1.
Electricity
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Electricity is the set of physical phenomena associated with the presence of electric charge. Although initially considered a separate to magnetism, since the development of Maxwells Equations both are recognized as part of a single phenomenon, electromagnetism. Various common phenomena are related to electricity, including lightning, static electricity, electric heating, electric discharges, in addition, electricity is at the heart of many modern technologies. The presence of a charge, which can be either positive or negative. On the other hand, the movement of charges, which is known as electric current. When a charge is placed in a location with non-zero electric field, the magnitude of this force is given by Coulombs Law. Thus, if that charge were to move, the field would be doing work on the electric charge. Electrical phenomena have been studied since antiquity, though progress in theoretical understanding remained slow until the seventeenth and eighteenth centuries. Even then, practical applications for electricity were few, and it would not be until the nineteenth century that engineers were able to put it to industrial and residential use. The rapid expansion in electrical technology at this time transformed industry, electricitys extraordinary versatility means it can be put to an almost limitless set of applications which include transport, heating, lighting, communications, and computation. Electrical power is now the backbone of modern industrial society, long before any knowledge of electricity existed, people were aware of shocks from electric fish. Ancient Egyptian texts dating from 2750 BCE referred to these fish as the Thunderer of the Nile, Electric fish were again reported millennia later by ancient Greek, Roman and Arabic naturalists and physicians. Patients suffering from such as gout or headache were directed to touch electric fish in the hope that the powerful jolt might cure them. Ancient cultures around the Mediterranean knew that certain objects, such as rods of amber, Thales was incorrect in believing the attraction was due to a magnetic effect, but later science would prove a link between magnetism and electricity. He coined the New Latin word electricus to refer to the property of attracting small objects after being rubbed and this association gave rise to the English words electric and electricity, which made their first appearance in print in Thomas Brownes Pseudodoxia Epidemica of 1646. Further work was conducted by Otto von Guericke, Robert Boyle, Stephen Gray, in the 18th century, Benjamin Franklin conducted extensive research in electricity, selling his possessions to fund his work. In June 1752 he is reputed to have attached a key to the bottom of a dampened kite string. A succession of jumping from the key to the back of his hand showed that lightning was indeed electrical in nature
2.
Spectral density
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The power spectrum S x x of a time series x describes the distribution of power into frequency components composing that signal. According to Fourier analysis any physical signal can be decomposed into a number of discrete frequencies, the statistical average of a certain signal or sort of signal as analyzed in terms of its frequency content, is called its spectrum. When the energy of the signal is concentrated around a time interval, especially if its total energy is finite. More commonly used is the spectral density, which applies to signals existing over all time. The power spectral density then refers to the energy distribution that would be found per unit time. Summation or integration of the spectral components yields the total power or variance, identical to what would be obtained by integrating x 2 over the time domain, the spectrum of a physical process x often contains essential information about the nature of x. For instance, the pitch and timbre of an instrument are immediately determined from a spectral analysis. The color of a source is determined by the spectrum of the electromagnetic waves electric field E as it fluctuates at an extremely high frequency. Obtaining a spectrum from time series such as these involves the Fourier transform, however this article concentrates on situations in which the time series is known or directly measured. The power spectrum is important in signal processing and in the statistical study of stochastic processes, as well as in many other branches of physics. Typically the process is a function of time but one can similarly discuss data in the domain being decomposed in terms of spatial frequency. Any signal that can be represented as an amplitude that varies in time has a frequency spectrum. This includes familiar entities such as light, musical notes, radio/TV. When these signals are viewed in the form of a frequency spectrum, in some cases the frequency spectrum may include a distinct peak corresponding to a sine wave component. And additionally there may be corresponding to harmonics of a fundamental peak. In physics, the signal might be a wave, such as an electromagnetic wave, the power spectral density of the signal describes the power present in the signal as a function of frequency, per unit frequency. Power spectral density is expressed in watts per hertz. When a signal is defined in terms only of a voltage, for instance, in this case power is simply reckoned in terms of the square of the signal, as this would always be proportional to the actual power delivered by that signal into a given impedance
3.
Acoustics
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Acoustics is the interdisciplinary science that deals with the study of all mechanical waves in gases, liquids, and solids including topics such as vibration, sound, ultrasound and infrasound. A scientist who works in the field of acoustics is an acoustician while someone working in the field of technology may be called an acoustical engineer. The application of acoustics is present in almost all aspects of society with the most obvious being the audio. Hearing is one of the most crucial means of survival in the animal world, accordingly, the science of acoustics spreads across many facets of human society—music, medicine, architecture, industrial production, warfare and more. Likewise, animal species such as songbirds and frogs use sound, art, craft, science and technology have provoked one another to advance the whole, as in many other fields of knowledge. Robert Bruce Lindsays Wheel of Acoustics is a well accepted overview of the fields in acoustics. The word acoustic is derived from the Greek word ἀκουστικός, meaning of or for hearing, ready to hear and that from ἀκουστός, heard, audible, which in turn derives from the verb ἀκούω, I hear. The Latin synonym is sonic, after which the term used to be a synonym for acoustics. Frequencies above and below the range are called ultrasonic and infrasonic. If, for example, a string of a length would sound particularly harmonious with a string of twice the length. In modern parlance, if a string sounds the note C when plucked, a string twice as long will sound a C an octave lower. In one system of tuning, the tones in between are then given by 16,9 for D,8,5 for E,3,2 for F,4,3 for G,6,5 for A. Aristotle understood that sound consisted of compressions and rarefactions of air which falls upon, a very good expression of the nature of wave motion. The physical understanding of acoustical processes advanced rapidly during and after the Scientific Revolution, mainly Galileo Galilei but also Marin Mersenne, independently, discovered the complete laws of vibrating strings. Experimental measurements of the speed of sound in air were carried out successfully between 1630 and 1680 by a number of investigators, prominently Mersenne, meanwhile, Newton derived the relationship for wave velocity in solids, a cornerstone of physical acoustics. The eighteenth century saw advances in acoustics as mathematicians applied the new techniques of calculus to elaborate theories of sound wave propagation. Also in the 19th century, Wheatstone, Ohm, and Henry developed the analogy between electricity and acoustics, the twentieth century saw a burgeoning of technological applications of the large body of scientific knowledge that was by then in place. The first such application was Sabine’s groundbreaking work in architectural acoustics, Underwater acoustics was used for detecting submarines in the first World War
4.
Optics
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Optics is the branch of physics which involves the behaviour and properties of light, including its interactions with matter and the construction of instruments that use or detect it. Optics usually describes the behaviour of visible, ultraviolet, and infrared light, because light is an electromagnetic wave, other forms of electromagnetic radiation such as X-rays, microwaves, and radio waves exhibit similar properties. Most optical phenomena can be accounted for using the classical description of light. Complete electromagnetic descriptions of light are, however, often difficult to apply in practice, practical optics is usually done using simplified models. The most common of these, geometric optics, treats light as a collection of rays that travel in straight lines, physical optics is a more comprehensive model of light, which includes wave effects such as diffraction and interference that cannot be accounted for in geometric optics. Historically, the model of light was developed first, followed by the wave model of light. Progress in electromagnetic theory in the 19th century led to the discovery that waves were in fact electromagnetic radiation. Some phenomena depend on the fact that light has both wave-like and particle-like properties, explanation of these effects requires quantum mechanics. When considering lights particle-like properties, the light is modelled as a collection of particles called photons, quantum optics deals with the application of quantum mechanics to optical systems. Optical science is relevant to and studied in many related disciplines including astronomy, various engineering fields, photography, practical applications of optics are found in a variety of technologies and everyday objects, including mirrors, lenses, telescopes, microscopes, lasers, and fibre optics. Optics began with the development of lenses by the ancient Egyptians and Mesopotamians, the earliest known lenses, made from polished crystal, often quartz, date from as early as 700 BC for Assyrian lenses such as the Layard/Nimrud lens. The ancient Romans and Greeks filled glass spheres with water to make lenses, the word optics comes from the ancient Greek word ὀπτική, meaning appearance, look. Greek philosophy on optics broke down into two opposing theories on how vision worked, the theory and the emission theory. The intro-mission approach saw vision as coming from objects casting off copies of themselves that were captured by the eye, plato first articulated the emission theory, the idea that visual perception is accomplished by rays emitted by the eyes. He also commented on the parity reversal of mirrors in Timaeus, some hundred years later, Euclid wrote a treatise entitled Optics where he linked vision to geometry, creating geometrical optics. Ptolemy, in his treatise Optics, held a theory of vision, the rays from the eye formed a cone, the vertex being within the eye. The rays were sensitive, and conveyed back to the observer’s intellect about the distance. He summarised much of Euclid and went on to describe a way to measure the angle of refraction, during the Middle Ages, Greek ideas about optics were resurrected and extended by writers in the Muslim world
5.
Transducer
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A transducer is a device that converts one form of energy to another. Usually a transducer converts a signal in one form of energy to a signal in another, Transducers are often employed at the boundaries of automation, measurement, and control systems, where electrical signals are converted to and from other physical quantities. The process of converting one form of energy to another is known as transduction, passive sensors require an external power source to operate, which is called an excitation signal. The signal is modulated by the sensor to produce an output signal, active sensors generate electric signals in response to an external stimulus without the need of an additional energy source. Such examples are a photodiode, and a sensor, thermocouple. A sensor is a device that receives and responds to a signal or stimulus, transducer is the other term that is sometimes interchangeably used instead of the term sensor, although there are subtle differences. A transducer is a term that can be used for the definition of many such as sensors, actuators. An actuator is a device that is responsible for moving or controlling a mechanism or system and it is operated by a source of energy, which can be mechanical force, electrical current, hydraulic fluid pressure, or pneumatic pressure, and converts that energy into motion. An actuator is the mechanism by which a system acts upon an environment. The control system can be simple, software-based, a human, bidirectional transducers convert physical phenomena to electrical signals and also convert electrical signals into physical phenomena. Likewise, DC electric motors may be used to generate electrical power if the shaft is turned by an external torque. Radio transmitters converts electrical signals to electromagnetic transmissions, horn analyzer List of sensors Tactile sensor Agarwal, Anant. Foundations of Analog and Digital Electronic Circuits. Department of Electrical Engineering and Computer Science, Massachusetts Institute of Technology,2005, p.43. Introduction to Closed Loop Hall Effect Current Transducers Federal Standard 1037C, August 7,1996, transducer A sound transducer with a flat flexible diaphragm working with bending waves
6.
Monochromator
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A monochromator is an optical device that transmits a mechanically selectable narrow band of wavelengths of light or other radiation chosen from a wider range of wavelengths available at the input. The name is from the Greek roots mono-, single, and chroma, colour, a device that can produce monochromatic light has many uses in science and in optics because many optical characteristics of a material are dependent on wavelength. Although there are a number of ways to select a narrow band of wavelengths. See below for a discussion of some of the uses of monochromators, in hard X-ray and neutron optics, crystal monochromators are used to define wave conditions on the instruments. A monochromator can use either the phenomenon of optical dispersion in a prism, or that of using a diffraction grating. It usually has a mechanism for directing the selected color to an exit slit, usually the grating or the prism is used in a reflective mode. A reflective prism is made by making a right prism with one side mirrored. The light enters through the face and is reflected back through it. The total refraction, and the dispersion, is the same as would occur if an equilateral prism were used in transmission mode. The dispersion or diffraction is only if the light is collimated. A source, like the sun, which is far away. Newton used sunlight in his famous experiments, in a practical monochromator, however, the light source is close by, and an optical system in the monochromator converts the diverging light of the source to collimated light. Although some monochromator designs do use focusing gratings that do not need separate collimators, reflective optics are preferred because they do not introduce dispersive effects of their own. In the common Czerny–Turner design, the illumination source is aimed at an entrance slit. The amount of energy available for use depends on the intensity of the source in the space defined by the slit. The slit is placed at the focus of a curved mirror so that the light from the slit reflected from the mirror is collimated. The collimated light is diffracted from the grating and then is collected by another mirror, in a prism monochromator, a reflective prism takes the place of the diffraction grating, in which case the light is refracted by the prism. At the exit slit, the colors of the light are spread out, because each color arrives at a separate point in the exit-slit plane, there are a series of images of the entrance slit focused on the plane
7.
Frequency
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Frequency is the number of occurrences of a repeating event per unit time. It is also referred to as frequency, which emphasizes the contrast to spatial frequency. The period is the duration of time of one cycle in a repeating event, for example, if a newborn babys heart beats at a frequency of 120 times a minute, its period—the time interval between beats—is half a second. Frequency is an important parameter used in science and engineering to specify the rate of oscillatory and vibratory phenomena, such as vibrations, audio signals, radio waves. For cyclical processes, such as rotation, oscillations, or waves, in physics and engineering disciplines, such as optics, acoustics, and radio, frequency is usually denoted by a Latin letter f or by the Greek letter ν or ν. For a simple motion, the relation between the frequency and the period T is given by f =1 T. The SI unit of frequency is the hertz, named after the German physicist Heinrich Hertz, a previous name for this unit was cycles per second. The SI unit for period is the second, a traditional unit of measure used with rotating mechanical devices is revolutions per minute, abbreviated r/min or rpm. As a matter of convenience, longer and slower waves, such as ocean surface waves, short and fast waves, like audio and radio, are usually described by their frequency instead of period. Spatial frequency is analogous to temporal frequency, but the axis is replaced by one or more spatial displacement axes. Y = sin = sin d θ d x = k Wavenumber, in the case of more than one spatial dimension, wavenumber is a vector quantity. For periodic waves in nondispersive media, frequency has a relationship to the wavelength. Even in dispersive media, the frequency f of a wave is equal to the phase velocity v of the wave divided by the wavelength λ of the wave. In the special case of electromagnetic waves moving through a vacuum, then v = c, where c is the speed of light in a vacuum, and this expression becomes, f = c λ. When waves from a monochrome source travel from one medium to another, their remains the same—only their wavelength. For example, if 71 events occur within 15 seconds the frequency is, the latter method introduces a random error into the count of between zero and one count, so on average half a count. This is called gating error and causes an error in the calculated frequency of Δf = 1/, or a fractional error of Δf / f = 1/ where Tm is the timing interval. This error decreases with frequency, so it is a problem at low frequencies where the number of counts N is small, an older method of measuring the frequency of rotating or vibrating objects is to use a stroboscope
8.
Electric power
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Electric power is the rate, per unit time, at which electrical energy is transferred by an electric circuit. The SI unit of power is the watt, one joule per second, Electric power is usually produced by electric generators, but can also be supplied by sources such as electric batteries. It is usually supplied to businesses and homes by the power industry through an electric power grid. Electric power is sold by the kilowatt hour which is the product of power in kilowatts multiplied by running time in hours. Electric utilities measure power using an electricity meter, which keeps a total of the electric energy delivered to a customer. Electrical power provides a low form of energy and can be carried long distances and converted into other forms of energy such as motion. Electric power, like power, is the rate of doing work, measured in watts. The term wattage is used colloquially to mean electric power in watts, the potential energy of the charges due to the voltage between the terminals is converted to kinetic energy in the device. These devices are called passive components or loads, they consume electric power from the circuit, converting it to other forms of such as mechanical work, heat, light. Examples are electrical appliances, such as bulbs, electric motors. In alternating current circuits the direction of the voltage periodically reverses, devices in which this occurs are called active devices or power sources, such as electric generators and batteries. Some devices can be either a source or a load, depending on the voltage, for example, a rechargeable battery acts as a source when it provides power to a circuit, but as a load when it is connected to a battery charger and is being recharged. Since electric power can flow either into or out of a component, Electric power flowing out of a circuit into a component is arbitrarily defined to have a positive sign, while power flowing into a circuit from a component is defined to have a negative sign. Thus passive components have positive power consumption, while power sources have negative power consumption and this is called the passive sign convention. In alternating current circuits, energy storage such as inductance and capacitance may result in periodic reversals of the direction of energy flow. The portion of power flow that, averaged over a cycle of the AC waveform. That portion of power due to stored energy, that returns to the source in each cycle, is known as reactive power. Real power is represented as a vector and reactive power is represented as a vertical vector
9.
Distortion
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Distortion is the alteration of the original shape of something, such as an object, image, sound or waveform. Distortion is usually unwanted, and so engineers strive to eliminate distortion, in some situations, however, distortion may be desirable. The important signal processing operation of heterodyning is based on nonlinear mixing of signals to cause intermodulation, Distortion is also used as a musical effect, particularly with electric guitars. The addition of noise or other outside signals is not deemed distortion, a quality measure that explicitly reflects both the noise and the distortion is the Signal-to-noise-and-distortion ratio. Distortion occurs when the transfer function F is more complicated than this, if F is a linear function, for instance a filter whose gain and/or delay varies with frequency, the signal suffers linear distortion. Linear distortion does not introduce new frequency components to a signal and this diagram shows the behaviour of a signal as it is passed through various distorting functions. The first trace shows the input and it also shows the output from a non-distorting transfer function. A high-pass filter distorts the shape of a wave by reducing its low frequency components. This is the cause of the droop seen on the top of the pulses and this pulse distortion can be very significant when a train of pulses must pass through an AC-coupled amplifier. As the sine wave contains only one frequency, its shape is unaltered, a low-pass filter rounds the pulses by removing the high frequency components. All systems are low pass to some extent, note that the phase of the sine wave is different for the lowpass and the highpass cases, due to the phase distortion of the filters. A slightly non-linear transfer function, this one gently compresses the peaks of the sine wave and this generates small amounts of low order harmonics. A hard-clipping transfer function generates high order harmonics, parts of the transfer function are flat, which indicates that all information about the input signal has been lost in this region. The transfer function of an amplifier, with perfect gain. The true behavior of the system is usually different, amplitude distortion is distortion occurring in a system, subsystem, or device when the output amplitude is not a linear function of the input amplitude under specified conditions. Harmonic distortion adds overtones that are whole number multiples of a sound waves frequencies, nonlinearities that give rise to amplitude distortion in audio systems are most often measured in terms of the harmonics added to a pure sinewave fed to the system. The level at which harmonic distortion becomes audible depends on the nature of the distortion. Different types of distortion are more audible than others if the THD measurements are identical
10.
Harmonic
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A harmonic is any member of the harmonic series, the divergent infinite series. Every term of the series after the first is the mean of the neighboring terms. The phrase harmonic mean likewise derives from music, the term is employed in various disciplines, including music, physics, acoustics, electronic power transmission, radio technology, and other fields. It is typically applied to repeating signals, such as sinusoidal waves, a harmonic of such a wave is a wave with a frequency that is a positive integer multiple of the frequency of the original wave, known as the fundamental frequency. The original wave is called the 1st harmonic, the following harmonics are known as higher harmonics. As all harmonics are periodic at the frequency, the sum of harmonics is also periodic at that frequency. On strings, harmonics that are bowed have a glassy, pure tone, harmonics may also be called overtones, partials or upper partials. In some music contexts, the harmonic, overtone and partial are used fairly interchangeably. Most acoustic instruments emit complex tones containing many individual partials, rather, a musical note is perceived as one sound, the quality or timbre of that sound being a result of the relative strengths of the individual partials. Oscillators that produce harmonic partials behave somewhat like one-dimensional resonators, and are long and thin. Wind instruments whose air column is open at one end, such as trumpets and clarinets. However they only produce partials matching the odd harmonics, at least in theory, the reality of acoustic instruments is such that none of them behaves as perfectly as the somewhat simplified theoretical models would predict. Partials whose frequencies are not integer multiples of the fundamental are referred to as inharmonic partials, antique singing bowls are known for producing multiple harmonic partials or multiphonics. An overtone is any partial higher than the lowest partial in a compound tone, the relative strengths and frequency relationships of the component partials determine the timbre of an instrument. This chart demonstrates how the three types of names are counted, In many musical instruments, it is possible to play the upper harmonics without the note being present. In a simple case this has the effect of making the note go up in pitch by an octave, in some cases it also changes the timbre of the note. This is part of the method of obtaining higher notes in wind instruments. The extended technique of playing multiphonics also produces harmonics, on string instruments it is possible to produce very pure sounding notes, called harmonics or flageolets by string players, which have an eerie quality, as well as being high in pitch
11.
Bandwidth (signal processing)
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Bandwidth is the difference between the upper and lower frequencies in a continuous set of frequencies. It is typically measured in hertz, and may refer to passband bandwidth, sometimes to baseband bandwidth. Passband bandwidth is the difference between the upper and lower frequencies of, for example, a band-pass filter, a communication channel. In the case of a filter or baseband signal, the bandwidth is equal to its upper cutoff frequency. A key characteristic of bandwidth is that any band of a given width can carry the amount of information. For example, a 3 kHz band can carry a telephone conversation whether that band is at baseband or modulated to some higher frequency, Bandwidth is a key concept in many telecommunications applications. In radio communications, for example, bandwidth is the range occupied by a modulated carrier signal. An FM radio receivers tuner spans a range of frequencies. A government agency may apportion the regionally available bandwidth to broadcast license holders so that their signals do not mutually interfere, each transmitter owns a slice of bandwidth. For different applications there are different precise definitions, which are different for signals than for systems. One definition of bandwidth, for a system, could be the range of frequencies over which the system produces a level of performance. A less strict and more practically useful definition will refer to the frequencies beyond which frequency response is small, small could mean less than 3 dB below the maximum value, or more rarely 10 dB below, or it could mean below a certain absolute value. As with any definition of the width of a function, many definitions are suitable for different purposes, in some contexts, the signal bandwidth in hertz refers to the frequency range in which the signals spectral density is nonzero or above a small threshold value. That definition is used in calculations of the lowest sampling rate that will satisfy the sampling theorem, the threshold value is often defined relative to the maximum value, and is most commonly the 3dB point, that is the point where the spectral density is half its maximum value. The word bandwidth applies to signals as described above, but it could apply to systems. To say that a system has a certain bandwidth means that the system can process signals of that bandwidth, or that the system reduces the bandwidth of a white noise input to that bandwidth. If the maximum gain is 0 dB, the 3 dB bandwidth is the range where the gain is more than −3 dB. This is also the range of frequencies where the gain is above 70. 7% of the maximum amplitude gain
12.
Waveform
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A waveform is the shape and form of a signal such as a wave moving in a physical medium or an abstract representation. In many cases the medium in which the wave propagates does not permit a direct observation of the true form, in these cases, the term waveform refers to the shape of a graph of the varying quantity against time. An instrument called an oscilloscope can be used to represent a wave as a repeating image on a screen. To be more specific, a waveform is depicted by a graph that shows the changes in a signals amplitude over the duration of recording. The amplitude of the signal is measured on the y -axis, most programs show waveforms to give the user a visual aid of what has been recorded. If the waveform is of low or high height, the recording was most likely conducted under conditions with a low or high input volume, respectively. From this example, it follows that the represented by the waveform is affected by both the input signal and conditions under which it is recorded. A periodic waveforms include these while t is time, λ is wavelength, the amplitude of the waveform follows a trigonometric sine function with respect to time. Square wave = { a, mod λ < duty − a and this waveform is commonly used to represent digital information. A square wave of constant period contains odd harmonics that decrease at −6 dB/octave, triangle wave =2 a π arcsin sin 2 π t − ϕ λ. It contains odd harmonics that decrease at −12 dB/octave, sawtooth wave =2 a π arctan tan 2 π t − ϕ2 λ. This looks like the teeth of a saw, found often in time bases for display scanning. It is used as the point for subtractive synthesis, as a sawtooth wave of constant period contains odd. Other waveforms are often called composite waveforms and can often be described as a combination of a number of waves or other basis functions added together. The Fourier series describes the decomposition of periodic waveforms, such that any periodic waveform can be formed by the sum of a set of fundamental, finite-energy non-periodic waveforms can be analyzed into sinusoids by the Fourier transform. AC waveform Arbitrary waveform generator Spectrum analyzer Waveform monitor Waveform viewer Wave packet Yuchuan Wei, common Waveform Analysis, A New And Practical Generalization of Fourier Analysis. Springer US, Aug 31,2000 Waveform Definition Hao He, Jian Li, Waveform design for active sensing systems, a computational approach. Solomon W. Golomb, and Guang Gong, Signal design for good correlation, for wireless communication, cryptography, and radar
13.
Wireless
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Wireless communication, or sometimes simply wireless, is the transfer of information or power between two or more points that are not connected by an electrical conductor. The most common wireless technologies use radio waves, with radio waves distances can be short, such as a few meters for television or as far as thousands or even millions of kilometers for deep-space radio communications. It encompasses various types of fixed, mobile, and portable applications, including radios, cellular telephones, personal digital assistants. Somewhat less common methods of achieving wireless communications include the use of other wireless technologies, such as light, magnetic. The term wireless has been used twice in history, with slightly different meaning. It was initially used from about 1890 for the first radio transmitting and receiving technology, as in wireless telegraphy and this became its primary usage in the 2000s, due to the advent of technologies such as LTE, LTE-Advanced, Wi-Fi and Bluetooth. Wireless operations permit services, such as communications, that are impossible or impractical to implement with the use of wires. The term is used in the telecommunications industry to refer to telecommunications systems which use some form of energy to transfer information without the use of wires. Information is transferred in this manner over both short and long distances, similar to free-space optical communication, the photophone also required a clear line of sight between its transmitter and its receiver. It would be decades before the photophones principles found their first practical applications in military communications. Marconi soon developed a system that was transmitting signals way beyond distances anyone could have predicted, guglielmo Marconi and Karl Ferdinand Braun were awarded the 1909 Nobel Prize for Physics for their contribution to this form of wireless telegraphy. Free space means the light travel through the open air or outer space. This contrasts with other technologies that use light beams traveling through transmission lines such as optical fiber or dielectric light pipes. The technology is useful where physical connections are due to high costs or other considerations. Sonic, especially ultrasonic short range communication involves the transmission and reception of sound, electromagnetic induction has short range communication and power. This has been used in situations such as pacemakers, as well as for short-range Rfid tags. Common examples of wireless equipment include, Infrared and ultrasonic remote control devices Professional LMR and SMR typically used by business, industrial, consumer Two-way radio including FRS Family Radio Service, GMRS and Citizens band radios. Consumer and professional Marine VHF radios, airband and radio navigation equipment used by aviators and air traffic control Cellular telephones and pagers, provide connectivity for portable and mobile applications, both personal and business
14.
Oscilloscope
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Other signals can be converted to voltages and displayed. Oscilloscopes are used to observe the change of a signal over time, such that voltage. The observed waveform can be analyzed for such properties as amplitude, frequency, rise time, time interval, distortion, modern digital instruments may calculate and display these properties directly. Originally, calculation of these values required manually measuring the waveform against the scales built into the screen of the instrument, the oscilloscope can be adjusted so that repetitive signals can be observed as a continuous shape on the screen. A storage oscilloscope allows single events to be captured by the instrument and displayed for a long time. Oscilloscopes are used in the sciences, medicine, engineering, automotive, general-purpose instruments are used for maintenance of electronic equipment and laboratory work. Special-purpose oscilloscopes may be used for purposes as analyzing an automotive ignition system or to display the waveform of the heartbeat as an electrocardiogram. Early oscilloscopes used cathode ray tubes as their display element and linear amplifiers for signal processing, storage oscilloscopes used special storage CRTs to maintain a steady display of a single brief signal. CROs were later superseded by digital storage oscilloscopes with thin panel displays, fast analog-to-digital converters. DSOs without integrated displays are available at lower cost and use a digital computer to process. The basic oscilloscope, as shown in the illustration, is divided into four sections. The display is usually a CRT or LCD panel which is out with both horizontal and vertical reference lines referred to as the graticule. In addition to the screen, most display sections are equipped with three controls, a focus knob, an intensity knob and a beam finder button. The vertical section controls the amplitude of the displayed signal and this section carries a Volts-per-Division selector knob, an AC/DC/Ground selector switch and the vertical input for the instrument. Additionally, this section is equipped with the vertical beam position knob. The horizontal section controls the base or sweep of the instrument. The primary control is the Seconds-per-Division selector switch, also included is a horizontal input for plotting dual X-Y axis signals. The horizontal beam position knob is located in this section
15.
Fast Fourier transform
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A fast Fourier transform algorithm computes the discrete Fourier transform of a sequence, or its inverse. Fourier analysis converts a signal from its domain to a representation in the frequency domain. An FFT rapidly computes such transformations by factorizing the DFT matrix into a product of sparse factors. As a result, it manages to reduce the complexity of computing the DFT from O, which if one simply applies the definition of DFT, to O. Fast Fourier transforms are used for many applications in engineering, science. The basic ideas were popularized in 1965, but some algorithms had been derived as early as 1805, the DFT is obtained by decomposing a sequence of values into components of different frequencies. This operation is useful in many fields but computing it directly from the definition is too slow to be practical. The difference in speed can be enormous, especially for data sets where N may be in the thousands or millions. In practice, the time can be reduced by several orders of magnitude in such cases. The best-known FFT algorithms depend upon the factorization of N, but there are FFTs with O complexity for all N, even for prime N. Since the inverse DFT is the same as the DFT, but with the sign in the exponent. The development of fast algorithms for DFT can be traced to Gausss unpublished work in 1805 when he needed it to interpolate the orbit of asteroids Pallas and Juno from sample observations. His method was similar to the one published in 1965 by Cooley and Tukey. While Gausss work predated even Fouriers results in 1822, he did not analyze the computation time, between 1805 and 1965, some versions of FFT were published by other authors. Yates in 1932 published his version called interaction algorithm, which provided efficient computation of Hadamard, yates algorithm is still used in the field of statistical design and analysis of experiments. In 1942, Danielson and Lanczos published their version to compute DFT for x-ray crystallography, Cooley and Tukey published a more general version of FFT in 1965 that is applicable when N is composite and not necessarily a power of 2. To analyze the output of these sensors, a fast Fourier transform algorithm would be needed, garwin gave Tukeys idea to Cooley for implementation. Cooley and Tukey published the paper in a short six months
16.
Distributed element filter
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Its purpose is to allow a range of signal frequencies to pass, but to block others. That model is simple, but it becomes increasingly unreliable as the frequency of the signal increases. The distributed element model applies at all frequencies, and is used in transmission line theory, in the distributed view of circuits, the elements are distributed along the length of conductors and are inextricably mixed together. The filter design is concerned only with inductance and capacitance. There is no precise frequency above which distributed element filters must be used, distributed element filters may be constructed to have any of the bandforms possible with lumped elements with the exception of high-pass, which is usually only approximated. All filter classes used in lumped element designs can be implemented using a distributed element approach, there are many component forms used to construct distributed element filters, but all have the common property of causing a discontinuity on the transmission line. The development of distributed element filters was spurred on by the military need for radar, lumped element analogue filters had long before been developed but these new military systems operated at microwave frequencies and new filter designs were required. Nowadays the technology can be found in several mass-produced consumer items, the symbol λ is used to mean the wavelength of the signal being transmitted on the line or a section of line of that electrical length. Distributed element filters are used at frequencies above the VHF band. The exact point at which distributed element modelling becomes necessary depends on the design under consideration. A common rule of thumb is to apply distributed element modelling when component dimensions are larger than 0. 1λ, the increasing miniaturisation of electronics has meant that circuit designs are becoming ever smaller compared to λ. The frequencies beyond which a distributed element approach to filter design becomes necessary are becoming higher as a result of these advances. On the other hand, antenna structure dimensions are comparable to λ in all frequency bands. These spurious passbands are undesirable in most cases, for clarity of presentation, the diagrams in this article are drawn with the components implemented in stripline format. This does not imply an industry preference, although planar transmission line formats are popular because they can be implemented using established printed circuit board manufacturing techniques. The open wire implementations, for instance, of a number of structures are shown in the column of figure 3. Planar transmission lines are used in integrated circuit designs. Development of distributed element filters began in the years before World War II, a major paper on the subject was published by Mason and Sykes in 1937
17.
Superheterodyne receiver
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It was invented by US engineer Edwin Armstrong in 1918 during World War I. Virtually all modern radio receivers use the superheterodyne principle, Superheterodyne is a contraction of supersonic heterodyne, where supersonic indicates frequencies above the range of human hearing. The word heterodyne is derived from the Greek roots hetero- different, however, the signal from a continuous wave transmitter did not have any such inherent modulation and Morse Code from one of those would only be heard as a series of clicks or thumps. Fessendens idea was to run two Alexanderson alternators, one producing a carrier frequency 3 kHz higher than the other. In the receivers detector the two carriers would beat together to produce a 3 kHz tone thus in the headphones the Morse signals would then be heard as a series of 3 kHz beeps, for this he coined the term heterodyne meaning generated by a difference. The superheterodyne principle was devised in 1918 by U. S. Army Major Edwin Armstrong in France during World War I and he invented this receiver as a means of overcoming the deficiencies of early vacuum tube triodes used as high-frequency amplifiers in radio direction finding equipment. The strategic value was so high, however, that the British Admiralty felt the high cost was justified, Armstrong realized that if radio direction-finding receivers could be operated at a higher frequency, this would allow better detection of enemy shipping. However, at time, no practical short wave amplifier existed. Armstrong eventually deduced that this was caused by a supersonic heterodyne between the carrier frequency and the regenerative receivers oscillation frequency. Thus if a station was transmitting on 300 kHz and the receiver was set to 400 kHz, the station would be heard not only at the original 300 kHz. Armstrong realized that this was a solution to the short wave amplification problem, since the beat frequency still retained its original modulation. He termed this the intermediate frequency often abbreviated to IF, in December 1919, Major E. H. Armstrong gave publicity to an indirect method of obtaining short-wave amplification, called the super-heterodyne. Armstrong was able to put his ideas into practice, and the technique was adopted by the military. For early domestic radios, tuned radio frequency receivers were more popular because they were cheaper, easier for an owner to use. Armstrong eventually sold his patent to Westinghouse, who then sold it to RCA. Early superheterodyne receivers used IFs as low as 20 kHz, often based on the self-resonance of iron-cored transformers and this made them extremely susceptible to image frequency interference, but at the time, the main objective was sensitivity rather than selectivity. Using this technique, a number of triodes could be made to do the work that formerly required dozens of triodes. By the mid-1930s however, superheterodynes were using much higher frequencies, with tuned coils similar in construction to the aerial
18.
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
19.
Voltage-controlled oscillator
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A voltage-controlled oscillator or VCO is an electronic oscillator whose oscillation frequency is controlled by a voltage input. The applied input voltage determines the oscillation frequency. Consequently, modulating signals applied to control input may cause frequency modulation or phase modulation, a VCO may also be part of a phase-locked loop. VCOs can be categorized into two groups based on the type of waveform produced. Linear or harmonic oscillators generate a sinusoidal waveform, harmonic oscillators in electronics usually consist of a resonator with an amplifier that replaces the resonator losses and isolates the resonator from the output. Some examples of harmonic oscillators are LC-tank oscillators and crystal oscillators, in a voltage-controlled oscillator, a voltage input controls the resonant frequency. A varactor diodes capacitance is controlled by the voltage across the diode, consequently, a varactor can be used to change the capacitance of an LC tank. A varactor can also change the resonant frequency of a crystal resonator, relaxation oscillators can generate a sawtooth or triangular waveform. They are commonly used in integrated circuits. They can provide a range of operational frequencies with a minimal number of external components. Relaxation oscillator VCOs can have three topologies, 1) grounded-capacitor VCOs, 2) emitter-coupled VCOs, and 3) delay-based ring VCOs, the first two of these types operate similarly. The time spent in each depends on the rate of charge or discharge of a capacitor. The delay-based ring VCO operates somewhat differently however, for this type, the gain stages are connected in a ring. The output frequency is then a function of the delay in each stage, harmonic oscillator VCOs have these advantages over relaxation oscillators, Frequency stability with respect to temperature, noise, and power supply is much better for harmonic oscillator VCOs. They have good accuracy for frequency control since the frequency is controlled by a crystal or tank circuit, a disadvantage of harmonic oscillator VCOs is that they cannot be easily implemented in monolithic ICs. Relaxation oscillator VCOs are better suited for this technology, relaxation VCOs are also tunable over a wider range of frequencies. A voltage-controlled capacitor is one method of making an LC oscillator vary its frequency in response to a control voltage, any reverse-biased semiconductor diode displays a measure of voltage-dependent capacitance and can be used to change the frequency of an oscillator by varying a control voltage applied to the diode. Special-purpose variable capacitance varactor diodes are available with well-characterized wide-ranging values of capacitance, such devices are very convenient in the manufacture of voltage-controlled oscillators
20.
Periodogram
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In signal processing, a periodogram is an estimate of the spectral density of a signal. The term was coined by Arthur Schuster in 1898, today, the periodogram is a component of more sophisticated methods. It is the most common tool for examining the amplitude vs frequency characteristics of FIR filters, FFT spectrum analyzers are also implemented as a time-sequence of periodograms. There are at least two different definitions in use today, one of them involves time-averaging, and one does not. Time-averaging is also the purview of other articles, so this article is not about that. The definition of interest here is that the spectral density of a continuous function, x, is the Fourier transform of its auto-correlation function. When evaluated for all integers, k, between 0 and N-1, the array, S = | ∑ N x N ⋅ e − i 2 π k n N |2 is a periodogram, the parameter N is known to Matlab and Octave users as NFFT. And when it is used to implement a bank, N is several sub-multiples of the non-zero duration of the x sequence. One of the deficiencies is that the variance at a given frequency does not decrease as the number of samples used in the computation increases. It does not provide the averaging needed to analyze noiselike signals or even sinusoids at low signal-to-noise ratios, window functions and filter impulse responses are noiseless, but many other signals require more sophisticated methods of spectral estimation. It computes a windowed periodogram of each one, and computes an array average, for stationary processes, this reduces the noise variance of each element by approximately a factor equal to the reciprocal of the number of periodograms. Smoothing is a technique in frequency, instead of time. The smoothed periodogram is sometimes referred to as a spectral plot, periodogram-based techniques introduce small biases that are unacceptable in some applications. Other techniques that do not rely on periodograms are presented in the density estimation article. Welchs method Bartletts method Discrete-time Fourier transform Least-squares spectral analysis, for computing periodograms in data that is not equally spaced
21.
Analog-to-digital converter
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In electronics, an analog-to-digital converter is a system that converts an analog signal, such as a sound picked up by a microphone or light entering a digital camera, into a digital signal. Typically the digital output is a twos complement binary number that is proportional to the input, due to the complexity and the need for precisely matched components, all but the most specialized ADCs are implemented as integrated circuits. A digital-to-analog converter performs the function, it converts a digital signal into an analog signal. The conversion involves quantization of the input, so it necessarily introduces a small amount of error, furthermore, instead of continuously performing the conversion, an ADC does the conversion periodically, sampling the input. The result is a sequence of values that have been converted from a continuous-time and continuous-amplitude analog signal to a discrete-time. An ADC is defined by its bandwidth and its signal-to-noise ratio, the bandwidth of an ADC is characterized primarily by its sampling rate. The dynamic range of an ADC is influenced by many factors, including the resolution, linearity and accuracy, aliasing and jitter. The dynamic range of an ADC is often summarized in terms of its number of bits. An ideal ADC has an ENOB equal to its resolution, ADCs are chosen to match the bandwidth and required signal-to-noise ratio of the signal to be quantized. If an ADC operates at a rate greater than twice the bandwidth of the signal, then perfect reconstruction is possible given an ideal ADC. The presence of quantization error limits the range of even an ideal ADC. However, if the range of the ADC exceeds that of the input signal. The resolution of the converter indicates the number of values it can produce over the range of analog values. The resolution determines the magnitude of the error and therefore determines the maximum possible average signal to noise ratio for an ideal ADC without the use of oversampling. The values are stored electronically in binary form, so the resolution is usually expressed in bits. In consequence, the number of discrete values available, or levels, is assumed to be a power of two, for example, an ADC with a resolution of 8 bits can encode an analog input to one in 256 different levels, since 28 =256. The values can represent the ranges from 0 to 255 or from −128 to 127, resolution can also be defined electrically, and expressed in volts. The minimum change in required to guarantee a change in the output code level is called the least significant bit voltage
22.
Sampling (signal processing)
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In signal processing, sampling is the reduction of a continuous-time signal to a discrete-time signal. A common example is the conversion of a wave to a sequence of samples. A sample is a value or set of values at a point in time and/or space, a sampler is a subsystem or operation that extracts samples from a continuous signal. A theoretical ideal sampler produces samples equivalent to the value of the continuous signal at the desired points. Sampling can be done for functions varying in space, time, or any other dimension, then the sampled function is given by the sequence, s, for integer values of n. The sampling frequency or sampling rate, fs, is the number of samples obtained in one second. Reconstructing a continuous function from samples is done by interpolation algorithms, the Whittaker–Shannon interpolation formula is mathematically equivalent to an ideal lowpass filter whose input is a sequence of Dirac delta functions that are modulated by the sample values. When the time interval between adjacent samples is a constant, the sequence of functions is called a Dirac comb. Mathematically, the modulated Dirac comb is equivalent to the product of the function with s. That purely mathematical abstraction is sometimes referred to as impulse sampling, most sampled signals are not simply stored and reconstructed. But the fidelity of a reconstruction is a customary measure of the effectiveness of sampling. That fidelity is reduced when s contains frequency components whose periodicity is smaller than 2 samples, the quantity ½ cycles/sample × fs samples/sec = fs/2 cycles/sec is known as the Nyquist frequency of the sampler. Therefore, s is usually the output of a lowpass filter, without an anti-aliasing filter, frequencies higher than the Nyquist frequency will influence the samples in a way that is misinterpreted by the interpolation process. In practice, the signal is sampled using an analog-to-digital converter. This results in deviations from the theoretically perfect reconstruction, collectively referred to as distortion, various types of distortion can occur, including, Aliasing. Some amount of aliasing is inevitable because only theoretical, infinitely long, Aliasing can be made arbitrarily small by using a sufficiently large order of the anti-aliasing filter. Aperture error results from the fact that the sample is obtained as a time average within a sampling region, in a capacitor-based sample and hold circuit, aperture error is introduced because the capacitor cannot instantly change voltage thus requiring the sample to have non-zero width. Jitter or deviation from the precise sample timing intervals, noise, including thermal sensor noise, analog circuit noise, etc
23.
Duty-cycle
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A duty cycle is the fraction of one period in which a signal or system is active. Duty cycle is expressed as a percentage or a ratio. A period is the time it takes for a signal to complete an on-and-off cycle, thus, a 60% duty cycle means the signal is on 60% of the time but off 40% of the time. The on time for a 60% duty cycle could be a fraction of a second, the duty factor for periodic signal expresses the same notion, but is usually scaled to a maximum of one rather than 100%. In electronics, duty cycle is the percentage of the ratio of pulse duration and it is generally used to represent time duration of a pulse when it is high. In digital electronics, signals are used in rectangular waveform which are represented by logic 1, logic 1 stands for presence of an electric pulse and 0 for absence of an electric pulse. For example, a signal has 50% duty cycle, because the pulse remains high for 1/2 of the period or low for 1/2 of the period. Similarly, for pulse the duty cycle will be 25% because the pulse remains high only for 1/4 of the period, electrical motors typically use less than a 100% duty cycle. For example, if a motor runs for one out of 100 seconds, or 1/100 of the time, then, its duty cycle is 1/100, pulse-width modulation is used in a variety of electronic situations, such as power delivery and voltage regulation. In electronic music, music synthesizers vary the duty cycle of their audio-frequency oscillators to obtain an effect on the tone colors. This technique is known as pulse-width modulation, in the printer / copier industry, the duty cycle specification refers to the rated throughput of a device per month. In a welding power supply, the duty cycle is defined as the percentage of time in a 10-minute period that it can be operated continuously before overheating. The concept of duty cycles is used to describe the activity of neurons. In a biological neural network for example, a duty cycle specifically refers to the proportion of a period in which a neuron remains active. One way to fairly accurate square wave signals with 1/n duty factor
24.
Central processing unit
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The computer industry has used the term central processing unit at least since the early 1960s. The form, design and implementation of CPUs have changed over the course of their history, most modern CPUs are microprocessors, meaning they are contained on a single integrated circuit chip. An IC that contains a CPU may also contain memory, peripheral interfaces, some computers employ a multi-core processor, which is a single chip containing two or more CPUs called cores, in that context, one can speak of such single chips as sockets. Array processors or vector processors have multiple processors that operate in parallel, there also exists the concept of virtual CPUs which are an abstraction of dynamical aggregated computational resources. Early computers such as the ENIAC had to be rewired to perform different tasks. Since the term CPU is generally defined as a device for software execution, the idea of a stored-program computer was already present in the design of J. Presper Eckert and John William Mauchlys ENIAC, but was initially omitted so that it could be finished sooner. On June 30,1945, before ENIAC was made, mathematician John von Neumann distributed the paper entitled First Draft of a Report on the EDVAC and it was the outline of a stored-program computer that would eventually be completed in August 1949. EDVAC was designed to perform a number of instructions of various types. Significantly, the programs written for EDVAC were to be stored in high-speed computer memory rather than specified by the wiring of the computer. This overcame a severe limitation of ENIAC, which was the considerable time, with von Neumanns design, the program that EDVAC ran could be changed simply by changing the contents of the memory. Early CPUs were custom designs used as part of a larger, however, this method of designing custom CPUs for a particular application has largely given way to the development of multi-purpose processors produced in large quantities. This standardization began in the era of discrete transistor mainframes and minicomputers and has accelerated with the popularization of the integrated circuit. The IC has allowed increasingly complex CPUs to be designed and manufactured to tolerances on the order of nanometers, both the miniaturization and standardization of CPUs have increased the presence of digital devices in modern life far beyond the limited application of dedicated computing machines. Modern microprocessors appear in electronic devices ranging from automobiles to cellphones, the so-called Harvard architecture of the Harvard Mark I, which was completed before EDVAC, also utilized a stored-program design using punched paper tape rather than electronic memory. Relays and vacuum tubes were used as switching elements, a useful computer requires thousands or tens of thousands of switching devices. The overall speed of a system is dependent on the speed of the switches, tube computers like EDVAC tended to average eight hours between failures, whereas relay computers like the Harvard Mark I failed very rarely. In the end, tube-based CPUs became dominant because the significant speed advantages afforded generally outweighed the reliability problems, most of these early synchronous CPUs ran at low clock rates compared to modern microelectronic designs. Clock signal frequencies ranging from 100 kHz to 4 MHz were very common at this time, the design complexity of CPUs increased as various technologies facilitated building smaller and more reliable electronic devices
25.
Battery pack
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A battery pack is a set of any number of identical batteries or individual battery cells. They may be configured in a series, parallel or a mixture of both to deliver the desired voltage, capacity, or power density, the term battery pack is often used in reference to RC hobby toys and battery electric vehicles. Components of battery packs include the batteries or cells. Rechargeable battery packs contain a temperature sensor, which the battery charger uses to detect the end of charging. Interconnects are also found in batteries as they are the part which connects each cell, when a pack contains groups of cells in parallel there are differing wiring configurations which take into consideration the electrical balance of the circuit. Active balancing can also be performed by battery balancer devices which can shuttle energy from strong cells to weaker ones in time for better balance. A well-balanced pack lasts longer and delivers better performance, for an inline package, cells are selected and stacked with solder in between them. The cells are pressed together and a current pulse generates heat to them together. SOC or State of charge is the equivalent of a gauge for a battery. SOC cannot be determined by a voltage measurement, because the terminal voltage of a battery may stay substantially constant until it is completely discharged. More complex state of charge estimation systems take into account the Peukert effect which relates the capacity of the battery to the discharge rate, an advantage of a battery pack is the ease with which it can be swapped into or out of a device. This allows multiple packs to deliver extended runtimes, freeing up the device for continued use while charging the removed pack separately. Another advantage is the flexibility of their design and implementation, allowing the use of cheaper high-production cells or batteries to be combined into a pack for nearly any application. At the end of life, batteries can be removed and recycled separately. Packs are often simpler for end users to repair or tamper with than a sealed non-serviceable battery or cell. Though some might consider this an advantage it is important to safety precautions when servicing a battery pack as they pose a danger as potential chemical, electrical
26.
AC power
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Power in an electric circuit is the rate of flow of energy past a given point of the circuit. In alternating current circuits, energy storage such as inductors and capacitors may result in periodic reversals of the direction of energy flow. The portion of power that, averaged over a cycle of the AC waveform. The portion of power due to stored energy, which returns to the source in each cycle, is known as reactive power, in a simple alternating current circuit consisting of a source and a linear load, both the current and voltage are sinusoidal. If the load is resistive, the two quantities reverse their polarity at the same time. At every instant the product of voltage and current is positive or zero, in this case, only active power is transferred. If the load is purely reactive, then the voltage and current are 90 degrees out of phase, There is no net energy flow over each half cycle. In this case, only reactive power flows, There is no net transfer of energy to the load, however electrical power does flow along the wires and returns by flowing in reverse along the same wires. During its travels both from the source to the reactive load and back to the power source, this purely reactive power flow loses energy to the line resistance. Practical loads have resistance as well as inductance, and/or capacitance, Power engineers analyse the apparent power as being the magnitude of the vector sum of active and reactive power. Apparent power is the product of the rms values of voltage, conductors, transformers and generators must be sized to carry the total current, not just the current that does useful work. Another consequence is that adding the apparent power for two loads will not accurately give the power unless they have the same phase difference between current and voltage. Conventionally, capacitors are treated as if they generate reactive power, if a capacitor and an inductor are placed in parallel, then the currents flowing through the capacitor and the inductor tend to cancel rather than add. The result of this is that capacitive and inductive circuit elements tend to each other out. Current lagging voltage, current leading voltage and these are all denoted in the diagram to the right. In the diagram, P is the power, Q is the reactive power, S is the complex power. Reactive power does not do any work, so it is represented as the axis of the vector diagram. Active power does do work, so it is the real axis, the unit for all forms of power is the watt, but this unit is generally reserved for active power
27.
Frequency mixer
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In electronics, a mixer, or frequency mixer, is a nonlinear electrical circuit that creates new frequencies from two signals applied to it. Other frequency components may also be produced in a frequency mixer. Mixers are widely used to shift signals from one range to another. For example, a key component of a receiver is a mixer used to move received signals to a common intermediate frequency. Frequency mixers are used to modulate a carrier signal in radio transmitters. The essential characteristic of a mixer is that it produces a component in its output which is the product of the two input signals, a device that has a non-linear characteristic can act as a mixer. Passive mixers use one or more diodes and rely on the relation between voltage and current to provide the multiplying element. In a passive mixer, the output signal is always of lower power than the input signals. Active mixers use a device to increase the strength of the product signal. Active mixers improve isolation between the ports, but may have higher noise and more power consumption, an active mixer can be less tolerant of overload. Mixers may be built of components, may be part of integrated circuits. Mixers may also be classified by their topology, An unbalanced mixer, in addition to producing a product signal, allows both input signals to pass through and appear as components in the output. A single balanced mixer is arranged one of its inputs applied to a balanced circuit so that either the local oscillator or signal input is suppressed at the output. A double balanced mixer has both its inputs applied to differential circuits, so neither of the input signals and only the product signal appears at the output. Double balanced mixers are more complex and require higher levels than unbalanced. Selection of a type is a trade off for a particular application. Mixer circuits are characterized by their properties such as conversion gain, nonlinear electronic components that are used as mixers include diodes, transistors biased near cutoff, and at lower frequencies, analog multipliers. Ferromagnetic-core inductors driven into saturation have also been used, in nonlinear optics, crystals with nonlinear characteristics are used to mix two frequencies of laser light to create optical heterodynes
28.
Intermediate frequency
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In communications and electronic engineering, an intermediate frequency is a frequency to which a carrier wave is shifted as an intermediate step in transmission or reception. The intermediate frequency is created by mixing the signal with a local oscillator signal in a process called heterodyning, resulting in a signal at the difference or beat frequency. Intermediate frequencies are used in radio receivers, in which an incoming signal is shifted to an IF for amplification before final detection is done. Conversion to a frequency is useful for several reasons. When several stages of filters are used, they can all be set to a fixed frequency, lower frequency transistors generally have higher gains so fewer stages are required. Its easier to make sharply selective filters at lower fixed frequencies, there may be several such stages of intermediate frequency in a superheterodyne receiver, two or three stages are called double or triple conversion, respectively. Intermediate frequencies are used for three general reasons, at very high frequencies, signal processing circuitry performs poorly. Active devices such as transistors cannot deliver much amplification, ordinary circuits using capacitors and inductors must be replaced with cumbersome high frequency techniques such as striplines and waveguides. So a high frequency signal is converted to a lower IF for more convenient processing, bringing the signal in at the original microwave frequency would require an expensive waveguide. A second reason, in receivers that can be tuned to different frequencies, is to convert the various different frequencies of the stations to a frequency for processing. It is difficult to build multistage amplifiers, filters, and detectors that can have all stages track in tuning different frequencies, a more important advantage is that it gives the receiver a constant bandwidth over its tuning range. The bandwidth of a filter is proportional to its center frequency, in receivers like the TRF in which the filtering is done at the incoming RF frequency, as the receiver is tuned to higher frequencies its bandwidth increases. The main reason for using a frequency is to improve frequency selectivity. In communication circuits, a common task is to separate out or extract signals or components of a signal that are close together in frequency. Some examples are, picking up a station among several that are close in frequency. With all known filtering techniques the filters bandwidth increases proportionately with the frequency, so a narrower bandwidth and more selectivity can be achieved by converting the signal to a lower IF and performing the filtering at that frequency. Perhaps the most commonly used frequencies for broadcast receivers are around 455 kHz for AM receivers and 10.7 MHz for FM receivers. In special purpose receivers other frequencies can be used, a dual-conversion receiver may have two intermediate frequencies, a higher one to improve image rejection and a second, lower one, for desired selectivity
29.
Preamplifier
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Without this, the final signal would be noisy or distorted. They are typically used to amplify signals from analog sensors such as microphones, because of this, the preamplifier is often placed close to the sensor to reduce the effects of noise and interference. An ideal preamp will be linear, have input impedance. It is used to boost the strength to drive the cable to the main instrument without significantly degrading the signal-to-noise ratio. The noise performance of a preamplifier is critical, according to Friiss formula, when the gain of the preamplifier is high, the SNR of the final signal is determined by the SNR of the input signal and the noise figure of the preamplifier. Three basic types of preamplifiers are available, current-sensitive preamplifier parasitic-capacitance preamplifier charge-sensitive preamplifier, in an audio system, they are typically used to amplify signals from analog sensors to line level. The second amplifier is typically a power amplifier, the preamplifier provides voltage gain but no significant current gain. The power amplifier provides the current necessary to drive loudspeakers. For these systems some common sensors are microphones, instrument pickups, preamplifiers are often integrated into the audio inputs on mixing consoles, DJ mixers, and sound cards. They can also be stand-alone devices, the integrated pre-amplifier in a foil electret microphone. The first stages of an instrument amplifier, which is sent to the power amplifier. With instrument amplifiers, the preamp is often designed to produce overdrive or distortion effects, a stand-alone unit for use in live music and recording studio applications. As part of a stand-alone channel strip or channel strip built into a mixing desk. A masthead amplifier used with television receiver antenna or a satellite receiver dish, the circuit inside of a hard drive connected to the magnetic heads or the circuit inside of CD/DVD drive which connects to the photodiodes
30.
Intermodulation
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Intermodulation or intermodulation distortion is the amplitude modulation of signals containing two or more different frequencies, caused by nonlinearities in a system. Intermodulation is caused by non-linear behaviour of the processing being used. Practically all audio equipment has some non-linearity, so it will exhibit some amount of IMD, due to the characteristics of the human auditory system, the same percentage of IMD is perceived as more bothersome when compared to the same amount of harmonic distortion. Intermodulation is also desirable in radio, as it creates unwanted spurious emissions. For radio transmissions this increases the bandwidth, leading to adjacent channel interference. IMD is only distinct from harmonic distortion in that the signal is different. The same nonlinear system will produce both THD and IMD, IMD is also distinct from intentional modulation where signals to be modulated are presented to an intentional nonlinear element. See non-linear mixers such as diodes and even single-transistor oscillator-mixer circuits. A linear system cannot produce intermodulation, if the input of a linear time-invariant system is a signal of a single frequency, then the output is a signal of the same frequency, only the amplitude and phase can differ from the input signal. Intermodulation occurs when the input to a system is composed of two or more frequencies. As explained in a section, intermodulation can only occur in non-linear systems. Non-linear systems are composed of active components, meaning that the components must be biased with an external power source which is not the input signal. Passive intermodulation, however, occurs in passive devices that are subjected to two or more high power tones, the PIM product is the result of the two high power tones mixing at device nonlinearities such as junctions of dissimilar metals or metal-oxide junctions, such as loose corroded connectors. It is also possible for a single broadband carrier to generate PIM if it passes through a PIM generating surface or defect and these distortions would show up as side lobes in a telecommunication signal and interfere with adjacent channels and impede reception. PIM can be severe problem in communication systems. Paths that share both high power transmit and the signal are most susceptible to this kind of interference. Once PIM interference finds its way to receive path, it can not be filtered or separated, ferromagnetic materials are the most common materials to avoid and include ferrites, nickel, and steels. These materials exhibit hysteresis when exposed to reversing magnetic fields, resulting in PIM generation, PIM can also be generated in components with manufacturing or workmanship defects, such as cold or cracked solder joints or poorly made mechanical contacts
31.
Nyquist rate
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In signal processing, the Nyquist rate, named after Harry Nyquist, is twice the bandwidth of a bandlimited function or a bandlimited channel. When a continuous function, x, is sampled at a constant rate, fs samples/second, but only one of them is bandlimited to ½ fs cycles/second, which means that its Fourier transform, X, is 0 for all |f| ≥ ½ fs. The mathematical algorithms that are used to recreate a continuous function from samples create arbitrarily good approximations to this theoretical. It follows that if the function, x, is bandlimited to ½ fs. In terms of a functions own bandwidth, as depicted above, and 2B is called the Nyquist rate for functions with bandwidth B. When the Nyquist criterion is not met, a condition called aliasing occurs, in most cases, the differences are viewed as distortion. Figure 2 depicts a type of function called baseband or lowpass, because its range of significant energy is [0. When instead, the range is, for some A > B, it is called bandpass. One way to do that is frequency-mixing the bandpass function down to the frequency range, one of the possible reasons is to reduce the Nyquist rate for more efficient storage. For a more general discussion, see bandpass sampling, long before Harry Nyquist had his name associated with sampling, the term Nyquist rate was used differently, with a meaning closer to what Nyquist actually studied. This rate is referred to as signaling at the Nyquist rate. According to the OED, Blacks statement regarding 2B may be the origin of the term Nyquist rate, Nyquists famous 1928 paper was a study on how many pulses could be transmitted per second, and recovered, through a channel of limited bandwidth. Signaling at the Nyquist rate meant putting as many code pulses through a channel as its bandwidth would allow. Shannon used Nyquists approach when he proved the theorem in 1948
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Spectrogram
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A spectrogram is a visual representation of the spectrum of frequencies in a sound or other signal as they vary with time or some other variable. Spectrograms are sometimes called spectral waterfalls, voiceprints, or voicegrams, spectrograms can be used to identify spoken words phonetically, and to analyse the various calls of animals. They are used extensively in the development of the fields of music, sonar, radar, the frequency and amplitude axes can be either linear or logarithmic, depending on what the graph is being used for. Audio would usually be represented with a logarithmic amplitude axis, and frequency would be linear to emphasize harmonic relationships, or logarithmic to emphasize musical, tonal relationships. Spectrograms are usually created in one of two ways, approximated as a filterbank that results from a series of filters, or calculated from the time signal using the Fourier transform. These two methods actually form two different time–frequency representations, but are equivalent under some conditions, creating a spectrogram using the FFT is a digital process. Digitally sampled data, in the domain, is broken up into chunks, which usually overlap. Each chunk then corresponds to a line in the image. These spectrums or time plots are laid side by side to form the image or a three-dimensional surface, or slightly overlapped in various ways. Early analog spectrograms were applied to a range of areas including the study of bird calls, with current research continuing using modern digital equipment. Contemporary use of the spectrogram is especially useful for studying frequency modulation in animal calls. Specifically, the characteristics of FM chirps, broadband clicks. By reversing the process of producing a spectrogram, it is possible to create a signal whose spectrogram is an arbitrary image and this technique can be used to hide a picture in a piece of audio and has been employed by several electronic music artists. See Audio timescale-pitch modification and Phase vocoder, spectrograms can be used to analyze the results of passing a test signal through a signal processor such as a filter in order to check its performance. The Analysis & Resynthesis Sound Spectrograph is an example of a program that attempts to do this. The Pattern Playback was a speech synthesizer, designed at Haskins Laboratories in the late 1940s. In fact, there is some information in the spectrogram. The size and shape of the window can be varied
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Digital filter
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In signal processing, a digital filter is a system that performs mathematical operations on a sampled, discrete-time signal to reduce or enhance certain aspects of that signal. This is in contrast to the major type of electronic filter, the analog filter. Finally a digital-to-analog converter to complete the output stage, program Instructions running on the microprocessor implement the digital filter by performing the necessary mathematical operations on the numbers received from the ADC. Digital filters may be more expensive than an equivalent analog filter due to their increased complexity, Digital filters are commonplace and an essential element of everyday electronics such as radios, cellphones, and AV receivers. A digital filter is characterized by its function, or equivalently. Mathematical analysis of the function can describe how it will respond to any input. As such, designing a filter consists of developing specifications appropriate to the problem, see Z-transforms LCCD equation for further discussion of this transfer function. A variety of techniques may be employed to analyze the behaviour of a given digital filter. Many of these techniques may also be employed in designs. Typically, one characterizes filters by calculating how they respond to a simple input such as an impulse. One can then extend this information to compute the filters response to more complex signals, the impulse response, often denoted h or h k, is a measurement of how a filter will respond to the Kronecker delta function. For example, given an equation, one would set x 0 =1 and x k =0 for k ≠0. The impulse response is a characterization of the filters behaviour, Digital filters are typically considered in two categories, infinite impulse response and finite impulse response. In discrete-time systems, the filter is often implemented by converting the transfer function to a linear constant-coefficient difference equation via the Z-transform. The discrete frequency-domain transfer function is written as the ratio of two polynomials, applying the filter to an input in this form is equivalent to a Direct Form I or II realization, depending on the exact order of evaluation. The design of filters is a deceptively complex topic. Although filters are easily understood and calculated, the challenges of their design. There are two categories of digital filter, the filter and the nonrecursive filter
34.
Vector signal analyzer
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A vector signal analyzer is an instrument that measures the magnitude and phase of the input signal at a single frequency within the IF bandwidth of the instrument. The primary use is to make measurements, such as error vector magnitude, code domain power. Vector signal analyzers are useful in measuring and demodulating digitally modulated signals like W-CDMA, LTE and these measurements are used to determine the quality of modulation and can be used for design validation and compliance testing of electronic devices. The vector signal analyzer spectrum analysis process typically has a down-convert & digitizing stage, a vector signal analyzer operates by first down converting the signal spectra by using superheterodyne techniques. A portion of the signal spectrum is down-converted to the center frequency of a band-pass filter. The use of a voltage-controlled oscillator allows for consideration of different carrier frequencies, after the conversion to an intermediate frequency, the signal is filtered in order to band-limit the signal and prevent aliasing. The signal is then digitized using an analog-to-digital converter, sampling rate is often varied in relation to the frequency span under consideration. Once the signal is digitized, it is separated into quadrature and in-phase components using a quadrature detector, usually there is a windowing function option to limit spectral leakage and enhance frequency resolution. This window is implemented by multiplying it with the values of the sample period before computing the FFT. Constellation Diagram A constellation diagram represents a signal modulated by a modulation scheme such as quadrature amplitude modulation or phase-shift keying. This diagram maps the magnitude of the quadrature and in-phase components to the vertical and horizontal directions, qualitative assessments of signal integrity can be made based on interpretation of this diagram. This requires knowledge of the signal in order to compare the received signal with the ideal signal. Many more measurement results can be displayed depending on the type of modulation being used, communications Systems Analysis Using Hardware and Software-Based Vector Signal Analyzers
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Envelope detector
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An envelope detector is an electronic circuit that takes a high-frequency signal as input and provides an output which is the envelope of the original signal. The capacitor in the stores up charge on the rising edge. The diode in series rectifies the incoming signal, allowing current flow only when the input terminal is at a higher potential than the negative input terminal. Most practical envelope detectors use either half-wave or full-wave rectification of the signal to convert the AC audio input into a pulsed DC signal, filtering is then used to smooth the final result. This filtering is rarely perfect and some ripple is likely to remain on the envelope follower output, more filtering gives a smoother result, but decreases the responsiveness, thus, real-world designs must be optimized for the application. Any AM or FM signal x can be written in the form x = R cos In the case of AM, φ is constant. In AM, the carrier frequency ω is also constant, thus, all the information in the AM signal is in R. R is called the envelope of the signal. Hence an AM signal is given by the function x = cos with m representing the original audio frequency message, C the carrier amplitude, so, if the envelope of the AM signal can be extracted, the original message can be recovered. In the case of FM, the signal x has a constant envelope R = R. However, many FM receivers measure the envelope anyway for received signal strength indication, the simplest form of envelope detector is the diode detector which is shown above. A diode detector is simply a diode between the input and output of a circuit, connected to a resistor and capacitor in parallel from the output of the circuit to the ground. If the resistor and capacitor are correctly chosen, the output of this circuit should approximate a voltage-shifted version of the original signal, a simple filter can then be applied to filter out the DC component. An envelope detector can also be constructed to use a precision rectifier feeding into a low-pass filter, the envelope detector has several drawbacks, The input to the detector must be band-pass filtered around the desired signal, or else the detector will simultaneously demodulate several signals. An envelope detector can be used to demodulate a previously modulated signal by removing all high frequency components of the signal, the capacitor and resistor form a low-pass filter to filter out the carrier frequency. Such a device is used to demodulate AM radio signals because the envelope of the modulated signal is equivalent to the baseband signal. An envelope detector is sometimes referred to as a follower in musical environments. It is still used to detect the amplitude variations of a signal to produce a control signal that resembles those variations. However, in case the input signal is made up of audible frequencies
36.
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
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Root mean square
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In statistics and its applications, the root mean square is defined as the square root of mean square. The RMS is also known as the mean and is a particular case of the generalized mean with exponent 2. RMS can also be defined for a continuously varying function in terms of an integral of the squares of the values during a cycle. For a cyclically alternating electric current, RMS is equal to the value of the current that would produce the same average power dissipation in a resistive load. In Estimation theory the mean square error of an estimator is a measure of the imperfection of the fit of the estimator to the data. The RMS value of a set of values is the root of the arithmetic mean of the squares of the values. In Physics, the RMS current is the value of the current that dissipates power in a resistor. In the case of a set of n values, the RMS x r m s =1 n, the RMS over all time of a periodic function is equal to the RMS of one period of the function. The RMS value of a function or signal can be approximated by taking the RMS of a sequence of equally spaced samples. Additionally, the RMS value of various waveforms can also be determined without calculus, in the case of the RMS statistic of a random process, the expected value is used instead of the mean. If the waveform is a sine wave, the relationships between amplitudes and RMS are fixed and known, as they are for any continuous periodic wave. However, this is not true for a waveform which may or may not be periodic or continuous. For a zero-mean sine wave, the relationship between RMS and peak-to-peak amplitude is, Peak-to-peak =22 × R M S ≈2.8 × R M S, for other waveforms the relationships are not the same as they are for sine waves. Waveforms made by summing known simple waveforms have an RMS that is the root of the sum of squares of the component RMS values, if the component waveforms are orthogonal. R M S Total = R M S12 + R M S22 + ⋯ + R M S n 2 A special case of this, another special case, useful in statistics, is given in #Relationship to other statistics. Electrical engineers often need to know the power, P, dissipated by an electrical resistance and it is easy to do the calculation when there is a constant current, I, through the resistance. Average power can also be using the same method that in the case of a time-varying voltage, V, with RMS value VRMS. This equation can be used for any waveform, such as a sinusoidal or sawtooth waveform
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Frequency response
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Frequency response is the quantitative measure of the output spectrum of a system or device in response to a stimulus, and is used to characterize the dynamics of the system. It is a measure of magnitude and phase of the output as a function of frequency, also for a linear system, doubling the amplitude of the input will double the amplitude of the output. In addition, if the system is time-invariant, then the frequency response also will not vary with time, thus for LTI systems, the frequency response can be seen as applying the systems transfer function to a purely imaginary number argument representing the frequency of the sinusoidal excitation. Two applications of frequency response analysis are related but have different objectives, for an audio system, the objective may be to reproduce the input signal with no distortion. That would require a uniform magnitude of response up to the limitation of the system. That amount of time could be seconds, or weeks or months in the case of recorded media, in contrast, for a feedback apparatus used to control a dynamic system, the objective is to give the closed-loop system improved response as compared to the uncompensated system. The feedback generally needs to respond to system dynamics within a small number of cycles of oscillation. For feedback of sufficient amplification, getting the phase angle wrong can lead to instability for a stable system. For audio systems with nearly uniform time delay at all frequencies, if the system under investigation is nonlinear then applying purely linear frequency domain analysis will not reveal all the nonlinear characteristics. In electronics this stimulus would be an input signal, in the audible range it is usually referred to in connection with electronic amplifiers, microphones and loudspeakers. Radio spectrum frequency response can refer to measurements of coaxial cable, twisted-pair cable, video switching equipment, wireless communications devices, infrasonic frequency response measurements include earthquakes and electroencephalography. Frequency response requirements differ depending on the application, frequency response curves are often used to indicate the accuracy of electronic components or systems. Once a frequency response has been measured, provided the system is linear and time-invariant, similarly, if a system is demonstrated to have a poor frequency response, a digital or analog filter can be applied to the signals prior to their reproduction to compensate for these deficiencies. Impulse response Bode plot Bandwidth Audio system measurements Transient response & steady-state response Spectral sensitivity Notes Bibliography Luther, live Sound Reinforcement, Vallejo, California, Artistpro. com, 1996–2002
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Noise (electronics)
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In electronics, noise is a random fluctuation in an electrical signal, a characteristic of all electronic circuits. Noise generated by electronic devices varies greatly as it is produced by different effects. In communication systems, noise is an error or undesired random disturbance of an information signal. The noise is a summation of unwanted or disturbing energy from natural, Noise is, however, typically distinguished from interference, for example in the signal-to-noise ratio, signal-to-interference ratio and signal-to-noise plus interference ratio measures. While noise is generally unwanted, it can serve a purpose in some applications. Johnson–Nyquist noise is unavoidable, and generated by the thermal motion of charge carriers, inside an electrical conductor. Thermal noise is white, meaning that its power spectral density is nearly equal throughout the frequency spectrum. The amplitude of the signal has very nearly a Gaussian probability density function, a communication system affected by thermal noise is often modeled as an additive white Gaussian noise channel. If electrons flow across a barrier, then they have discrete arrival times and those discrete arrivals exhibit shot noise. The output of a noise generator is easily set by the current. Typically, the barrier in a diode is used, shot noise in electronic devices results from unavoidable random statistical fluctuations of the electric current when the charge carriers traverse a gap. The current is a flow of charges, and the fluctuation in the arrivals of those charges creates shot noise. Shot noise is similar to the created by rain falling on a tin roof. The flow of rain may be constant, but the raindrops arrive discretely. The shot noise assumes independent arrivals, vacuum tubes have shot noise because the electrons randomly leave the cathode and arrive at the anode. A tube may not exhibit the full shot noise effect, the presence of a space charge tends to smooth out the arrival times. Conductors and resistors typically do not exhibit shot noise because the electrons thermalize and move diffusively within the material, shot noise has been demonstrated in mesoscopic resistors when the size of the resistive element becomes shorter than the electron-phonon scattering length. Flicker noise, also known as 1/f noise, is a signal or process with a spectrum that falls off steadily into the higher frequencies
40.
Telecommunication
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Telecommunication is the transmission of signs, signals, messages, writings, images and sounds or intelligence of any nature by wire, radio, optical or other electromagnetic systems. Telecommunication occurs when the exchange of information between communication participants includes the use of technology and it is transmitted either electrically over physical media, such as cables, or via electromagnetic radiation. Such transmission paths are divided into communication channels which afford the advantages of multiplexing. The term is used in its plural form, telecommunications. Early means of communicating over a distance included visual signals, such as beacons, smoke signals, semaphore telegraphs, signal flags, other examples of pre-modern long-distance communication included audio messages such as coded drumbeats, lung-blown horns, and loud whistles. Zworykin, John Logie Baird and Philo Farnsworth, the word telecommunication is a compound of the Greek prefix tele, meaning distant, far off, or afar, and the Latin communicare, meaning to share. Its modern use is adapted from the French, because its use was recorded in 1904 by the French engineer. Communication was first used as an English word in the late 14th century, in the Middle Ages, chains of beacons were commonly used on hilltops as a means of relaying a signal. Beacon chains suffered the drawback that they could pass a single bit of information. One notable instance of their use was during the Spanish Armada, in 1792, Claude Chappe, a French engineer, built the first fixed visual telegraphy system between Lille and Paris. However semaphore suffered from the need for skilled operators and expensive towers at intervals of ten to thirty kilometres, as a result of competition from the electrical telegraph, the last commercial line was abandoned in 1880. Homing pigeons have occasionally used throughout history by different cultures. Pigeon post is thought to have Persians roots and was used by the Romans to aid their military, frontinus said that Julius Caesar used pigeons as messengers in his conquest of Gaul. The Greeks also conveyed the names of the victors at the Olympic Games to various cities using homing pigeons, in the early 19th century, the Dutch government used the system in Java and Sumatra. And in 1849, Paul Julius Reuter started a service to fly stock prices between Aachen and Brussels, a service that operated for a year until the gap in the telegraph link was closed. Sir Charles Wheatstone and Sir William Fothergill Cooke invented the telegraph in 1837. Also, the first commercial electrical telegraph is purported to have constructed by Wheatstone and Cooke. Both inventors viewed their device as an improvement to the electromagnetic telegraph not as a new device, samuel Morse independently developed a version of the electrical telegraph that he unsuccessfully demonstrated on 2 September 1837
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Electromagnetic compatibility
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The goal of EMC is the correct operation of different equipment in a common electromagnetic environment. EMC pursues two main classes of issue, emission is the generation of electromagnetic energy, whether deliberate or accidental, by some source and its release into the environment. EMC studies the unwanted emissions and the countermeasures which may be taken in order to reduce unwanted emissions, a third class studied is coupling, which is the mechanism by which emitted interference reaches the victim. In practice, many of the techniques used, such as grounding and shielding. Another way of saying this is that EMC is the control of EMI so that effects are prevented. Electromagnetic interference divides into several categories according to the source and signal characteristics, the origin of interference, often called noise in this context, can be man-made or natural. Continuous, or Continuous Wave, interference arises where the source continuously emits at a range of frequencies. This type is divided into sub-categories according to frequency range. Audio Frequency, from low frequencies up to around 20 kHz. Frequencies up to 100 kHz may sometimes be classified as Audio, Sources include, Mains hum from, power supply units, nearby power supply wiring, transmission lines and substations. Audio processing equipment, such as power amplifiers and loudspeakers. Demodulation of a carrier wave such as an FM radio transmission. Radio Frequency Interference, from typically 20 kHz to an upper limit which constantly increases as technology pushes it higher, the energy is usually broadband by nature, although it often excites a relatively narrow-band damped sine wave response in the victim. Sources divide broadly into isolated and repetitive events, Sources of isolated EMP events include, Switching action of electrical circuitry, including inductive loads such as relays, solenoids, or electric motors. Electrostatic discharge, as a result of two charged objects coming into close proximity or contact, lightning electromagnetic pulse, although typically a short series of pulses. Nuclear electromagnetic pulse, as a result of a nuclear explosion, a variant of this is the high altitude EMP nuclear device, designed to create the pulse as its primary destructive effect. Power line surges/pulses Sources of repetitive EMP events, sometimes as regular pulse trains, include, Electric motors Electrical ignition systems, continual switching actions of digital electronic circuitry. Some of the technical words employed can be used with differing meanings and these terms are used here in a widely accepted way, which is consistent with other articles in the encyclopedia
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