1.
Sound effect
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In motion picture and television production, a sound effect is a sound recorded and presented to make a specific storytelling or creative point without the use of dialogue or music. The term often refers to a process applied to a recording, in professional motion picture and television production, dialogue, music, and sound effects recordings are treated as separate elements. Dialogue and music recordings are never referred to as sound effects, even though the processes applied to such as reverberation or flanging effects, the term sound effect ranges back to the early days of radio. In its Year Book 1931 the BBC published an article about The Use of Sound Effects. It considers sounds effect deeply linked with broadcasting and states, It would be a mistake to think of them as anologous to punctuation marks. They should never be inserted into an already existing. The author of a broadcast play or broadcast construction ought to have used Sound Effects as bricks with which to build, treating them as of value with speech. If it fails to do so its presence could not be justified, the sound of people talking in the background is also considered a BG, but only if the speaker is unintelligible and the language is unrecognizable. These background noises are also called ambience or atmos, Foley sound effects are sounds that synchronize on screen, and require the expertise of a foley artist to record properly. Footsteps, the movement of hand props, and the rustling of cloth are common foley units, design sound effects are sounds that do not normally occur in nature, or are impossible to record in nature. These sounds are used to suggest futuristic technology in a fiction film. Each of these sound effect categories is specialized, with sound editors known as specialists in an area of sound effects, Foley is another method of adding sound effects. With this technique the action onscreen is essentially recreated to try to match it as closely as possible, if done correctly it is very hard for audiences to tell what sounds were added and what sounds were originally recorded. In the early days of film and radio, foley artists would add sounds in realtime or pre-recorded sound effects would be played back from analogue discs in realtime. Today, with effects held in digital format, it is easy to any required sequence to be played in any desired timeline. In the days of silent film, sound effects were added by the operator of an organ or photoplayer. Theater organ sound effects are usually electric or electro-pneumatic, and activated by a button pressed with the hand or foot, photoplayer operators activate sound effects either by flipping switches on the machine or pulling cow-tail pull-strings, which hang above. Sounds like bells and drums are made mechanically, sirens and horns electronically, due to its smaller size, a photoplayer usually has less special effects than a theater organ, or less complex ones
2.
Effects unit
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An effects unit or pedal is an electronic or digital device that alters how a musical instrument or other audio source sounds. In the 2010s, most effects use solid state electronics and/or computer chips, some vintage effects units from the 1930s to the 1970s and modern reissues of these effects use mechanical components as well or vacuum tubes. Musicians, audio engineers and record producers use effects units during live performances or in the studio, typically with electric guitar, electronic keyboard, while guitar effects are most frequently used with electric or electronic instruments, effects can also be used with acoustic instruments, drums and vocals. Rackmounted or audio console-integrated reverb effects are used with vocals in live sound and sound recording. Examples of common units include wah-wah pedals, fuzzboxes and reverb units. A stompbox or pedal is a metal or plastic box placed on the floor in front of the musician and connected to the instrument. Pedals are usually the least expensive format, a rackmount device mounts on a standard 19-inch equipment rack and usually contains several types of effects. Rackmount effects typically have buttons and/or knobs on the face of the chassis for controlling the effects, an effects unit is also called an effect box, effects device, effects processor or simply effects. In audio engineer parlance, a signal without effects is dry, the abbreviation F/X or FX is sometimes used. A pedal-style unit may be called a box, stompbox. When a musician has multiple effects in a rack mounted road case, Effects units are available in a variety of formats or form factors. Stompboxes are usually the smallest, least expensive, and most rugged effects units, rackmount devices are generally more expensive and offer a wider range of functions. An effects unit can consist of analog or digital circuitry or a combination of the two, during a live performance, the effect is plugged into the electrical signal path of the instrument. In the studio, the instrument or other sound-sources auxiliary output is patched into the effect, form factors are part of a studio or musicians outboard gear. Stompboxes are small plastic or metal chassis which usually lie on the floor or in a pedalboard to be operated by the users feet, pedals are often rectangle-shaped, but there are a range of other shapes. Typical simple stompboxes have a single footswitch, one to three potentiometers for controlling the effect, and a single LED that indicates if the effect is on, depending on the type of pedal, the potentiometers may control different parameters of the effect. For a chorus effect, for example, the knobs may control the depth, some pedals have two knobs stacked on top of each other, enabling the unit to provide two knobs per single knob space. An effects chain or signal chain is formed by connecting two or more stompboxes, effect chains are typically created between the guitar and the amp or between the preamplifier and the power amp
3.
Digital signal (signal processing)
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If that discrete set is finite, the discrete values can be represented with digital words of a finite width. Most commonly, these values are represented as fixed-point words or floating-point words. The process of analog-to-digital conversion produces a digital signal, common practical digital signals are represented as 8-bit, 16-bit, 24-bit and 32-bit. But the number of levels is not necessarily limited to powers of two
4.
Signal processing
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According to Alan V. Oppenheim and Ronald W. Schafer, the principles of signal processing can be found in the classical numerical analysis techniques of the 17th century. Oppenheim and Schafer further state that the digitalization or digital refinement of techniques can be found in the digital control systems of the 1940s and 1950s. Feature extraction, such as understanding and speech recognition. Quality improvement, such as reduction, image enhancement. Including audio compression, image compression, and video compression and this involves linear electronic circuits as well as non-linear ones. The former are, for instance, passive filters, active filters, additive mixers, integrators, non-linear circuits include compandors, multiplicators, voltage-controlled filters, voltage-controlled oscillators and phase-locked loops. Continuous-time signal processing is for signals that vary with the change of continuous domain, the methods of signal processing include time domain, frequency domain, and complex frequency domain. This technology was a predecessor of digital processing, and is still used in advanced processing of gigahertz signals. Digital signal processing is the processing of digitized discrete-time sampled signals, processing is done by general-purpose computers or by digital circuits such as ASICs, field-programmable gate arrays or specialized digital signal processors. Typical arithmetical operations include fixed-point and floating-point, real-valued and complex-valued, other typical operations supported by the hardware are circular buffers and look-up tables. Examples of algorithms are the Fast Fourier transform, finite impulse response filter, Infinite impulse response filter, nonlinear signal processing involves the analysis and processing of signals produced from nonlinear systems and can be in the time, frequency, or spatio-temporal domains. Nonlinear systems can produce complex behaviors including bifurcations, chaos, harmonics
5.
Sound wave
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In physics, sound is a vibration that propagates as a typically audible mechanical wave of pressure and displacement, through a transmission medium such as air or water. In physiology and psychology, sound is the reception of such waves, humans can hear sound waves with frequencies between about 20 Hz and 20 kHz. Sound above 20 kHz is ultrasound and below 20 Hz is infrasound, other animals have different hearing ranges. Acoustics is the science that deals with the study of mechanical waves in gases, liquids, and solids including vibration, sound, ultrasound. A scientist who works in the field of acoustics is an acoustician, an audio engineer, on the other hand, is concerned with the recording, manipulation, mixing, and reproduction of sound. Auditory sensation evoked by the oscillation described in, sound can propagate through a medium such as air, water and solids as longitudinal waves and also as a transverse wave in solids. The sound waves are generated by a source, such as the vibrating diaphragm of a stereo speaker. The sound source creates vibrations in the surrounding medium, as the source continues to vibrate the medium, the vibrations propagate away from the source at the speed of sound, thus forming the sound wave. At a fixed distance from the source, the pressure, velocity, at an instant in time, the pressure, velocity, and displacement vary in space. Note that the particles of the medium do not travel with the sound wave and this is intuitively obvious for a solid, and the same is true for liquids and gases. During propagation, waves can be reflected, refracted, or attenuated by the medium, the behavior of sound propagation is generally affected by three things, A complex relationship between the density and pressure of the medium. This relationship, affected by temperature, determines the speed of sound within the medium, if the medium is moving, this movement may increase or decrease the absolute speed of the sound wave depending on the direction of the movement. For example, sound moving through wind will have its speed of propagation increased by the speed of the if the sound and wind are moving in the same direction. If the sound and wind are moving in opposite directions, the speed of the wave will be decreased by the speed of the wind. Medium viscosity determines the rate at which sound is attenuated, for many media, such as air or water, attenuation due to viscosity is negligible. When sound is moving through a medium that does not have constant physical properties, the mechanical vibrations that can be interpreted as sound can travel through all forms of matter, gases, liquids, solids, and plasmas. The matter that supports the sound is called the medium, sound cannot travel through a vacuum. Sound is transmitted through gases, plasma, and liquids as longitudinal waves and it requires a medium to propagate
6.
Decibel
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The decibel is a logarithmic unit used to express the ratio of two values of a physical quantity. One of these values is often a reference value, in which case the decibel is used to express the level of the other value relative to this reference. When used in way, the decibel symbol is often qualified with a suffix that indicates the reference quantity that has been used or some other property of the quantity being measured. For example, dBm indicates a power of one milliwatt. There are two different scales used when expressing a ratio in decibels depending on the nature of the quantities, when expressing power quantities, the number of decibels is ten times the logarithm to base 10 of the ratio of two power quantities. That is, a change in power by a factor of 10 corresponds to a 10 dB change in level, when expressing field quantities, a change in amplitude by a factor of 10 corresponds to a 20 dB change in level. The difference in scales relates to the square law of fields in three-dimensional linear space. The decibel scales differ so that comparisons can be made between related power and field quantities when they are expressed in decibels. The definition of the decibel is based on the measurement of power in telephony of the early 20th century in the Bell System in the United States. One decibel is one tenth of one bel, named in honor of Alexander Graham Bell, however, today, the decibel is used for a wide variety of measurements in science and engineering, most prominently in acoustics, electronics, and control theory. In electronics, the gains of amplifiers, attenuation of signals, the decibel originates from methods used to quantify signal loss in telegraph and telephone circuits. The unit for loss was originally Miles of Standard Cable, the standard telephone cable implied was a cable having uniformly distributed resistance of 88 ohms per loop mile and uniformly distributed shunt capacitance of 0.054 microfarad per mile. 1 TU was defined such that the number of TUs was ten times the logarithm of the ratio of measured power to a reference power level. The definition was conveniently chosen such that 1 TU approximated 1 MSC, in 1928, the Bell system renamed the TU into the decibel, being one tenth of a newly defined unit for the base-10 logarithm of the power ratio. It was named the bel, in honor of the telecommunications pioneer Alexander Graham Bell, the bel is seldom used, as the decibel was the proposed working unit. However, the decibel is recognized by international bodies such as the International Electrotechnical Commission. The term field quantity is deprecated by ISO 80000-1, which favors root-power, in spite of their widespread use, suffixes are not recognized by the IEC or ISO. The ISO Standard 80000-3,2006 defines the following quantities, the decibel is one-tenth of a bel,1 dB =0.1 B
7.
Radio broadcasting
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Radio broadcasting is a unidirectional wireless transmission over radio waves intended to reach a wide audience. Stations can be linked in radio networks to broadcast a radio format. Audio broadcasting also can be done via radio, local wire television networks, satellite radio. The signal types can be either analog audio or digital audio, the earliest radio stations were simply radiotelegraphy systems and did not carry audio. For audio broadcasts to be possible, electronic detection and amplification devices had to be incorporated, the thermionic valve was invented in 1904 by the English physicist John Ambrose Fleming. He developed a device he called an oscillation valve, the heated filament, or cathode, was capable of thermionic emission of electrons that would flow to the plate when it was at a higher voltage. Electrons, however, could not pass in the direction because the plate was not heated. Later known as the Fleming valve, it could be used as a rectifier of alternating current and this greatly improved the crystal set which rectified the radio signal using an early solid-state diode based on a crystal and a so-called cats whisker. However, what was required was an amplifier. The triode was patented on March 4,1906, by the Austrian Robert von Lieben independent from that, on October 25,1906 and it wasnt put to practical use until 1912 when its amplifying ability became recognized by researchers. By about 1920, valve technology had matured to the point where radio broadcasting was quickly becoming viable, however, an early audio transmission that could be termed a broadcast may have occurred on Christmas Eve in 1906 by Reginald Fessenden, although this is disputed. Charles Herrold started broadcasting in California in 1909 and was carrying audio by the next year, in The Hague, the Netherlands, PCGG started broadcasting on November 6,1919, making it, arguably the first commercial broadcasting station. In 1916, Frank Conrad, an engineer employed at the Westinghouse Electric Corporation, began broadcasting from his Wilkinsburg. Later, the station was moved to the top of the Westinghouse factory building in East Pittsburgh, Westinghouse relaunched the station as KDKA on November 2,1920, as the first commercially licensed radio station in America. The commercial broadcasting designation came from the type of broadcast license, the first licensed broadcast in the United States came from KDKA itself, the results of the Harding/Cox Presidential Election. In 1920, wireless broadcasts for entertainment began in the UK from the Marconi Research Centre 2MT at Writtle near Chelmsford, England. A famous broadcast from Marconis New Street Works factory in Chelmsford was made by the famous soprano Dame Nellie Melba on 15 June 1920 and she was the first artist of international renown to participate in direct radio broadcasts. The 2MT station began to broadcast regular entertainment in 1922, the BBC was amalgamated in 1922 and received a Royal Charter in 1926, making it the first national broadcaster in the world, followed by Czech Radio and other European broadcasters in 1923
8.
Continuous function
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In mathematics, a continuous function is a function for which sufficiently small changes in the input result in arbitrarily small changes in the output. Otherwise, a function is said to be a discontinuous function, a continuous function with a continuous inverse function is called a homeomorphism. Continuity of functions is one of the concepts of topology. The introductory portion of this focuses on the special case where the inputs and outputs of functions are real numbers. In addition, this article discusses the definition for the general case of functions between two metric spaces. In order theory, especially in theory, one considers a notion of continuity known as Scott continuity. Other forms of continuity do exist but they are not discussed in this article, as an example, consider the function h, which describes the height of a growing flower at time t. By contrast, if M denotes the amount of money in an account at time t, then the function jumps at each point in time when money is deposited or withdrawn. A form of the definition of continuity was first given by Bernard Bolzano in 1817. Cauchy defined infinitely small quantities in terms of quantities. The formal definition and the distinction between pointwise continuity and uniform continuity were first given by Bolzano in the 1830s but the work wasnt published until the 1930s, all three of those nonequivalent definitions of pointwise continuity are still in use. Eduard Heine provided the first published definition of continuity in 1872. This is not a definition of continuity since the function f =1 x is continuous on its whole domain of R ∖ A function is continuous at a point if it does not have a hole or jump. A “hole” or “jump” in the graph of a function if the value of the function at a point c differs from its limiting value along points that are nearby. Such a point is called a discontinuity, a function is then continuous if it has no holes or jumps, that is, if it is continuous at every point of its domain. Otherwise, a function is discontinuous, at the points where the value of the function differs from its limiting value, there are several ways to make this definition mathematically rigorous. These definitions are equivalent to one another, so the most convenient definition can be used to determine whether a function is continuous or not. In the definitions below, f, I → R. is a function defined on a subset I of the set R of real numbers and this subset I is referred to as the domain of f
9.
Digital technology
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Digital electronics or digital circuits are electronics that handle digital signals rather than by continuous ranges as used in analog electronics. All levels within a band of values represent the information state. In most cases, the number of states is two, and they are represented by two voltage bands, one near a reference value, and the other a value near the supply voltage. These correspond to the false and true values of the Boolean domain respectively, Digital techniques are useful because it is easier to get an electronic device to switch into one of a number of known states than to accurately reproduce a continuous range of values. Digital electronic circuits are made from large assemblies of logic gates. The binary number system was refined by Gottfried Wilhelm Leibniz and he established that by using the binary system. Digital logic as we know it was the brain-child of George Boole, Boole died young, but his ideas lived on. In an 1886 letter, Charles Sanders Peirce described how logical operations could be carried out by electrical switching circuits, eventually, vacuum tubes replaced relays for logic operations. Lee De Forests modification, in 1907, of the Fleming valve can be used as an AND logic gate, ludwig Wittgenstein introduced a version of the 16-row truth table as proposition 5.101 of Tractatus Logico-Philosophicus. Walther Bothe, inventor of the circuit, got part of the 1954 Nobel Prize in physics. Mechanical analog computers started appearing in the first century and were used in the medieval era for astronomical calculations. In World War II, mechanical computers were used for specialized military applications such as calculating torpedo aiming. During this time the first electronic computers were developed. Originally they were the size of a room, consuming as much power as several hundred modern personal computers. The Z3 was a computer designed by Konrad Zuse, finished in 1941. It was the worlds first working programmable, fully automatic digital computer and its operation was facilitated by the invention of the vacuum tube in 1904 by John Ambrose Fleming. Purely electronic circuit elements soon replaced their mechanical and electromechanical equivalents, the bipolar junction transistor was invented in 1947. From 1955 onwards transistors replaced vacuum tubes in computer designs, giving rise to the generation of computers
10.
Digital signal processing
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Digital signal processing is the use of digital processing, such as by computers, to perform a wide variety of signal processing operations. The signals processed in this manner are a sequence of numbers that represent samples of a variable in a domain such as time, space. Digital signal processing and analog signal processing are subfields of signal processing, digital signal processing can involve linear or nonlinear operations. Nonlinear signal processing is closely related to system identification and can be implemented in the time, frequency. DSP is applicable to both streaming data and static data, the increasing use of computers has resulted in the increased use of, and need for, digital signal processing. To digitally analyze and manipulate an analog signal, it must be digitized with an analog-to-digital converter, sampling is usually carried out in two stages, discretization and quantization. Discretization means that the signal is divided into equal intervals of time, quantization means each amplitude measurement is approximated by a value from a finite set. Rounding real numbers to integers is an example, the Nyquist–Shannon sampling theorem states that a signal can be exactly reconstructed from its samples if the sampling frequency is greater than twice the highest frequency of the signal. In practice, the frequency is often significantly higher than twice that required by the signals limited bandwidth. Theoretical DSP analyses and derivations are typically performed on discrete-time signal models with no amplitude inaccuracies, numerical methods require a quantized signal, such as those produced by an analog-to-digital converter. The processed result might be a frequency spectrum or a set of statistics, but often it is another quantized signal that is converted back to analog form by a digital-to-analog converter. In DSP, engineers usually study digital signals in one of the domains, time domain, spatial domain, frequency domain. They choose the domain in which to process a signal by making an assumption as to which domain best represents the essential characteristics of the signal. The most common processing approach in the time or space domain is enhancement of the signal through a method called filtering. Digital filtering generally consists of linear transformation of a number of surrounding samples around the current sample of the input or output signal. There are various ways to characterize filters, for example, A linear filter is a transformation of input samples. A causal filter uses only samples of the input or output signals. A non-causal filter can usually be changed into a filter by adding a delay to it
11.
Sound pressure
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Sound pressure or acoustic pressure is the local pressure deviation from the ambient atmospheric pressure, caused by a sound wave. In air, sound pressure can be measured using a microphone, the SI unit of sound pressure is the pascal. A sound wave in a medium causes a deviation in the local ambient pressure. Sound pressure, denoted p, is defined by p t o t a l = p s t a t + p, in a sound wave, the complementary variable to sound pressure is the particle velocity. Together they determine the intensity of the wave. Sound intensity, denoted I and measured in W·m−2 in SI units, is defined by I = p v. Consequently, the amplitude of the displacement is related to that of the acoustic velocity. This relationship is an inverse-proportional law, if the sound pressure p1 is measured at a distance r1 from the centre of the sphere, the sound pressure p2 at another position r2 can be calculated, p 2 = r 1 r 2 p 1. The inverse-proportional law for sound pressure comes from the law for sound intensity. Indeed, I = p v = p ∝ p 2, the sound pressure may vary in direction from the centre of the sphere as well, so measurements at different angles may be necessary, depending on the situation. An obvious example of a sound source whose spherical sound wave varies in level in different directions is a bullhorn, Sound pressure level or acoustic pressure level is a logarithmic measure of the effective pressure of a sound relative to a reference value. The commonly used reference sound pressure in air is p 0 =20 μ P a, which is often considered as the threshold of human hearing. The proper notations for sound pressure level using this reference are Lp/ or Lp, most sound level measurements will be made relative to this reference, meaning 1 Pa will equal an SPL of 94 dB. In other media, such as underwater, a level of 1 μPa is used. These references are defined in ANSI S1. 1-1994, the lower limit of audibility is defined as SPL of 0 dB, but the upper limit is not as clearly defined. Ears detect changes in sound pressure, human hearing does not have a flat spectral sensitivity relative to frequency versus amplitude. Humans do not perceive low- and high-frequency sounds as well as they perceive sounds between 3,000 and 4,000 Hz, as shown in the equal-loudness contour. Because the frequency response of human hearing changes with amplitude, three weightings have been established for measuring pressure, A, B and C. A-weighting applies to sound pressures levels up to 55 dB, B-weighting applies to sound levels between 55 dB and 85 dB, and C-weighting is for measuring sound pressure levels above 85 dB
12.
Binary numeral system
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The base-2 system is a positional notation with a radix of 2. Because of its implementation in digital electronic circuitry using logic gates. Each digit is referred to as a bit, the modern binary number system was devised by Gottfried Leibniz in 1679 and appears in his article Explication de lArithmétique Binaire. Systems related to binary numbers have appeared earlier in multiple cultures including ancient Egypt, China, Leibniz was specifically inspired by the Chinese I Ching. The scribes of ancient Egypt used two different systems for their fractions, Egyptian fractions and Horus-Eye fractions, the method used for ancient Egyptian multiplication is also closely related to binary numbers. This method can be seen in use, for instance, in the Rhind Mathematical Papyrus, the I Ching dates from the 9th century BC in China. The binary notation in the I Ching is used to interpret its quaternary divination technique and it is based on taoistic duality of yin and yang. Eight trigrams and a set of 64 hexagrams, analogous to the three-bit and six-bit binary numerals, were in use at least as early as the Zhou Dynasty of ancient China. The Song Dynasty scholar Shao Yong rearranged the hexagrams in a format that resembles modern binary numbers, the Indian scholar Pingala developed a binary system for describing prosody. He used binary numbers in the form of short and long syllables, Pingalas Hindu classic titled Chandaḥśāstra describes the formation of a matrix in order to give a unique value to each meter. The binary representations in Pingalas system increases towards the right, the residents of the island of Mangareva in French Polynesia were using a hybrid binary-decimal system before 1450. Slit drums with binary tones are used to encode messages across Africa, sets of binary combinations similar to the I Ching have also been used in traditional African divination systems such as Ifá as well as in medieval Western geomancy. The base-2 system utilized in geomancy had long been applied in sub-Saharan Africa. Leibnizs system uses 0 and 1, like the modern binary numeral system, Leibniz was first introduced to the I Ching through his contact with the French Jesuit Joachim Bouvet, who visited China in 1685 as a missionary. Leibniz saw the I Ching hexagrams as an affirmation of the universality of his own beliefs as a Christian. Binary numerals were central to Leibnizs theology and he believed that binary numbers were symbolic of the Christian idea of creatio ex nihilo or creation out of nothing. Is not easy to impart to the pagans, is the ex nihilo through Gods almighty power. In 1854, British mathematician George Boole published a paper detailing an algebraic system of logic that would become known as Boolean algebra
13.
Digital electronics
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Digital electronics or digital circuits are electronics that handle digital signals rather than by continuous ranges as used in analog electronics. All levels within a band of values represent the information state. In most cases, the number of states is two, and they are represented by two voltage bands, one near a reference value, and the other a value near the supply voltage. These correspond to the false and true values of the Boolean domain respectively, Digital techniques are useful because it is easier to get an electronic device to switch into one of a number of known states than to accurately reproduce a continuous range of values. Digital electronic circuits are made from large assemblies of logic gates. The binary number system was refined by Gottfried Wilhelm Leibniz and he established that by using the binary system. Digital logic as we know it was the brain-child of George Boole, Boole died young, but his ideas lived on. In an 1886 letter, Charles Sanders Peirce described how logical operations could be carried out by electrical switching circuits, eventually, vacuum tubes replaced relays for logic operations. Lee De Forests modification, in 1907, of the Fleming valve can be used as an AND logic gate, ludwig Wittgenstein introduced a version of the 16-row truth table as proposition 5.101 of Tractatus Logico-Philosophicus. Walther Bothe, inventor of the circuit, got part of the 1954 Nobel Prize in physics. Mechanical analog computers started appearing in the first century and were used in the medieval era for astronomical calculations. In World War II, mechanical computers were used for specialized military applications such as calculating torpedo aiming. During this time the first electronic computers were developed. Originally they were the size of a room, consuming as much power as several hundred modern personal computers. The Z3 was a computer designed by Konrad Zuse, finished in 1941. It was the worlds first working programmable, fully automatic digital computer and its operation was facilitated by the invention of the vacuum tube in 1904 by John Ambrose Fleming. Purely electronic circuit elements soon replaced their mechanical and electromechanical equivalents, the bipolar junction transistor was invented in 1947. From 1955 onwards transistors replaced vacuum tubes in computer designs, giving rise to the generation of computers
14.
Microprocessor
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A microprocessor is a computer processor which incorporates the functions of a computers central processing unit on a single integrated circuit, or at most a few integrated circuits. Microprocessors contain both combinational logic and sequential digital logic, Microprocessors operate on numbers and symbols represented in the binary numeral system. The integration of a whole CPU onto a chip or on a few chips greatly reduced the cost of processing power. Integrated circuit processors are produced in numbers by highly automated processes resulting in a low per unit cost. Single-chip processors increase reliability as there are many electrical connections to fail. As microprocessor designs get better, the cost of manufacturing a chip generally stays the same, before microprocessors, small computers had been built using racks of circuit boards with many medium- and small-scale integrated circuits. Microprocessors combined this into one or a few large-scale ICs, the internal arrangement of a microprocessor varies depending on the age of the design and the intended purposes of the microprocessor. Advancing technology makes more complex and powerful chips feasible to manufacture, a minimal hypothetical microprocessor might only include an arithmetic logic unit and a control logic section. The ALU performs operations such as addition, subtraction, and operations such as AND or OR, each operation of the ALU sets one or more flags in a status register, which indicate the results of the last operation. The control logic retrieves instruction codes from memory and initiates the sequence of operations required for the ALU to carry out the instruction, a single operation code might affect many individual data paths, registers, and other elements of the processor. As integrated circuit technology advanced, it was feasible to manufacture more and more complex processors on a single chip, the size of data objects became larger, allowing more transistors on a chip allowed word sizes to increase from 4- and 8-bit words up to todays 64-bit words. Additional features were added to the architecture, more on-chip registers sped up programs. Floating-point arithmetic, for example, was not available on 8-bit microprocessors. Integration of the point unit first as a separate integrated circuit and then as part of the same microprocessor chip. Occasionally, physical limitations of integrated circuits made such practices as a bit slice approach necessary, instead of processing all of a long word on one integrated circuit, multiple circuits in parallel processed subsets of each data word. With the ability to put large numbers of transistors on one chip and this CPU cache has the advantage of faster access than off-chip memory, and increases the processing speed of the system for many applications. Processor clock frequency has increased more rapidly than external memory speed, except in the recent past, a microprocessor is a general purpose system. Several specialized processing devices have followed from the technology, A digital signal processor is specialized for signal processing, graphics processing units are processors designed primarily for realtime rendering of 3D images
15.
Computer
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A computer is a device that can be instructed to carry out an arbitrary set of arithmetic or logical operations automatically. The ability of computers to follow a sequence of operations, called a program, such computers are used as control systems for a very wide variety of industrial and consumer devices. The Internet is run on computers and it millions of other computers. Since ancient times, simple manual devices like the abacus aided people in doing calculations, early in the Industrial Revolution, some mechanical devices were built to automate long tedious tasks, such as guiding patterns for looms. More sophisticated electrical machines did specialized analog calculations in the early 20th century, the first digital electronic calculating machines were developed during World War II. The speed, power, and versatility of computers has increased continuously and dramatically since then, conventionally, a modern computer consists of at least one processing element, typically a central processing unit, and some form of memory. The processing element carries out arithmetic and logical operations, and a sequencing, peripheral devices include input devices, output devices, and input/output devices that perform both functions. Peripheral devices allow information to be retrieved from an external source and this usage of the term referred to a person who carried out calculations or computations. The word continued with the same meaning until the middle of the 20th century, from the end of the 19th century the word began to take on its more familiar meaning, a machine that carries out computations. The Online Etymology Dictionary gives the first attested use of computer in the 1640s, one who calculates, the Online Etymology Dictionary states that the use of the term to mean calculating machine is from 1897. The Online Etymology Dictionary indicates that the use of the term. 1945 under this name, theoretical from 1937, as Turing machine, devices have been used to aid computation for thousands of years, mostly using one-to-one correspondence with fingers. The earliest counting device was probably a form of tally stick, later record keeping aids throughout the Fertile Crescent included calculi which represented counts of items, probably livestock or grains, sealed in hollow unbaked clay containers. The use of counting rods is one example, the abacus was initially used for arithmetic tasks. The Roman abacus was developed from used in Babylonia as early as 2400 BC. Since then, many forms of reckoning boards or tables have been invented. In a medieval European counting house, a checkered cloth would be placed on a table, the Antikythera mechanism is believed to be the earliest mechanical analog computer, according to Derek J. de Solla Price. It was designed to calculate astronomical positions and it was discovered in 1901 in the Antikythera wreck off the Greek island of Antikythera, between Kythera and Crete, and has been dated to circa 100 BC
16.
Audio storage
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Sound recording and reproduction is an electrical, mechanical, electronic, or digital inscription and re-creation of sound waves, such as spoken voice, singing, instrumental music, or sound effects. The two main classes of sound recording technology are analog recording and digital recording, prior to the development of analog recording, there were mechanical systems for reproducing instrumental music, such as wind-up music boxes and later, in the late 19th century, player pianos. Analog sound reproduction is the process, with a bigger loudspeaker diaphragm causing changes to atmospheric pressure to form acoustic sound waves. Digital recording and reproduction converts the sound signal picked up by the microphone to a digital form by the process of digitization. This lets the audio data be stored and transmitted by a variety of media. Whereas successive copies of an analog recording tend to degrade in quality, as noise is added. A digital audio signal must be reconverted to analog form during playback before it is amplified and connected to a loudspeaker to produce sound, long before sound was first recorded on cylinders or records, music was recorded—first by written music notation, then also by mechanical devices. Fowler, this. cylinder with raised pins on the surface remained the device to produce and reproduce music mechanically until the second half of the nineteenth century. The Banu Musa brothers also invented an automatic flute player, which appears to have been the first programmable machine, according to Fowler, the automata were a robot band that performed. more than fifty facial and body actions during each musical selection. In the 14th century, Flanders introduced a mechanical bell-ringer controlled by a rotating cylinder, similar designs appeared in barrel organs, musical clocks, barrel pianos, and musical boxes. A music box is a musical instrument that produces sounds by the use of a set of pins placed on a revolving cylinder or disc so as to pluck the tuned teeth of a steel comb. They were developed from musical snuff boxes of the 18th century, some of the more complex boxes also have a tiny drum and/or bells, in addition to the metal comb. The fairground organ, developed in 1892, used a system of accordion-folded punched cardboard books, the player piano, first demonstrated in 1876, used a punched paper scroll that could store an long piece of music. The most sophisticated of the rolls were hand-played, meaning that the roll represented the actual performance of an individual. This technology to record a live performance onto a piano roll was not developed until 1904, piano rolls were in continuous mass production from 1896 to 2008. A1908 U. S. Supreme Court copyright case noted that, in 1902 alone, the use of piano rolls began to decline in the 1920s although one type is still being made today. The first device that could record actual sounds as they passed through the air was the phonautograph, the earliest known recordings of the human voice are phonautograph recordings, called phonautograms, made in 1857. They consist of sheets of paper with sound-wave-modulated white lines created by a stylus that cut through a coating of soot as the paper was passed under it
17.
Audio level compression
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Compression is commonly used in sound recording and reproduction, broadcasting, live sound reinforcement and in some instrument amplifiers. A dedicated electronic hardware unit or audio software that applies compression is called a compressor, in the 2000s, compressors became available as software plugins that run in digital audio workstation software. In recorded and live music, compression parameters may be adjusted to change the way they affect sounds, compression and limiting are identical in process but different in degree and perceived effect. A limiter is a compressor with a ratio and, generally. Downward compression reduces loud sounds over a certain threshold while quiet sounds remain unaffected, upward compression increases the loudness of sounds below a certain threshold while leaving louder sounds unaffected. Both downward and upward compression reduce the range of an audio signal. An expander performs the function, increasing the dynamic range of the audio signal. Expanders are generally used to make quiet sounds even quieter by reducing the level of a signal that falls below a set threshold level. A noise gate is a type of expander and this design, known as a feed-forward type, is used today in most compressors. Earlier designs were based on a layout where the signal level was measured after the amplifier. There are a number of used for variable-gain amplification, each having different advantages and disadvantages. Vacuum tubes are used in a configuration called variable-mu where the voltage changes to alter the gain. Optical compressors use a photoresistor and a lamp to create changes in signal gain. This technique is believed by some to sound smoother because the times of the light. Other technologies used include field effect transistors and a diode bridge, when working with digital audio, digital signal processing techniques are commonly used to implement compression via digital audio editors, or dedicated workstations. Often the algorithms used emulate the above analog technologies, a compressor reduces the level of an audio signal if its amplitude exceeds a certain threshold. It is commonly set in decibels dB, where a lower threshold means a portion of the signal is treated. The amount of reduction is determined by ratio, a ratio of 4,1 means that if input level is 4 dB over the threshold
18.
Audio 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
19.
Equalization (audio)
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Equalization is the process of adjusting the balance between frequency components within an electronic signal. The most well known use of equalization is in recording and reproduction but there are many other applications in electronics. The circuit or equipment used to achieve equalization is called an equalizer and these devices strengthen or weaken the energy of specific frequency bands or frequency ranges. In sound recording and reproduction, equalization is the commonly used to alter the frequency response of an audio system using linear filters. Most hi-fi equipment uses relatively simple filters to make bass and treble adjustments, graphic and parametric equalizers have much more flexibility in tailoring the frequency content of an audio signal. An equalizer is the circuit or equipment used to achieve equalization, since equalizers, adjust the amplitude of audio signals at particular frequencies, they are, in other words, frequency-specific volume knobs. The term EQ specifically refers to this variant of the term, stereos and basic guitar amplifiers typically have adjustable equalizers which boost or cut bass or treble frequencies. Mid- to high-priced guitar and bass amplifiers usually have more bands of frequency control, such as bass, mid-range and treble or bass, low-mid, high-mid, some amps have an additional knob for controlling very high frequencies. Broadcast and recording studios use sophisticated equalizers capable of more detailed adjustments. The most common equalizers in music production are parametric, semi-parametric, graphic, peak, graphic equalizers are often included in consumer audio equipment and software which plays music on home computers. Parametric equalizers require more expertise than graphic equalizers, and they can provide more specific compensation or alteration around a chosen frequency and this may be used in order to remove unwanted resonances or boost certain frequencies. An acoustic guitarist who finds that her instrument sounds too boomy may ask the engineer to reduce the low frequency range response. A guitarist who finds that the instrument sound in the PA has too much finger noise may ask the engineer to reduce the frequency range response. The concept of equalization was first applied in correcting the frequency response of telephone lines using passive networks, a plot of the systems net frequency response would be flat, as its response to all frequencies would literally be equal. Much later the concept was applied in engineering to adjust the frequency response in recording, reproduction. Audio amplifiers have long had filters or controls to modify their frequency response and these are most often in the form of variable bass and treble controls, and switches to apply low-cut or high-cut filters for elimination of low frequency rumble and high frequency hiss respectively. Graphic equalizers and other equipment developed for improving fidelity have since used by recording engineers to modify frequency responses for aesthetic reasons. Hence in the field of audio electronics the term equalization is now used to describe the application of such filters regardless of intent
20.
Audio filter
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An audio filter is a frequency dependent amplifier circuit, working in the audio frequency range,0 Hz to beyond 20 kHz. Audio filters can amplify, pass or attenuate some frequency ranges, common types include low-pass filters, which pass through frequencies below their cutoff frequencies, and progressively attenuates frequencies above the cutoff frequency. Low-pass filters are used in audio crossovers to remove content from signals being sent to a low-frequency subwoofer system. A high-pass filter does the opposite, passing high frequencies above the cutoff frequency, a high-pass filter can be used in an audio crossover to remove low-frequency content from a signal being sent to a tweeter. A bandpass filter passes frequencies between its two frequencies, while attenuating those outside the range. A band-reject filter, attenuates frequencies between its two frequencies, while passing those outside the reject range. An all-pass filter passes all frequencies, but affects the phase of any given sinusoidal component, in more complex cases, an audio filter can provide a feedback loop, which introduces resonance alongside attenuation. Audio filters can also be designed to gain as well as attenuation. In other applications, such as with synthesizers or sound effects, audio filters can be implemented in analog circuitry as analog filters or in DSP code or computer software as digital filters. Generically, the term audio filter can be applied to anything which changes the timbre. Electrical resonance Electronic filter Equalization Feedback Linear filter Oscillation Self-resonant frequency
21.
Noise cancellation
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Active noise control, also known as noise cancellation, or active noise reduction, is a method for reducing unwanted sound by the addition of a second sound specifically designed to cancel the first. Sound is a wave, which consists of alternating periods of compression and rarefaction. A noise-cancellation speaker emits a sound wave with the same amplitude, the waves combine to form a new wave, in a process called interference, and effectively cancel each other out – an effect which is called destructive interference. Modern active noise control is achieved through the use of analog circuits or digital signal processing. This inverted signal is amplified and a transducer creates a sound wave directly proportional to the amplitude of the original waveform. This effectively reduces the volume of the perceivable noise, a noise-cancellation speaker may be co-located with the sound source to be attenuated. In this case it must have the same power level as the source of the unwanted sound. Alternatively, the transducer emitting the cancellation signal may be located at the location where sound attenuation is wanted and this requires a much lower power level for cancellation but is effective only for a single user. In small enclosed spaces global noise reduction can be achieved via multiple speakers and feedback microphones, applications can be 1-dimensional or 3-dimensional, depending on the type of zone to protect. Periodic sounds, even ones, are easier to cancel than random sounds due to the repetition in the wave form. Protection of a 1-dimension zone is easier and requires one or two microphones and speakers to be effective. Several commercial applications have been successful, noise-cancelling headphones, active mufflers, the term 1-dimension refers to a simple pistonic relationship between the noise and the active speaker or between the active speaker and the listener. Protection of a 3-dimension zone requires many microphones and speakers, making it more expensive, high frequency waves are difficult to reduce in three dimensions due to their relatively short audio wavelength in air. High frequency sounds above 1000 Hz tend to cancel and reinforce unpredictably from many directions, in sum, the most effective noise reduction in three-dimensional space involves low frequency sounds. This is because an engines cyclic nature makes fast Fourier transform analysis, noise control is an active or passive means of reducing sound emissions, often for personal comfort, environmental considerations or legal compliance. Active noise control is sound reduction using a power source, passive noise control is sound reduction by noise-isolating materials such as insulation, sound-absorbing tiles, or a muffler rather than a power source. Active noise canceling is best suited for low frequencies, for higher frequencies, the spacing requirements for free space and zone of silence techniques become prohibitive. In acoustic cavity and duct based systems, the number of nodes grows rapidly with increasing frequency, passive treatments become more effective at higher frequencies and often provide an adequate solution without the need for active control
22.
Echo (phenomenon)
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In audio signal processing and acoustics, echo is a reflection of sound that arrives at the listener with a delay after the direct sound. The delay is proportional to the distance of the surface from the source. Typical examples are the echo produced by the bottom of a well, by a building, or by the walls of an enclosed room, a true echo is a single reflection of the sound source. The word echo derives from the Greek ἠχώ, itself from ἦχος, echo in the folk story of Greek is a mountain nymph whose ability to speak was cursed, only able to repeat the last words anyone spoke to her. Some animals use echo for location sensing and navigation, such as cetaceans, acoustic waves are reflected by walls or other hard surfaces, such as mountains and privacy fences. The reason of reflection may be explained as a discontinuity in the propagation medium and this can be heard when the reflection returns with sufficient magnitude and delay to be perceived distinctly. When sound, or the echo itself, is reflected multiple times from multiple surfaces, the human ear cannot distinguish echo from the original direct sound if the delay is less than 1/15 of a second. The velocity of sound in dry air is approximately 343 m/s at a temperature of 25 °C, therefore, the reflecting object must be more than 17. 2m from the sound source for echo to be perceived by a person located at the source. When a sound produces an echo in two seconds, the object is 343m away. In nature, canyon walls or rock cliffs facing water are the most common settings for hearing echoes. The strength of echo is frequently measured in dB sound pressure level relative to the transmitted wave. Echoes may be desirable or undesirable, in music performance and recording, electric echo effects have been used since the 1950s. The Echoplex is a delay effect, first made in 1959 that recreates the sound of an acoustic echo. Designed by Mike Battle, the Echoplex set a standard for the effect in the 1960s and was used by most of the guitar players of the era. While Echoplexes were used heavily by guitar players, many recording studios used the Echoplex. Beginning in the 1970s, Market built the solid-state Echoplex for Maestro, in the 2000s, most echo effects units use electronic or digital circuitry to recreate the echo effect. Hamilton Mausoleum, Hamilton, South Lanarkshire, Scotland, Its high stone holds the record for the longest echo in the world, gol Gumbaz of Bijapur, India, Any whisper, clap or sound gets echoed repeatedly. The gazebo of Napier Museum in Trivandrum, Kerala, India, listen to Duck echoes and an animated demonstration of how an echo is formed
23.
Reverb
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Reverberation, in psychoacoustics and acoustics, is the persistence of sound after a sound is produced. This is most noticeable when the sound source stops but the reflections continue, decreasing in amplitude, until they reach zero amplitude. In comparison to an echo that is a minimum of 50 to 100 ms after the initial sound. As time passes, the amplitude of the reflections is reduced until it is reduced to zero, Reverberation is not limited to indoor spaces as it exists in forests and other outdoor environments where reflection exists. Reverberation occurs naturally when a person sings, talks or plays an instrument acoustically in a hall or performance space with sound-reflective surfaces. The sound of reverberation is often added to the vocals of singers in live sound systems. The time it takes for a signal to drop by 60 dB is the reverberation time, RT60 is the time required for reflections of a direct sound to decay 60 dB. Reverberation time is stated as a single value, if measured as a wide band signal, however, being frequency dependent. Being frequency dependent, the time measured in narrow bands will differ depending on the frequency band being measured. For precision, it is important to know what ranges of frequencies are being described by a time measurement. In the late 19th century, Wallace Clement Sabine started experiments at Harvard University to investigate the impact of absorption on the reverberation time. Using a portable wind chest and organ pipes as a sound source and he found that the reverberation time is proportional to room dimensions and inversely proportional to the amount of absorption present. The optimum reverberation time for a space in which music is played depends on the type of music that is to be played in the space, rooms used for speech typically need a shorter reverberation time so that speech can be understood more clearly. If the reflected sound from one syllable is still heard when the syllable is spoken. Cat, Cab, and Cap may all sound very similar, if on the other hand the reverberation time is too short, tonal balance and loudness may suffer. Reverberation effects are used in studios to add depth to sounds. Reverberation changes the perceived spectral structure of a sound, but does not alter the pitch, basic factors that affect a rooms reverberation time include the size and shape of the enclosure as well as the materials used in the construction of the room. Every object placed within the enclosure can also affect this time, including people
24.
Mixing console
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In the 2010s, a mixer is able to control analog or digital signals, depending on the type of mixer. The modified signals are summed to produce the output signals. A typical, simple application combines signals from two microphones into an amplifier that drives one set of speakers simultaneously, a DJ mixer may have only two channels, for mixing two record players. A coffeehouses tiny stage might only have a six channel mixer, a nightclub stages mixer for rock music shows may have 24 channels for mixing the signals from a rhythm section, lead guitar and several vocalists. A mixing console for a large concert may have 48 channels, a mixing console in a professional recording studio may have as many as 96 channels. In practice, mixers do more than simply mix signals, some mixers have onboard electronic effects, such as reverb. Some mixers intended for small venue live performance applications may include a power amplifier. A mixing console is also known as a mixer, audio console, mixing desk, sound mixer, sound board. A typical analog mixing board has three sections, Channel inputs Master controls Audio level metering The channel input strips are usually a bank of identical monaural or stereo input channels. Each channel has rotary knobs, buttons and/or faders for controlling the gain, the master control section is used to adjust the levels of the overall output of the mixer. The master control section has sub-group faders, master faders, master auxiliary mixing bus level controls, in addition it may have solo monitoring controls, a stage talk-back microphone control, muting controls and an output matrix mixer. On smaller mixers the inputs are on the left of the mixing board, in larger mixers, the master controls are in the center with input faders and channel strips on both sides. The audio level meters may be above the input and master sections or they may be integrated into the input, the audio level meters indicate when the signals are clipping. Each signal that is plugged into the mixer has its own channel, depending on the specific mixer, each channel is stereo or monaural. On most mixers, each channel has an XLR input, the smallest, least expensive mixers may only have one XLR input with the other inputs being line inputs. These can be used by a singer-guitarist or other small acts, below each input, there are usually several rotary controls. The first knob is typically a trim or gain control, the inputs buffer the signal from the external device and this controls the amount of amplification or attenuation needed to bring the signal to a nominal level for processing. This stage is where most noise of interference is picked up, balanced inputs and connectors, such as XLR or phone connectors, reduce interference problems
25.
Medium wave
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Medium wave is the part of the medium frequency radio band used mainly for AM radio broadcasting. Practical groundwave reception typically extends to 200–300 miles, with distances over terrain with higher ground conductivity. Most broadcast stations use groundwave to cover their listening area, Medium waves can also reflect off charged particle layers in the ionosphere and return to Earth at much greater distances, this is called the skywave. At night, especially in winter months and at times of low solar activity, when this happens, MF radio waves can easily be received many hundreds or even thousands of miles away as the signal will be reflected by the higher F layer. This can allow very long-distance broadcasting, but can also interfere with distant local stations, due to the limited number of available channels in the MW broadcast band, the same frequencies are re-allocated to different broadcasting stations several hundred miles apart. On nights of good skywave propagation, the signals of distant station may interfere with the signals of local stations on the same frequency. These channels are called clear channels, and they are required to broadcast at higher powers of 10 to 50 kW and this arrangement had numerous practical difficulties. The Commerce Department rarely intervened in such cases but left it up to stations to enter into voluntary timesharing agreements amongst themselves, the addition of a third entertainment wavelength,400 meters, did little to solve this overcrowding. In 1923, the Commerce Department realized that as more and more stations were applying for commercial licenses, on 15 May 1923, Commerce Secretary Herbert Hoover announced a new bandplan which set aside 81 frequencies, in 10 kHz steps, from 550 kHz to 1350 kHz. Each station would be assigned one frequency, no longer having to broadcast weather, class A and B stations were segregated into sub-bands. Today in most of the Americas, mediumwave broadcast stations are separated by 10 kHz and have two sidebands of up to ±5 kHz in theory, in the rest of the world, the separation is 9 kHz, with sidebands of ±4.5 kHz. Both provide adequate quality for voice, but are insufficient for high-fidelity broadcasting. In the US and Canada the maximum power is restricted to 50 kilowatts. Those stations which shut down completely at night are known as daytimers. In most cases there are two limits, a lower one for omnidirectional and a higher one for directional radiation with minima in certain directions. The power limit can also be depending on daytime and it is possible, other countries may only operate low-powered transmitters on the same frequency, again subject to agreement. For example, Russia operates a transmitter, located in its Kaliningrad exclave and used for external broadcasting. Due to the demand for frequencies in Europe, many countries operate single frequency networks, in Britain
26.
Shortwave
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Shortwave radio is radio transmission using shortwave frequencies, generally 1. 6–30 MHz, just above the medium wave AM broadcast band. Radio waves in this band can be reflected or refracted from a layer of charged atoms in the atmosphere called the ionosphere. Therefore, short waves directed at an angle into the sky can be reflected back to Earth at great distances and this is called skywave or skip propagation. Shortwave radio is used for broadcasting of voice and music to listeners over very large areas. It is also used for military radar, diplomatic communication. The widest popular definition of the frequency interval is the ITU Region 1 definition, and is the span 1. 6–30 MHz, just above the medium wave band. In reality, the definition of the frequency band is a mess. The broadcast medium wave band now extends above the 200 m/1500 kHz limit, early radio telegraphy had used long wave transmissions. The drawbacks to this included a very limited spectrum available for long distance communication. It was also difficult to beam the radio wave directionally with long wave, prior to the 1920s, the shortwave frequencies above 1.5 MHz were regarded as useless for long distance communication and were designated in many countries for amateur use. Franklin rigged up an antenna at Poldhu Wireless Station, Cornwall. In June and July 1923, wireless transmissions were completed during nights on 97 meters from Poldhu to Marconis yacht Elettra in the Cape Verde Islands, in September 1924, Marconi transmitted daytime and nighttime on 32 meters from Poldhu to his yacht in Beirut. Franklin went on to refine the directional transmission, by inventing the curtain array aerial system, the UK-to-Canada shortwave Beam Wireless Service went into commercial operation on 25 October 1926. Beam Wireless Services from the UK to Australia, South Africa, Shortwave communications began to grow rapidly in the 1920s, similar to the internet in the late 20th century. Shortwave also ended the need for investments in massive longwave wireless stations. The cable companies began to lose large sums of money in 1927, the British government convened the Imperial Wireless and Cable Conference in 1928 to examine the situation that had arisen as a result of the competition of Beam Wireless with the Cable Services. The name of the company was changed to Cable and Wireless Ltd. in 1934, long-distance cables had a resurgence beginning in 1956 with the laying of TAT-1 across the Atlantic Ocean, the first voice frequency cable on this route. This provided 36 high quality telephone channels and was followed by even higher capacity cables all around the world
27.
Loudness
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Loudness is the characteristic of a sound that is primarily a psycho-physiological correlate of physical strength. More formally, it is defined as, That attribute of auditory sensation in terms of sounds can be ordered on a scale extending from quiet to loud. The relation of physical attributes of sound to perceived loudness consists of physical, physiological and psychological components, in different industries, loudness may have different meanings, and different standards exist, each purporting to define the measurement. Some definitions such as LKFS refer to relative loudness of different segments of electronically reproduced sounds such as for broadcasting and cinema. Others, such as ISO 532A, ISO 532B, DIN45631 and it is sometimes stated that loudness is a subjective measure, often confused with physical measures of sound strength such as sound pressure, sound pressure level, sound intensity or sound power. It is often possible to separate the truly subjective components such as social considerations from the physical and physiological. Filters such as A-weighting attempt to adjust sound measurements to correspond to loudness as perceived by the typical human, A-weighting follows human sensitivity to sound and describes relative perceived loudness for at quiet to moderate speech levels, around 40 phons. However, physiological loudness perception is a more complex process than can be captured with a single correction curve. Not only do equal-loudness contours vary with intensity, but perceived loudness of a complex sound depends on whether its spectral components are closely or widely spaced in frequency. When generating neural impulses in response to sounds of one frequency, the ear is sensitive to nearby frequencies. Sounds containing spectral components in critical bands are perceived as louder even if the total sound pressure remains constant. The perception of loudness is related to sound level, frequency content. The human auditory system averages the effects of SPL over a 600–1000 ms interval, for sounds of duration greater than 1 second, the moment-by-moment perception of loudness will be related to the average loudness during the preceding 600–1000 ms. For sounds having a longer than 1 second, the relationship between SPL and loudness of a single tone can be approximated by Stevens power law in which SPL has an exponent of 0.6. More precise measurements indicate that loudness increases with a higher exponent at low and high levels, the sensitivity of the human ear changes as a function of frequency, as shown in the equal-loudness graph. Each line on this shows the SPL required for frequencies to be perceived as equally loud. It also shows that humans with normal hearing are most sensitive to sounds around 2–4 kHz, a complete model of the perception of loudness will include the integration of SPL by frequency. Historically, loudness was measured using an ear-balance audiometer in which the amplitude of a wave was adjusted by the user to equal the perceived loudness of the sound being evaluated
28.
Delay (audio effect)
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Delay is an audio effect and an effects unit which records an input signal to an audio storage medium, and then plays it back after a period of time. The delayed signal may either be played multiple times, or played back into the recording again. Delay effects range from an echo effect to a pronounced blending of previous sounds with new sounds. The first delay effects were achieved using tape loops improvised on reel-to-reel magnetic recording systems, by shortening or lengthening the loop of tape and adjusting the read and write heads, the nature of the delayed echo could be controlled. Audio engineers working in music quickly adapted similar techniques, to augment their use of plate reverb. In the late 1940s and early 1950s, several sound engineers began making devices for use in recording studios, guitarist and instrument designer Les Paul was an early pioneer in delay devices. A landmark device was the EchoSonic made by American Ray Butts and it is a portable guitar amplifier with a built-in tape echo, which became used widely in country music and especially in rock and roll. Tape echoes became commercially available in the 1950s, an echo machine is the early name for a sound processing device used with electronic instruments to repeat the sound and produce a simulated echo. The device was popular with guitarists and was used by Brian May, Jimmy Page, One example is the Echoplex which used a tape loop. The length of delay was adjusted by changing the distance between the record and playback heads. Another example is the Roland Space Echo with a record and multiple tape heads. The time between echo repeats was adjusted by varying the tape speed, the length or intensity of the echo effect was adjusted by changing the amount of echo signal was fed back into the pre-echo signal. Different effects could be created by combining the different playback heads, some models also included a spring reverb. Before the invention of audio technology, music employing a delayed echo had to be recorded in a naturally reverberant space, often an inconvenience for musicians. The presence of multiple taps made it possible to have delays at varying rhythmic intervals, many delay processors based on analog tape recording, such as Ray Butts EchoSonic, Mike Battles Echoplex, or the Roland Space Echo, used magnetic tape as their recording and playback medium. Electric motors guided a tape loop through a device with a variety of mechanisms allowing modification of the effects parameters, in the Space Echo, all of the heads are fixed, but the speed of the tape could be adjusted, changing the delay time. Thin magnetic tape was not entirely suited for operation, however. Several designs were made, the 1959 Ecco-Fonic had a spinning head, many were temperamental, such as the Vox Echomatic
29.
Spring reverb
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Reverberation, in psychoacoustics and acoustics, is the persistence of sound after a sound is produced. This is most noticeable when the sound source stops but the reflections continue, decreasing in amplitude, until they reach zero amplitude. In comparison to an echo that is a minimum of 50 to 100 ms after the initial sound. As time passes, the amplitude of the reflections is reduced until it is reduced to zero, Reverberation is not limited to indoor spaces as it exists in forests and other outdoor environments where reflection exists. Reverberation occurs naturally when a person sings, talks or plays an instrument acoustically in a hall or performance space with sound-reflective surfaces. The sound of reverberation is often added to the vocals of singers in live sound systems. The time it takes for a signal to drop by 60 dB is the reverberation time, RT60 is the time required for reflections of a direct sound to decay 60 dB. Reverberation time is stated as a single value, if measured as a wide band signal, however, being frequency dependent. Being frequency dependent, the time measured in narrow bands will differ depending on the frequency band being measured. For precision, it is important to know what ranges of frequencies are being described by a time measurement. In the late 19th century, Wallace Clement Sabine started experiments at Harvard University to investigate the impact of absorption on the reverberation time. Using a portable wind chest and organ pipes as a sound source and he found that the reverberation time is proportional to room dimensions and inversely proportional to the amount of absorption present. The optimum reverberation time for a space in which music is played depends on the type of music that is to be played in the space, rooms used for speech typically need a shorter reverberation time so that speech can be understood more clearly. If the reflected sound from one syllable is still heard when the syllable is spoken. Cat, Cab, and Cap may all sound very similar, if on the other hand the reverberation time is too short, tonal balance and loudness may suffer. Reverberation effects are used in studios to add depth to sounds. Reverberation changes the perceived spectral structure of a sound, but does not alter the pitch, basic factors that affect a rooms reverberation time include the size and shape of the enclosure as well as the materials used in the construction of the room. Every object placed within the enclosure can also affect this time, including people
30.
Reverberation
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Reverberation, in psychoacoustics and acoustics, is the persistence of sound after a sound is produced. This is most noticeable when the sound source stops but the reflections continue, decreasing in amplitude, until they reach zero amplitude. In comparison to an echo that is a minimum of 50 to 100 ms after the initial sound. As time passes, the amplitude of the reflections is reduced until it is reduced to zero, Reverberation is not limited to indoor spaces as it exists in forests and other outdoor environments where reflection exists. Reverberation occurs naturally when a person sings, talks or plays an instrument acoustically in a hall or performance space with sound-reflective surfaces. The sound of reverberation is often added to the vocals of singers in live sound systems. The time it takes for a signal to drop by 60 dB is the reverberation time, RT60 is the time required for reflections of a direct sound to decay 60 dB. Reverberation time is stated as a single value, if measured as a wide band signal, however, being frequency dependent. Being frequency dependent, the time measured in narrow bands will differ depending on the frequency band being measured. For precision, it is important to know what ranges of frequencies are being described by a time measurement. In the late 19th century, Wallace Clement Sabine started experiments at Harvard University to investigate the impact of absorption on the reverberation time. Using a portable wind chest and organ pipes as a sound source and he found that the reverberation time is proportional to room dimensions and inversely proportional to the amount of absorption present. The optimum reverberation time for a space in which music is played depends on the type of music that is to be played in the space, rooms used for speech typically need a shorter reverberation time so that speech can be understood more clearly. If the reflected sound from one syllable is still heard when the syllable is spoken. Cat, Cab, and Cap may all sound very similar, if on the other hand the reverberation time is too short, tonal balance and loudness may suffer. Reverberation effects are used in studios to add depth to sounds. Reverberation changes the perceived spectral structure of a sound, but does not alter the pitch, basic factors that affect a rooms reverberation time include the size and shape of the enclosure as well as the materials used in the construction of the room. Every object placed within the enclosure can also affect this time, including people
31.
Flanging
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Flanging /ˈflændʒɪŋ/ is an audio effect produced by mixing two identical signals together, one signal delayed by a small and gradually changing period, usually smaller than 20 milliseconds. This produces a swept comb filter effect, peaks and notches are produced in the frequency spectrum. Varying the time delay causes these to sweep up and down the frequency spectrum, a flanger is an effects unit that creates this effect. Part of the signal is usually fed back to the input, producing a resonance effect which further enhances the intensity of the peaks. The phase of the signal is sometimes inverted, producing another variation on the flanger sound. As an audio effect, a listener hears a drainpipe or swoosh or jet plane sweeping effect as shifting sum-and-difference harmonics are created analogous to use of a notch filter. The term flanging comes from one of the methods of producing the effect. The finished music track is recorded simultaneously to two matching tape machines, then replayed with both decks in sync, the playback-head output from the two recorders is mixed to a third recorder. The engineer slows down one recorder by lightly pressing a finger on the flange of one of the playout reels, the drainpipe or subtle swoosh flange flango effect sweeps in one direction, and the playback of that recorder remains slightly behind the other when the finger is removed. By pressing a finger on the flange of the other deck, as the weight of the tape built on one reel, the pressure on the capstans caused flanging in mixdown or dubbing. His 1955 single Nuevo Laredo features phase shifting, as does the 1953 single Mammys Boogie, american music industry veterans David S. Flanging is also heard in the opening of The Ventures 1962 cover of The Tornados Telstar, in a rocket launch sound effect. Further development of the effect is attributed to Ken Townsend, an engineer at EMIs Abbey Road Studio. Tired of laboriously re-recording dual vocal tracks, John Lennon asked Townsend if there was some way for the Beatles to get the sound of double-tracked vocals without doing the work, Townsend devised Artificial Double Tracking or ADT. According to historian Mark Lewisohn, it was Lennon who first called the technique flanging, Lennon asked George Martin to explain how ADT worked, and Martin answered with the nonsense explanation Now listen, its very simple. We take the image and we split it through a double vibrocated sploshing flange with double negative feedback. Martin replied, Well, lets flange it again and see, from that point, when Lennon wanted ADT he would ask for his voice to be flanged, or call out for Kens flanger. According to Lewisohn, the Beatles influence meant the term flanging is still in use today, the first Beatles track to feature flanging was Tomorrow Never Knows from Revolver, which was recorded on April 6,1966. When Revolver was released on August 5,1966, almost every song had been subjected to flanging, others have attributed it to George Chkiantz, an engineer at Olympic Studios in Barnes, London
32.
Tachometer
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A tachometer is an instrument measuring the rotation speed of a shaft or disk, as in a motor or other machine. The device usually displays the revolutions per minute on an analogue dial. The word comes from Greek ταχος and metron, essentially the words tachometer and speedometer have identical meaning, a device that measures speed. It is by convention that in the automotive world one is used for engine. In formal engineering nomenclature, more terms are used to distinguish the two. The first mechanical tachometers were based on measuring the centrifugal force, the inventor is assumed to be the German engineer Dietrich Uhlhorn, he used it for measuring the speed of machines in 1817. Since 1840, it has used to measure the speed of locomotives. Tachometers or revolution counters on cars, aircraft, and other vehicles show the rate of rotation of the engines crankshaft and this can assist the driver in selecting appropriate throttle and gear settings for the driving conditions. This is more applicable to manual transmissions than to automatics, the red zone is superfluous on most modern cars, since their engines typically have a revolution limiter which electronically limits engine speed to prevent damage. Tractors fitted with a take off system have tachometers showing the engine speed needed to rotate the PTO at the standardized speed required by most PTO-driven implements. In many countries, tractors are required to have a speedometer for use on a road, to save fitting a second dial, the vehicles tachometer is often marked with a second scale in units of speed. This scale is only accurate in a gear, but since many tractors only have one gear that is practical for use on-road. Tractors with multiple road gears often have tachometers with more than one speed scale, aircraft tachometers have a green arc showing the engines designed cruising speed range. This is from a connection called an AC tap which is a connection to one of the stators coil output. Tachometers driven by a cable from a drive unit fitted to the engine exist - usually on simple diesel-engined machinery with basic or no electrical systems. On recent EMS found on vehicles, the signal for the tachometer is usually generated from an ECU which derives the information from either the crankshaft or camshaft speed sensor. Tachometers are used to estimate traffic speed and volume, a vehicle is equipped with the sensor and conducts tach runs which record the traffic data. These data are a substitute or complement to loop detector data, to get statistically significant results requires a high number of runs, and bias is introduced by the time of day, day of week, and the season
33.
Phaser (effect)
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A phaser is an electronic sound processor used to filter a signal by creating a series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs of the waveform being affected is typically modulated so that they vary over time, for this purpose, phasers usually include a low-frequency oscillator. The electronic phasing effect is created by splitting an audio signal into two paths, one path treats the signal with an all-pass filter, which preserves the amplitude of the original signal and alters the phase. The amount of change in phase depends on the frequency, when signals from the two paths are mixed, the frequencies that are out of phase will cancel each other out, creating the phasers characteristic notches. Changing the mix ratio changes the depth of the notches, the deepest notches occur when the mix ratio is 50%, the definition of phaser typically excludes such devices where the all-pass section is a delay line, such a device is called a flanger. Using a delay line creates a series of equally spaced notches. It is possible to cascade a delay line with another type of all-pass filter and this combines the unlimited number of notches from the flanger with the uneven spacing of the phaser. Traditional electronic phasers use a series of variable all-pass phase-shift networks which alter the phases of the different frequency components in the signal and these networks pass all frequencies at equal volume, introducing only phase change to the signal. Human ears are not very responsive to phase differences, but this creates audible interferences when mixed back with the dry signal, creating notches. The simplified structure of a phaser is shown below, The number of all-pass filters varies with different models. Digital phasers may offer up to 32 or even more and this determines the number of notches/peaks in the sound, affecting the general sound character. A phaser with n stages generally has n/2 notches in the spectrum, additionally, the output can be fed back to the input for a more intense effect, creating a resonant effect by emphasizing frequencies between notches. This involves feeding the output of the all-pass filter chain back to the input, as shown here, note that the peaks between the notches are sharper when theres feedback, giving a distinct sound. A stereo phaser is usually two identical phasers modulated by a signal, the output of the oscillators for the left. Most modern phasers are a part of a signal processor. Phasers are mostly found as plugins for sound editing software, as a part of a monolithic rackmount sound effect unit, phasing is a popular effect for electric guitar. The term was used to refer to the original tape flanging effect heard on many psychedelic records of the late 1960s. By the early 1970s, phasing was available as a portable guitar effect, in the late 1970s, Brian May used large amounts of phasing, in such songs as Sheer Heart Attack
34.
All-pass filter
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An all-pass filter is a signal processing filter that passes all frequencies equally in gain, but changes the phase relationship among various frequencies. It does this by varying its phase shift as a function of frequency, generally, the filter is described by the frequency at which the phase shift crosses 90°. They are generally used to compensate for other undesired phase shifts that arise in the system, or for mixing with an unshifted version of the original to implement a notch comb filter. The operational amplifier circuit shown in Figure 1 implements an active all-pass filter with the transfer function H = s R C −1 s R C +1, which has one pole at -1/RC and one zero at 1/RC. The magnitude and phase of H for some angular frequency ω are | H | =1 and ∠ H =180 ∘ −2 arctan , as expected, the filter has unity-gain magnitude for all ω. The filter introduces a different delay at each frequency and reaches input-to-output quadrature at ω=1/RC and this implementation uses a high-pass filter at the non-inverting input to generate the phase shift and negative feedback to compensate for the filters attenuation. At high frequencies, the capacitor is a circuit, thereby creating a unity-gain voltage buffer. At low frequencies and DC, the capacitor is an open circuit, at the corner frequency ω=1/RC of the high-pass filter, the circuit introduces a +90 degree shift. In fact, the shift of the all-pass filter is double the phase shift of the high-pass filter at its non-inverting input. The Laplace transform of a delay is given by exp . By setting R C = T /2 we recover H from above, a similar all-pass filter can be implemented by interchanging the position of the resistor and capacitor, which turns the high-pass filter into a low-pass filter. The result is a phase shifter with the same frequency but a −180 degree shift at high frequencies. In other words, the function is negated, and so it has the same pole at −1/RC. Again, the shift of the all-pass filter is double the phase shift of the first-order filter at its non-inverting input. The resistor can be replaced with a FET in its mode to implement a voltage-controlled phase shifter. In electronic music, a phaser typically consists of two, four or six of these phase-shifting sections connected in tandem and summed with the original, a low-frequency oscillator ramps the control voltage to produce the characteristic swooshing sound. These circuits are used as phase shifters and in systems of phase shaping, filters such as the above can be cascaded with unstable or mixed-phase filters to create a stable or minimum-phase filter without changing the magnitude response of the system. For example, by choice of pole, a pole of an unstable system that is in the right-hand plane can be canceled and reflected on the left-hand plane
35.
C-3PO
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C-3PO or See-Threepio is a humanoid robot character from the Star Wars franchise who appears in the original Star Wars films, the prequel trilogy and the sequel trilogy. Created by Anakin Skywalker, C-3PO was designed as a protocol droid intended to assist in etiquette, customs, along with his astromech droid counterpart and friend R2-D2, C-3PO provides comic relief within the narrative structure of the films, and serving as foils. C-3PO also appears frequently in the Star Wars Canon and Star Wars Legends continuities of novels, comic books, and video games, and was a major character in the ABC television show Droids. In Star Wars, C-3PO is introduced to the audience when he, when R2-D2 attempts to leave the ship to deliver a secret message to Obi-Wan Kenobi, C-3PO follows R2-D2 into an escape pod, which lands on the planet Tatooine. There, C-3PO and R2-D2 are captured by Jawas, and are taken to be sold, in the process of being sold to Owen Lars, C-3PO convinces his new owner to buy R2-D2 as well. The duo ultimately lead Lars nephew, Luke Skywalker, to Obi-Wan, when the ship is taken hostage on the Death Star, C-3PO helps R2-D2 shut down the space stations trash computer, saving their human companions lives, before escaping on board the Falcon. When R2-D2 is damaged during the Battle of Yavin, a grief-stricken C-3PO offers to donate his spare parts to repair his counterpart. C-3PO and a repaired R2-D2 are present at the celebration of the Death Stars destruction at the end of the film, throughout the film C-3PO is a foil to R2-D2s antics, even when C-3PO translates R2-D2s machine speech for the audience. When R2 is damaged in the Battle of Yavin, C-3PO offers to donate any mechanical parts helpful in his repair, C-3PO also displays the ability to deceive humans. In The Empire Strikes Back, C-3PO is responsible for identifying the Empires probe droid, C-3PO escapes with Han, Chewbacca, and Leia in the Millennium Falcon, while R2-D2 joins Luke in his search for Yoda. During this time C-3PO and Solo are often shown as foils, C-3PO quoting odds, after a chase through the Hoth asteroid field, the Falcon escapes to Cloud City on Bespin. While exploring a room in Cloud City, C-3PO is blasted by an off-camera stormtrooper, in search of R2-D2, Chewbacca heads to the Ugnaught recycling facility where he finds the dismembered parts of the droid. When Darth Vader reveals his presence to the group that day, Chewbacca is sent into a holding cell, but is permitted to rebuild the droid. Thereafter, Chewbacca carries the partially rebuilt C-3PO on his back during Hans encasement in carbonite, with the help of the citys administrator, Lando Calrissian, Princess Leia, Chewbacca, and C-3PO escape the city. Having C-3PO on Chewbaccas back proves to be beneficial, in that when Boba Fett escapes the city with Han Solo, while escaping Vaders flagship Executor, R2-D2 begins repairing C-3PO. After making their way to the Rebels rendezvous point, C-3PO is fully repaired, in Return of the Jedi, Luke commands C-3PO and R2-D2 to deliver a message to Jabba the Hutt, whereby C-3PO is used as Jabbas translator while R2-D2 serves on his flying sail barge. C-3POs first translation for the lord is of the bounty hunter Boushh — Leia in disguise — claiming the bounty for Chewbacca. When Luke attempts escape, R2-D2 tosses him his lightsaber, with which he attacks Jabbas guards, in the midst of the battle, C-3PO is attacked by Salacious Crumb, who pulls out his right photoreceptor before being driven off by R2-D2
36.
Star Wars
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Star Wars is an American epic space opera franchise, centered on a film series created by George Lucas. It depicts the adventures of characters a long time ago in a galaxy far, far away. The franchise began in 1977 with the release of the film Star Wars and it was followed by the successful sequels The Empire Strikes Back and Return of the Jedi, these three films constitute the original Star Wars trilogy. A prequel trilogy was released between 1999 and 2005, which received mixed-to-negative reactions, a sequel trilogy began in 2015 with the release of Star Wars, The Force Awakens. All seven films were nominated for Academy Awards and have been successes, with a combined box office revenue of over $7.5 billion. Spin-off films include Star Wars, The Clone Wars and Rogue One, Star Wars also holds a Guinness World Records title for the Most successful film merchandising franchise. In 2015, the value of the Star Wars franchise was estimated at USD $42 billion. In 2012, The Walt Disney Company acquired Lucasfilm for $4.06 billion and earned the rights to all subsequent Star Wars films. Walt Disney Studios owns digital distribution rights to all the Star Wars films, the events depicted in the Star Wars franchise take place in an unnamed fictional galaxy at an undetermined point in the distant past. Many species of creatures are depicted. Robotic droids are also commonplace and are built to serve their owners. Space travel is common, and many planets in the galaxy are members of a single galactic government. In the prequel trilogy, this is depicted in the form of the Galactic Republic, at the end of the trilogy and throughout the original trilogy. Preceding and during the trilogy, this government is the New Republic. One of the prominent elements of Star Wars is the Force and it is described in the first produced film as an energy field created by all living things surrounds us, penetrates us, binds the galaxy together. While the Force can be used for good, known as the side, it also has a dark side that, when pursued, imbues users with hatred, aggression. The sequel trilogy introduces the Knights of Ren, an order of practitioners of the side of the Force aligned with the First Order. The first film in the series, Star Wars, was released on May 25,1977 and this was followed by two sequels, The Empire Strikes Back, released on May 21,1980, and Return of the Jedi, released on May 25,1983
37.
Chorus effect
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In music, a chorus effect occurs when individual sounds with approximately the same timbre, and very similar pitch converge and are perceived as one. When the effect is produced successfully, none of the constituent sounds are perceived as being out of tune and it is characteristic of sounds with a rich, shimmering quality that would be absent if the sound came from a single source. The shimmer occurs because of beating, the effect is more apparent when listening to sounds that sustain for longer periods of time. The chorus effect is especially easy to hear when listening to a choir or string ensemble, a choir has multiple people singing each part. A string ensemble has multiple violinists and possibly multiples of other stringed instruments, in spite of the name, most electronic chorus effects do not accurately emulate this acoustic ensemble effect. Instead, they create a moving electronic shimmer. Some instruments produce an effect as part of their design, however. Piano - Each hammer strikes multiple strings tuned to nearly the same pitch, professional piano tuners carefully control the mistuning of each string to add movement without losing clarity. 12 string guitar - Six pairs of strings tuned in octaves, synthesizer - The effect can be achieved by using multiple, slightly detuned oscillators for each note, or by passing all the notes played through a separate electronic chorus circuit. Mandolin -4 pairs of identically tuned strings, as opposed to octaves, accordion - two or three reed blocks tuned to nearly the same pitch produce a unique and distinctive sound exclusive to the accordion. The chorus effect can be simulated by signal processing equipment, the signal processor may be software running on a computer, software running in a digital effect processor, or an analog effect processor. If the processor is hardware-based, it may be packaged as a pedal, a rack-mount module, some keyboard instruments have an electronic chorus effect built in, such as some electronic pianos and some Hammond organs. Regardless of the technology or form factor, the processor achieves the effect by taking an audio signal, the pitch of the added voices is typically modulated by an LFO, which makes the overall effect similar to that of a flanger, except with longer delays and without feedback. Stereo chorus effect processors produce the effect, but it is varied between the left and right channels by offsetting the delay or phase of the LFO. The effect is enhanced because sounds are produced from multiple locations in the stereo field. Used on instruments like clean electric guitar and keyboards, it can yield very dreamy or ambient sounds, examples of the use of obviously chorused guitar tracks include Fripp & Enos Evensong, Nirvanas Come As You Are, Mike Sterns Swunk, and Satellite Partys Mr. Sunshine. The voix céleste is an organ consisting of either one or two ranks of pipes slightly out of tune. The term celeste refers to a rank of pipes detuned slightly so as to produce an effect when combined with a normally tuned rank
38.
Attenuation
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In physics, attenuation is the gradual loss in intensity of any kind of flux through a medium. For instance, dark glasses attenuate sunlight, lead attenuates X-rays, in electrical engineering and telecommunications, attenuation affects the propagation of waves and signals in electrical circuits, in optical fibers, and in air. Electrical attenuators and optical attenuators are commonly manufactured components in this field, in many cases, attenuation is an exponential function of the path length through the medium. In chemical spectroscopy, this is known as the Beer–Lambert law, in engineering, attenuation is usually measured in units of decibels per unit length of medium and is represented by the attenuation coefficient of the medium in question. Attenuation also occurs in earthquakes, when the waves move farther away from the epicenter. One area of research in which figures strongly is in ultrasound physics. Attenuation in ultrasound is the reduction in amplitude of the beam as a function of distance through the imaging medium. Accounting for attenuation effects in ultrasound is important because a reduced signal amplitude can affect the quality of the image produced. By knowing the attenuation that an ultrasound beam experiences traveling through a medium, ultrasound attenuation measurement in heterogeneous systems, like emulsions or colloids, yields information on particle size distribution. There is an ISO standard on this technique, ultrasound attenuation can be used for extensional rheology measurement. There are acoustic rheometers that employ Stokes law for measuring extensional viscosity, wave equations which take acoustic attenuation into account can be written on a fractional derivative form, see the article on acoustic attenuation or e. g. the survey paper. Attenuation coefficients are used to different media according to how strongly the transmitted ultrasound amplitude decreases as a function of frequency. Attenuation coefficients vary widely for different media, in biomedical ultrasound imaging however, biological materials and water are the most commonly used media. Ultrasound propagation through homogeneous media is associated only with absorption and can be characterized with absorption coefficient only, propagation through heterogeneous media requires taking into account scattering. When the sun’s radiation reaches the sea-surface, the radiation is attenuated by the water. The intensity of light at depth can be calculated using the Beer-Lambert Law, in clear open waters, visible light is absorbed at the longest wavelengths first. Thus, red, orange, and yellow wavelengths are absorbed at higher water depths, because the blue and violet wavelengths are absorbed last compared to the other wavelengths, open ocean waters appear deep-blue to the eye. In near-shore waters, sea water contains more phytoplankton than the very clear central ocean waters, chlorophyll-a pigments in the phytoplankton absorb light, and the plants themselves scatter light, making coastal waters less clear than open waters
39.
Low-pass filter
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A low-pass filter is a filter that passes signals with a frequency lower than a certain cutoff frequency and attenuates signals with frequencies higher than the cutoff frequency. The exact frequency response of the filter depends on the filter design, the filter is sometimes called a high-cut filter, or treble cut filter in audio applications. A low-pass filter is the complement of a high-pass filter, Low-pass filters provide a smoother form of a signal, removing the short-term fluctuations, and leaving the longer-term trend. Filter designers will often use the form as a prototype filter. That is, a filter with unity bandwidth and impedance, the desired filter is obtained from the prototype by scaling for the desired bandwidth and impedance and transforming into the desired bandform. Examples of low-pass filters occur in acoustics, optics and electronics, a stiff physical barrier tends to reflect higher sound frequencies, and so acts as a low-pass filter for transmitting sound. When music is playing in another room, the low notes are easily heard, an optical filter with the same function can correctly be called a low-pass filter, but conventionally is called a longpass filter, to avoid confusion. For current signals, a circuit, using a resistor and capacitor in parallel. Electronic low-pass filters are used on inputs to subwoofers and other types of loudspeakers, radio transmitters use low-pass filters to block harmonic emissions that might interfere with other communications. The tone knob on many electric guitars is a filter used to reduce the amount of treble in the sound. An integrator is another time constant low-pass filter, telephone lines fitted with DSL splitters use low-pass and high-pass filters to separate DSL and POTS signals sharing the same pair of wires. Low-pass filters also play a significant role in the sculpting of sound created by analogue, the transition region present in practical filters does not exist in an ideal filter. The filter would therefore need to have infinite delay, or knowledge of the future and past. It is effectively realizable for pre-recorded digital signals by assuming extensions of zero into the past and future, or more typically by making the signal repetitive and this delay is manifested as phase shift. Greater accuracy in approximation requires a longer delay, an ideal low-pass filter results in ringing artifacts via the Gibbs phenomenon. These can be reduced or worsened by choice of windowing function, for example, simple truncation causes severe ringing artifacts, in signal reconstruction, and to reduce these artifacts one uses window functions which drop off more smoothly at the edges. The Whittaker–Shannon interpolation formula describes how to use a perfect low-pass filter to reconstruct a signal from a sampled digital signal. Real digital-to-analog converters use real filter approximations, there are many different types of filter circuits, with different responses to changing frequency
40.
High-pass filter
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A high-pass filter is an electronic filter that passes signals with a frequency higher than a certain cutoff frequency and attenuates signals with frequencies lower than the cutoff frequency. The amount of attenuation for each frequency depends on the filter design, a high-pass filter is usually modeled as a linear time-invariant system. It is sometimes called a filter or bass-cut filter. High-pass filters have many uses, such as blocking DC from circuitry sensitive to non-zero average voltages or radio frequency devices and they can also be used in conjunction with a low-pass filter to produce a bandpass filter. Figure 2 shows an active electronic implementation of a first-order high-pass filter using an operational amplifier and that is, high-frequency signals are inverted and amplified by R2/R1. Discrete-time high-pass filters can also be designed, discrete-time filter design is beyond the scope of this article, however, a simple example comes from the conversion of the continuous-time high-pass filter above to a discrete-time realization. That is, the behavior can be discretized. Substituting Equation into Equation and then Equation into Equation gives, V out = C ⏞ I R = R C This equation can be discretized, for simplicity, assume that samples of the input and output are taken at evenly spaced points in time separated by Δ T time. Let the samples of V in be represented by the sequence, the expression for parameter α yields the equivalent time constant R C in terms of the sampling period Δ T and α, R C = Δ T. If α =0.5, then the R C time constant equal to the sampling period, if α ≪0.5, then R C is significantly smaller than the sampling interval, and R C ≈ α Δ T. The filter recurrence relation provides a way to determine the output samples in terms of the input samples, in particular, A large α implies that the output will decay very slowly but will also be strongly influenced by even small changes in input. By the relationship between parameter α and time constant R C above, a large α corresponds to a large R C, hence, this case corresponds to a high-pass filter with a very narrow stop band. Because it is excited by changes and tends to hold its prior output values for a long time. However, a constant input will always decay to zero, as would be expected with a filter with a large R C. A small α implies that the output will decay quickly and will require changes in the input to cause the output to change much. By the relationship between parameter α and time constant R C above, a small α corresponds to a small R C, hence, this case corresponds to a high-pass filter with a very wide stop band. Because it requires large changes and tends to forget its prior output values, it can only pass relatively high frequencies. They are used as part of a crossover to direct high frequencies to a tweeter while attenuating bass signals which could interfere with, or damage
41.
Band-pass filter
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A band-pass filter is a device that passes frequencies within a certain range and rejects frequencies outside that range. An example of an analogue electronic band-pass filter is an RLC circuit and these filters can also be created by combining a low-pass filter with a high-pass filter. Bandpass is an adjective that describes a type of filter or filtering process, it is to be distinguished from passband, hence, one might say A dual bandpass filter has two passbands. A bandpass signal is a signal containing a band of frequencies not adjacent to zero frequency, an ideal bandpass filter would have a completely flat passband and would completely attenuate all frequencies outside the passband. Additionally, the out of the passband would have brickwall characteristics. In practice, no bandpass filter is ideal and this is known as the filter roll-off, and it is usually expressed in dB of attenuation per octave or decade of frequency. Generally, the design of a filter seeks to make the roll-off as narrow as possible, often, this is achieved at the expense of pass-band or stop-band ripple. The bandwidth of the filter is simply the difference between the upper and lower cutoff frequencies, optical band-pass filters are common in photography and theatre lighting work. These filters take the form of a transparent coloured film or sheet, a band-pass filter can be characterized by its Q factor. The Q-factor is the inverse of the fractional bandwidth, a high-Q filter will have a narrow passband and a low-Q filter will have a wide passband. These are respectively referred to as narrow-band and wide-band filters, Bandpass filters are widely used in wireless transmitters and receivers. The main function of such a filter in a transmitter is to limit the bandwidth of the signal to the band allocated for the transmission. This prevents the transmitter from interfering with other stations, in a receiver, a bandpass filter allows signals within a selected range of frequencies to be heard or decoded, while preventing signals at unwanted frequencies from getting through. A bandpass filter also optimizes the ratio and sensitivity of a receiver. Outside of electronics and signal processing, one example of the use of filters is in the atmospheric sciences. It is common to band-pass filter recent meteorological data with a range of, for example,3 to 10 days. In neuroscience, visual cortical simple cells were first shown by David Hubel and Torsten Wiesel to have properties that resemble Gabor filters. In astronomy, band-pass filters are used to only a single portion of the light spectrum into an instrument
42.
Band-stop filter
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In signal processing, a band-stop filter or band-rejection filter is a filter that passes most frequencies unaltered, but attenuates those in a specific range to very low levels. It is the opposite of a band-pass filter, a notch filter is a band-stop filter with a narrow stopband. Other names include band limit filter, T-notch filter, band-elimination filter, typically, the width of the stopband is 1 to 2 decades. However, in the band, a notch filter has high. Anti-hum filter For countries using 60 Hz power lines, Low Freq,59 Hz Middle Freq 60 Hz High Freq,61 Hz This means that the filter passes all frequencies, except for the range of 59–61 Hz. This would be used to out the mains hum from the 60 Hz power line. For countries where power transmission is at 50 Hz, the filter would have a 49–51 Hz range, non-linearities of power amplifiers When measuring the non-linearities of power amplifiers, a very narrow notch filter can be very useful to avoid the carrier frequency. Use of the filter may ensure that the input power of a spectrum analyser used to detect spurious content will not be exceeded. Wave trap A notch filter, usually a simple LC circuit, is used to remove a specific interfering frequency and this is a technique used with radio receivers that are so close to a transmitter that it swamps all other signals. The wave trap is used to remove, or greatly reduce, optical notch filters rely on destructive interference
