and lips, enabling it to produce consonants as well as vowels, in 1837, Charles Wheatstone produced a speaking machine based on von Kempelens design, and in 1846, Joseph Faber exhibited the Euphonia. In 1923 Paget resurrected Wheatstones design, in the 1930s Bell Labs developed the vocoder, which automatically analyzed speech into its fundamental tones and resonances. From his work on the vocoder, Homer Dudley developed a keyboard-operated voice-synthesizer called The Voder, dr. Franklin S. Cooper and his colleagues at Haskins Laboratories built the Pattern playback in the late 1940s and completed it in 1950
6.
Digital audio
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Digital audio is technology that can be used to record, store, generate, manipulate, and reproduce sound using audio signals that have been encoded in digital form. A microphone converts sound to an electrical signal, then an analog-to-digital converter —typically using pulse-code modulation—converts the analog signal into a digital signal. This digital signal can then be recorded, edited and modified using digital audio tools, digital audio systems may include compression, storage, processing and transmission components. Conversion to a digital format allows convenient manipulation, storage, transmission, modern online music distribution depends on digital recording and data compression. The availability of music as data files, rather than as objects, has significantly reduced the costs of distribution. Before digital audio, the music industry distributed and sold music by selling copies in the form of records. With digital audio and online distribution systems such as iTunes, companies sell digital files to consumers. This digital audio/Internet distribution model is less expensive than producing physical copies of recordings, packaging them. An analog audio system captures sounds, and converts their physical waveforms into electrical representations of those waveforms by use of a transducer, the sounds are then stored, as on tape, or transmitted. The process is reversed for playback, the signal is amplified. Analog audio retains its fundamental wave-like characteristics throughout its storage, transformation, duplication, analog audio signals are susceptible to noise and distortion, due to the innate characteristics of electronic circuits and associated devices. Disturbances in a system do not result in error unless the disturbance is so large as to result in a symbol being misinterpreted as another symbol or disturb the sequence of symbols. A digital audio signal may be encoded for correction of any errors that occur in the storage or transmission of the signal. This technique, known as channel coding, is essential for broadcast or recorded digital systems to maintain bit accuracy, the discrete time and level of the binary signal allow a decoder to recreate the analog signal upon replay. Eight to Fourteen Bit Modulation is a code used in the audio Compact Disc. A digital audio system starts with an ADC that converts a signal to a digital signal. The ADC runs at a sampling rate and converts at a known bit resolution. CD audio, for example, has a rate of 44.1 kHz
7.
UMTS
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The Universal Mobile Telecommunications System is a third generation mobile cellular system for networks based on the GSM standard. UMTS uses wideband code division multiple access radio access technology to offer greater spectral efficiency, UMTS specifies a complete network system, which includes the radio access network, the core network and the authentication of users via SIM cards. The technology described in UMTS is sometimes referred to as Freedom of Mobile Multimedia Access or 3GSM. Unlike EDGE and CDMA2000, UMTS requires new base stations and new frequency allocations, UMTS supports maximum theoretical data transfer rates of 42 Mbit/s when Evolved HSPA is implemented in the network. Users in deployed networks can expect a transfer rate of up to 384 kbit/s for Release 99 handsets, since 2006, UMTS networks in many countries have been or are in the process of being upgraded with High-Speed Downlink Packet Access, sometimes known as 3. 5G. Currently, HSDPA enables downlink transfer speeds of up to 21 Mbit/s, work is also progressing on improving the uplink transfer speed with the High-Speed Uplink Packet Access. The first national consumer UMTS networks launched in 2002 with a emphasis on telco-provided mobile applications such as mobile TV. UMTS combines three different terrestrial air interfaces, GSMs Mobile Application Part core, and the GSM family of speech codecs, the air interfaces are called UMTS Terrestrial Radio Access. All air interface options are part of ITUs IMT-2000, in the currently most popular variant for cellular mobile telephones, W-CDMA is used. Please note that the terms W-CDMA, TD-CDMA and TD-SCDMA are misleading, while they suggest covering just a channel access method, they are actually the common names for the whole air interface standards. W-CDMA or WCDMA, along with UMTS-FDD, UTRA-FDD, or IMT-2000 CDMA Direct Spread is an air interface found in 3G mobile telecommunications networks. W-CDMA uses the DS-CDMA channel access method with a pair of 5 MHz wide channels, in contrast, the competing CDMA2000 system uses one or more available 1.25 MHz channels for each direction of communication. W-CDMA systems are criticized for their large spectrum usage, which delayed deployment in countries that acted relatively slowly in allocating new frequencies specifically for 3G services. The specific frequency bands originally defined by the UMTS standard are 1885–2025 MHz for the mobile-to-base, in the US, 1710–1755 MHz and 2110–2155 MHz are used instead, as the 1900 MHz band was already used. Some carriers such as T-Mobile use band numbers to identify the UMTS frequencies, in the late 1990s, W-CDMA was developed by NTT DoCoMo as the air interface for their 3G network FOMA. Later NTT DoCoMo submitted the specification to the International Telecommunication Union as a candidate for the international 3G standard known as IMT-2000. The ITU eventually accepted W-CDMA as part of the IMT-2000 family of 3G standards, as an alternative to CDMA2000, EDGE, later, W-CDMA was selected as an air interface for UMTS. As NTT DoCoMo did not wait for the finalisation of the 3G Release 99 specification, however, this has been resolved by NTT DoCoMo updating their network
8.
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
9.
Secure voice
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Secure voice is a term in cryptography for the encryption of voice communication over a range of communication types such as radio, telephone or IP. The implementation of voice encryption dates back to World War II when secure communication was paramount to the US armed forces, during that time, noise was simply added to a voice signal to prevent enemies from listening to the conversations. Noise was added by playing a record of noise in synch with the signal and when the voice signal reached the receiver. In order to out the noise, the receiver need to have exactly the same noise signal. Having only two copies of records made it impossible for the receiver to decrypt the signal. To implement the system, the army contracted Bell Laboratories and they developed a system called SIGSALY, with SIGSALY, ten channels were used to sample the voice frequency spectrum from 250 Hz to 3 kHz and two channels were allocated to sample voice pitch and background hiss. In the time of SIGSALY, the transistor had not been developed, each SIGSALY terminal used 40 racks of equipment weighing 55 tons and filled a large room. This equipment included radio transmitters and receivers and large phonograph turntables, the voice was keyed to two 16-inch vinyl phonograph records that contained a Frequency Shift Keying audio tone. The records were played on large precise turntables in synch with the voice transmission, from the introduction of voice encryption to today, encryption techniques have evolved drastically. Digital technology has effectively replaced old analog methods of encryption and by using complex algorithms, voice encryption has become much more secure. One relatively modern voice encryption method is Sub-band coding, with Sub-band Coding, the voice signal is split into multiple frequency bands, using multiple bandpass filters that cover specific frequency ranges of interest. The output signals from the bandpass filters are then translated to reduce the bandwidth. The lowpass signals are quantized and encoded using special techniques like. After the encoding stage, the signals are multiplexed and sent out along the communication network, when the signal reaches the receiver, the inverse operations are applied to the signal to get it back to its original state. A speech scrambling system was developed at Bell Laboratories in the 1970s by Subhash Kak, in this system permutation matrices were used to scramble coded representations of the speech data. Motorola developed an encryption system called Digital Voice Protection as part of their first generation of voice encryption techniques. DVP uses an encryption technique known as cipher feedback. The basic DVP algorithm is capable of 2.36 x 1021 different keys based on a key length of 32 bits, the extremely high number of possible keys associated with the early DVP algorithm, makes the algorithm very robust and gives a high level of security
10.
Code-division multiple access
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Code division multiple access is a channel access method used by various radio communication technologies. CDMA is an example of access, where several transmitters can send information simultaneously over a single communication channel. This allows several users to share a band of frequencies, to permit this without undue interference between the users, CDMA employs spread-spectrum technology and a special coding scheme. CDMA is used as the method in many mobile phone standards. The technology of code division multiple access channels has long been known, in the Soviet Union, the first work devoted to this subject was published in 1935 by professor Dmitriy V. Ageev. It was shown that through the use of methods, there are three types of signal separation, frequency, time and compensatory. LK-1 has a weight of 3 kg, 20–30 km operating distance, the base station, as described by the author, could serve several customers. In 1958, Kupriyanovich made the new experimental pocket model of mobile phone, to serve more customers, Kupriyanovich proposed the device, named by him as correllator. In 1958, the USSR also started the development of the Altai national civil mobile phone service for cars, the phone system weighed 11 kg. It was placed in the trunk of the vehicles of high-ranking officials, the main developers of the Altai system were VNIIS and GSPI. In 1963 this service started in Moscow and in 1970 Altai service was used in 30 USSR cities, one of the early applications for code division multiplexing is in the Global Positioning System. This predates and is distinct from its use in mobile phones, the Qualcomm standard IS-95, marketed as cdmaOne. The Qualcomm standard IS-2000, known as CDMA2000, is used by mobile phone companies. The UMTS 3G mobile phone standard, which uses W-CDMA, CDMA has been used in the OmniTRACS satellite system for transportation logistics. CDMA is a multiple access technique. A spread spectrum technique spreads the bandwidth of the data uniformly for the transmitted power. A spreading code is a code that has a narrow ambiguity function. In CDMA a locally generated code runs at a higher rate than the data to be transmitted
