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Transcript
Introduction to Audio This beginner-level tutorial covers the basics of audio production. It is suitable for anyone wanting to learn more about working with sound, in either amateur or professional situations. The tutorial is five pages and takes about 20 minutes to complete. What is "Audio"? Audio means "of sound" or "of the reproduction of sound". Specifically, it refers to the range of frequencies detectable by the human ear — approximately 20Hz to 20kHz. It's not a bad idea to memorise those numbers — 20Hz is the lowest-pitched (bassiest) sound we can hear, 20kHz is the highest pitch we can hear. Audio work involves the production, recording, manipulation and reproduction of sound waves. To understand audio you must have a grasp of two things: 1. Sound Waves: What they are, how they are produced and how we hear them. 2. Sound Equipment: What the different components are, what they do, how to choose the correct equipment and use it properly. Fortunately it's not particularly difficult. Audio theory is simpler than video theory and once you understand the basic path from the sound source through the sound equipment to the ear, it all starts to make sense. Technical note: In physics, sound is a form of energy known as acoustical energy. The Field of Audio Work The field of audio is vast, with many areas of specialty. Hobbyists use audio for all sorts of things, and audio professionals can be found in a huge range of vocations. Some common areas of audio work include: Studio Sound Engineer Live Sound Engineer Musician Music Producer DJ Radio technician Film/Television Sound Recordist Field Sound Engineer Audio Editor Post-Production Audio Creator In addition, many other professions require a level of audio proficiency. For example, video camera operators should know enough about audio to be able to record good quality sound with their pictures. Speaking of video-making, it's important to recognise the importance of audio in film and video. A common mistake amongst amateurs is to concentrate only on the vision and assume that as long as the microphone is working the audio will be fine. However, satisfactory audio requires skill and effort. Sound is critical to the flow of the programme — indeed in many situations high quality sound is more important than high quality video. Most jobs in audio production require some sort of specialist skill set, whether it be micing up a drum kit or creating synthetic sound effects. Before you get too carried away with learning specific tasks, you should make sure you have a general grounding in the principles of sound. Once you have done this homework you will be well placed to begin specialising. The first thing to tackle is basic sound wave theory... Acoustics Acoustics is the interdisciplinary science that deals with the study of all mechanical waves in gases, liquids, and solids including vibration, sound, ultrasound and infrasound. A scientist who works in the field of acoustics is an acoustician while someone working in the field of acoustics technology may be called an acoustical engineer. The application of acoustics can be seen in almost all aspects of modern society with the most obvious being the audio and noise control industries. Hearing is one of the most crucial means of survival in the animal world, and speech is one of the most distinctive characteristics of human development and culture. So it is no surprise that the science of acoustics spreads across so many facets of our society—music, medicine, architecture, industrial production, warfare and more. Art, craft, science and technology have provoked one another to advance the whole, as in many other fields of knowledge. Lindsay's 'Wheel of Acoustics' is a well accepted overview of the various fields in acoustics.[1] The word "acoustic" is derived from the Greek word ἀκουστικός (akoustikos), meaning "of or for hearing, ready to hear"[2] and that from ἀκουστός (akoustos), "heard, audible",[3] which in turn derives from the verb ἀκούω (akouo), "I hear".[4] The Latin synonym is "sonic", after which the term sonics used to be a synonym for acoustics[5] and later a branch of acoustics.[6] After acousticians had extended their studies to frequencies above and below the audible range, it became conventional to identify these frequency ranges as "ultrasonic" and "infrasonic" respectively, while letting the word "acoustic" refer to the entire frequency range without limit . Nature of Sound Waves Sound is one kind of longitudinal wave, in which the particles oscillate to and fro in the same direction of wave propagation. Sound waves cannot be transmitted through vacuum. The transmission of sound requires at least a medium, which can be solid, liquid, or gas. condensation rarefaction The wavelength, l is the distance between two successive rarefactions or condensations. Figure 1 Propagation of Sound Wave The displacement of any point on the wave, y, along the direction of propagation is related to time by the following formula: (1) (2) (3) (4) Table 1 shows the velocities of sound in same common media. Material Air Velocity of Sound (m/s) 344 Water 1,372 Concrete 3,048 Glass 3,658 Iron 5,182 Lead 1,219 Steel 5,182 Wood (hard) 4,267 Wood (soft) 3,353 Table 1 Approximate Velocities of Sound in Some Common Media Fundamental characteristics of sound Sound is a mechanical wave that is an oscillation of pressure transmitted through a solid, liquid, or gas, composed of frequencies within the range of hearing and of a level sufficiently strong to be heard, or the sensation stimulated in organs of hearing by such vibrations Propagation of sound Sound is a sequence of waves of pressure that propagates through compressible media such as air or water. (Sound can propagate through solids as well, but there are additional modes of propagation). During propagation, waves can be reflected, refracted, or attenuated by the medium.[2] The behavior of sound propagation is generally affected by three things: A relationship between density and pressure. This relationship, affected by temperature, determines the speed of sound within the medium. The propagation is also affected by the motion of the medium itself. For example, sound moving through wind. Independent of the motion of sound through the medium, if the medium is moving, the sound is further transported. The viscosity of the medium also affects the motion of sound waves. It determines the rate at which sound is attenuated. For many media, such as air or water, attenuation due to viscosity is negligible. When sound is moving through a medium that does not have constant physical properties, it may be refracted (either dispersed or focused).[2] Perception of sound Human ear The perception of sound in any organism is limited to a certain range of frequencies. For humans, hearing is normally limited to frequencies between about 20 Hz and 20,000 Hz (20 kHz)[3], although these limits are not definite. The upper limit generally decreases with age. Other species have a different range of hearing. For example, dogs can perceive vibrations higher than 20 kHz, but are deaf to anything below 40 Hz. As a signal perceived by one of the major senses, sound is used by many species for detecting danger, navigation, predation, and communication. Earth's atmosphere, water, and virtually any physical phenomenon, such as fire, rain, wind, surf, or earthquake, produces (and is characterized by) its unique sounds. Many species, such as frogs, birds, marine and terrestrial mammals, have also developed special organs to produce sound. In some species, these produce song and speech. Furthermore, humans have developed culture and technology (such as music, telephone and radio) that allows them to generate, record, transmit, and broadcast sound. The scientific study of human sound perception is known as psychoacoustics. Physics of sound The mechanical vibrations that can be interpreted as sound are able to travel through all forms of matter: gases, liquids, solids, and plasmas. The matter that supports the sound is called the medium. Sound cannot travel through a vacuum. Longitudinal and transverse waves Sinusoidal waves of various frequencies; the bottom waves have higher frequencies than those above. The horizontal axis represents time. Sound is transmitted through gases, plasma, and liquids as longitudinal waves, also called compression waves. Through solids, however, it can be transmitted as both longitudinal waves and transverse waves. Longitudinal sound waves are waves of alternating pressure deviations from the equilibrium pressure, causing local regions of compression and rarefaction, while transverse waves (in solids) are waves of alternating shear stress at right angle to the direction of propagation. Matter in the medium is periodically displaced by a sound wave, and thus oscillates. The energy carried by the sound wave converts back and forth between the potential energy of the extra compression (in case of longitudinal waves) or lateral displacement strain (in case of transverse waves) of the matter and the kinetic energy of the oscillations of the medium. Sound wave properties and characteristics Sound waves are often simplified to a description in terms of sinusoidal plane waves, which are characterized by these generic properties: Frequency, or its inverse, the period Wavelength Wavenumber Amplitude Sound pressure Sound intensity Speed of sound Direction Sometimes speed and direction are combined as a velocity vector; wavenumber and direction are combined as a wave vector. Transverse waves, also known as shear waves, have the additional property, polarization, and are not a characteristic of sound waves. Speed of sound U.S. Navy F/A-18 breaking the sound barrier. The white halo is formed by condensed water droplets thought to result from a drop in air pressure around the aircraft (see Prandtl-Glauert Singularity).[4][5] Main article: Speed of sound The speed of sound depends on the medium the waves pass through, and is a fundamental property of the material. In general, the speed of sound is proportional to the square root of the ratio of the elastic modulus (stiffness) of the medium to its density. Those physical properties and the speed of sound change with ambient conditions. For example, the speed of sound in gases depends on temperature. In 20 °C (68 °F) air at the sea level, the speed of sound is approximately 343 m/s (1,230 km/h; 767 mph) using the formula "v = (331 + 0.6 T) m/s". In fresh water, also at 20 °C, the speed of sound is approximately 1,482 m/s (5,335 km/h; 3,315 mph). In steel, the speed of sound is about 5,960 m/s (21,460 km/h; 13,330 mph).[6] The speed of sound is also slightly sensitive (a second-order anharmonic effect) to the sound amplitude, which means that there are nonlinear propagation effects, such as the production of harmonics and mixed tones not present in the original sound (see parametric array). Acoustics Main article: Acoustics Acoustics is the interdisciplinary science that deals with the study of all mechanical waves in gases, liquids, and solids including vibration, sound, ultrasound and infrasound. A scientist who works in the field of acoustics is an acoustician while someone working in the field of acoustics technology may be called an acoustical or audio engineer. The application of acoustics can be seen in almost all aspects of modern society with the most obvious being the audio and noise control industries. Noise Main article: Noise Noise is a term often used to refer to an unwanted sound. In science and engineering, noise is an undesirable component that obscures a wanted signal. Sound pressure level Main article: Sound pressure Sound pressure is the difference, in a given medium, between average local pressure and the pressure in the sound wave. A square of this difference (i.e., a square of the deviation from the equilibrium pressure) is usually averaged over time and/or space, and a square root of this average provides a root mean square (RMS) value. For example, 1 Pa RMS sound pressure (94 dBSPL) in atmospheric air implies that the actual pressure in the sound wave oscillates between (1 atm Pa) and (1 atm Pa), that is between 101323.6 and 101326.4 Pa. Such a tiny (relative to atmospheric) variation in air pressure at an audio frequency is perceived as a deafening sound, and can cause hearing damage, according to the table below. As the human ear can detect sounds with a wide range of amplitudes, sound pressure is often measured as a level on a logarithmic decibel scale. The sound pressure level (SPL) or Lp is defined as Sound measurements Sound pressure p, SPL Particle velocity v, SVL Particle displacement ξ Sound intensity I, SIL Sound power Pac Sound power level SWL where p is the root-mean-square sound pressure and pref is a reference sound pressure. Commonly used reference sound pressures, defined in the standard ANSI S1.1-1994, are 20 µPa in air and 1 µPa in water. Without a specified reference sound pressure, a value expressed in decibels cannot represent a sound pressure level. Sound energy Since the human ear does not have a flat spectral response, sound pressures are often frequency weighted so that the measured level matches perceived levels more closely. The International Electrotechnical Commission (IEC) has defined several weighting schemes. A-weighting attempts to match the response of the human ear to noise and Aweighted sound pressure levels are labeled dBA. Cweighting is used to measure peak levels. Acoustic impedance Z Sound energy density E Sound energy flux q Speed of sound c Audio frequency AF v·d·e Equipment for dealing with sound Equipment for generating or using sound includes musical instruments, hearing aids, sonar systems and sound reproduction and broadcasting equipment. Many of these use electro-acoustic transducers such as microphones and loudspeakers. Sound measurement Decibel, Sone, mel, Phon, Hertz Sound pressure level, Sound pressure Particle velocity, Acoustic velocity Particle displacement, Particle amplitude, Particle acceleration Sound power, Acoustic power, Sound power level Sound energy flux Sound intensity, Acoustic intensity, Sound intensity level Acoustic impedance, Sound impedance, Characteristic impedance Speed of sound, Amplitude Microphone A microphone (colloquially called a mic or mike; both pronounced /ˈmaɪk/[1]) is an acoustic-toelectric transducer or sensor that converts sound into an electrical signal. In 1877, Emile Berliner invented the first microphone used as a telephone voice transmitter.[2] Microphones are used in many applications such as telephones, tape recorders, karaoke systems, hearing aids, motion picture production, live and recorded audio engineering, FRS radios, megaphones, in radio and television broadcasting and in computers for recording voice, speech recognition, VoIP, and for non-acoustic purposes such as ultrasonic checking or knock sensors. Most microphones today use electromagnetic induction (dynamic microphone), capacitance change (condenser microphone), piezoelectric generation, or light modulation to produce an electrical voltage signal from mechanical vibration. Components The sensitive transducer element of a microphone is called its element or capsule. A complete microphone also includes a housing, some means of bringing the signal from the element to other equipment, and often an electronic circuit to adapt the output of the capsule to the equipment being driven. A wireless microphone contains a radio transmitter. Varieties Microphones are referred to by their transducer principle, such as condenser, dynamic, etc., and by their directional characteristics. Sometimes other characteristics such as diaphragm size, intended use or orientation of the principal sound input to the principal axis (end- or sideaddress) of the microphone are used to describe the microphone. Condenser microphone Inside the Oktava 319 condenser microphone The condenser microphone, invented at Bell Labs in 1916 by E. C. Wente[3] is also called a capacitor microphone or electrostatic microphone—capacitors were historically called condensers. Here, the diaphragm acts as one plate of a capacitor, and the vibrations produce changes in the distance between the plates. There are two types, depending on the method of extracting the audio signal from the transducer: DC-biased and radio frequency (RF) or high frequency (HF) condenser microphones. With a DC-biased microphone, the plates are biased with a fixed charge (Q). The voltage maintained across the capacitor plates changes with the vibrations in the air, according to the capacitance equation (C = Q⁄V), where Q = charge in coulombs, C = capacitance in farads and V = potential difference in volts. The capacitance of the plates is inversely proportional to the distance between them for a parallel-plate capacitor. (See capacitance for details.) The assembly of fixed and movable plates is called an "element" or "capsule". A nearly constant charge is maintained on the capacitor. As the capacitance changes, the charge across the capacitor does change very slightly, but at audible frequencies it is sensibly constant. The capacitance of the capsule (around 5 to 100 pF) and the value of the bias resistor (100 MΩ to tens of GΩ) form a filter that is high-pass for the audio signal, and low-pass for the bias voltage. Note that the time constant of an RC circuit equals the product of the resistance and capacitance. Within the time-frame of the capacitance change (as much as 50 ms at 20 Hz audio signal), the charge is practically constant and the voltage across the capacitor changes instantaneously to reflect the change in capacitance. The voltage across the capacitor varies above and below the bias voltage. The voltage difference between the bias and the capacitor is seen across the series resistor. The voltage across the resistor is amplified for performance or recording. In most cases, the electronics in the microphone itself contribute no voltage gain as the voltage differential is quite significant, up to several volts for high sound levels. Since this is a very high impedance circuit, current gain only is usually needed with the voltage remaining constant. AKG C451B small-diaphragm condenser microphone RF condenser microphones use a comparatively low RF voltage, generated by a low-noise oscillator. The signal from the oscillator may either be amplitude modulated by the capacitance changes produced by the sound waves moving the capsule diaphragm, or the capsule may be part of a resonant circuit that modulates the frequency of the oscillator signal. Demodulation yields a low-noise audio frequency signal with a very low source impedance. The absence of a high bias voltage permits the use of a diaphragm with looser tension, which may be used to achieve wider frequency response due to higher compliance. The RF biasing process results in a lower electrical impedance capsule, a useful by-product of which is that RF condenser microphones can be operated in damp weather conditions that could create problems in DC-biased microphones with contaminated insulating surfaces. The Sennheiser "MKH" series of microphones use the RF biasing technique. Condenser microphones span the range from telephone transmitters through inexpensive karaoke microphones to high-fidelity recording microphones. They generally produce a high-quality audio signal and are now the popular choice in laboratory and recording studio applications. The inherent suitability of this technology is due to the very small mass that must be moved by the incident sound wave, unlike other microphone types that require the sound wave to do more work. They require a power source, provided either via microphone inputs on equipment as phantom power or from a small battery. Power is necessary for establishing the capacitor plate voltage, and is also needed to power the microphone electronics (impedance conversion in the case of electret and DC-polarized microphones, demodulation or detection in the case of RF/HF microphones). Condenser microphones are also available with two diaphragms that can be electrically connected to provide a range of polar patterns (see below), such as cardioid, omnidirectional, and figure-eight. It is also possible to vary the pattern continuously with some microphones, for example the Røde NT2000 or CAD M179. Electret condenser microphone Main article: Electret microphone First patent on foil electret microphone by G. M. Sessler et al. (pages 1 to 3) An electret microphone is a type of capacitor microphone invented at Bell laboratories in 1962 by Gerhard Sessler and Jim West.[4] The externally applied charge described above under condenser microphones is replaced by a permanent charge in an electret material. An electret is a ferroelectric material that has been permanently electrically charged or polarized. The name comes from electrostatic and magnet; a static charge is embedded in an electret by alignment of the static charges in the material, much the way a magnet is made by aligning the magnetic domains in a piece of iron. Due to their good performance and ease of manufacture, hence low cost, the vast majority of microphones made today are electret microphones; a semiconductor manufacturer[5] estimates annual production at over one billion units. Nearly all cell-phone, computer, PDA and headset microphones are electret types. They are used in many applications, from high-quality recording and lavalier use to built-in microphones in small sound recording devices and telephones. Though electret microphones were once considered low quality, the best ones can now rival traditional condenser microphones in every respect and can even offer the long-term stability and ultra-flat response needed for a measurement microphone. Unlike other capacitor microphones, they require no polarizing voltage, but often contain an integrated preamplifier that does require power (often incorrectly called polarizing power or bias). This preamplifier is frequently phantom powered in sound reinforcement and studio applications. Monophonic microphones designed for personal computer (PC) use, sometimes called multimedia microphones, use a 3.5 mm plug as usually used, without power, for stereo; the ring, instead of carrying the signal for a second channel, carries power via a resistor from (normally) a 5 V supply in the computer. Stereophonic microphones use the same connector; there is no obvious way to determine which standard is used by equipment and microphones. Only the best electret microphones rival good DC-polarized units in terms of noise level and quality; electret microphones lend themselves to inexpensive mass-production, while inherently expensive non-electret condenser microphones are made to higher quality. Dynamic microphone Patti Smith singing into a Shure SM58 (dynamic cardioid type) microphone Dynamic microphones work via electromagnetic induction. They are robust, relatively inexpensive and resistant to moisture. This, coupled with their potentially high gain before feedback, makes them ideal for on-stage use. Moving-coil microphones use the same dynamic principle as in a loudspeaker, only reversed. A small movable induction coil, positioned in the magnetic field of a permanent magnet, is attached to the diaphragm. When sound enters through the windscreen of the microphone, the sound wave moves the diaphragm. When the diaphragm vibrates, the coil moves in the magnetic field, producing a varying current in the coil through electromagnetic induction. A single dynamic membrane does not respond linearly to all audio frequencies. Some microphones for this reason utilize multiple membranes for the different parts of the audio spectrum and then combine the resulting signals. Combining the multiple signals correctly is difficult and designs that do this are rare and tend to be expensive. There are on the other hand several designs that are more specifically aimed towards isolated parts of the audio spectrum. The AKG D 112, for example, is designed for bass response rather than treble.[6] In audio engineering several kinds of microphones are often used at the same time to get the best result. Ribbon microphone Main article: Ribbon microphone Edmund Lowe using a ribbon microphone Ribbon microphones use a thin, usually corrugated metal ribbon suspended in a magnetic field. The ribbon is electrically connected to the microphone's output, and its vibration within the magnetic field generates the electrical signal. Ribbon microphones are similar to moving coil microphones in the sense that both produce sound by means of magnetic induction. Basic ribbon microphones detect sound in a bi-directional (also called figure-eight) pattern because the ribbon, which is open to sound both front and back, responds to the pressure gradient rather than the sound pressure. Though the symmetrical front and rear pickup can be a nuisance in normal stereo recording, the high side rejection can be used to advantage by positioning a ribbon microphone horizontally, for example above cymbals, so that the rear lobe picks up only sound from the cymbals. Crossed figure 8, or Blumlein pair, stereo recording is gaining in popularity, and the figure 8 response of a ribbon microphone is ideal for that application. Other directional patterns are produced by enclosing one side of the ribbon in an acoustic trap or baffle, allowing sound to reach only one side. The classic RCA Type 77-DX microphone has several externally adjustable positions of the internal baffle, allowing the selection of several response patterns ranging from "Figure-8" to "Unidirectional". Such older ribbon microphones, some of which still provide high quality sound reproduction, were once valued for this reason, but a good low-frequency response could only be obtained when the ribbon was suspended very loosely, which made them relatively fragile. Modern ribbon materials, including new nanomaterials[7] have now been introduced that eliminate those concerns, and even improve the effective dynamic range of ribbon microphones at low frequencies. Protective wind screens can reduce the danger of damaging a vintage ribbon, and also reduce plosive artifacts in the recording. Properly designed wind screens produce negligible treble attenuation. In common with other classes of dynamic microphone, ribbon microphones don't require phantom power; in fact, this voltage can damage some older ribbon microphones. Some new modern ribbon microphone designs incorporate a preamplifier and, therefore, do require phantom power, and circuits of modern passive ribbon microphones, i.e., those without the aforementioned preamplifier, are specifically designed to resist damage to the ribbon and transformer by phantom power. Also there are new ribbon materials available that are immune to wind blasts and phantom power. Carbon microphone Main article: Carbon microphone A carbon microphone, also known as a carbon button microphone (or sometimes just a button microphone), use a capsule or button containing carbon granules pressed between two metal plates like the Berliner and Edison microphones. A voltage is applied across the metal plates, causing a small current to flow through the carbon. One of the plates, the diaphragm, vibrates in sympathy with incident sound waves, applying a varying pressure to the carbon. The changing pressure deforms the granules, causing the contact area between each pair of adjacent granules to change, and this causes the electrical resistance of the mass of granules to change. The changes in resistance cause a corresponding change in the current flowing through the microphone, producing the electrical signal. Carbon microphones were once commonly used in telephones; they have extremely low-quality sound reproduction and a very limited frequency response range, but are very robust devices. The Boudet microphone, which used relatively large carbon balls, was similar to the granule carbon button microphones.[8] Unlike other microphone types, the carbon microphone can also be used as a type of amplifier, using a small amount of sound energy to control a larger amount of electrical energy. Carbon microphones found use as early telephone repeaters, making long distance phone calls possible in the era before vacuum tubes. These repeaters worked by mechanically coupling a magnetic telephone receiver to a carbon microphone: the faint signal from the receiver was transferred to the microphone, with a resulting stronger electrical signal to send down the line. One illustration of this amplifier effect was the oscillation caused by feedback, resulting in an audible squeal from the old "candlestick" telephone if its earphone was placed near the carbon microphone. Piezoelectric microphone A crystal microphone or piezo microphone uses the phenomenon of piezoelectricity—the ability of some materials to produce a voltage when subjected to pressure—to convert vibrations into an electrical signal. An example of this is potassium sodium tartrate, which is a piezoelectric crystal that works as a transducer, both as a microphone and as a slimline loudspeaker component. Crystal microphones were once commonly supplied with vacuum tube (valve) equipment, such as domestic tape recorders. Their high output impedance matched the high input impedance (typically about 10 megohms) of the vacuum tube input stage well. They were difficult to match to early transistor equipment, and were quickly supplanted by dynamic microphones for a time, and later small electret condenser devices. The high impedance of the crystal microphone made it very susceptible to handling noise, both from the microphone itself and from the connecting cable. Piezoelectric transducers are often used as contact microphones to amplify sound from acoustic musical instruments, to sense drum hits, for triggering electronic samples, and to record sound in challenging environments, such as underwater under high pressure. Saddle-mounted pickups on acoustic guitars are generally piezoelectric devices that contact the strings passing over the saddle. This type of microphone is different from magnetic coil pickups commonly visible on typical electric guitars, which use magnetic induction, rather than mechanical coupling, to pick up vibration. Fiber optic microphone The Optoacoustics 1140 fiber optic microphone A fiber optic microphone converts acoustic waves into electrical signals by sensing changes in light intensity, instead of sensing changes in capacitance or magnetic fields as with conventional microphones.[9][10] During operation, light from a laser source travels through an optical fiber to illuminate the surface of a reflective diaphragm. Sound vibrations of the diaphragm modulate the intensity of light reflecting off the diaphragm in a specific direction. The modulated light is then transmitted over a second optical fiber to a photo detector, which transforms the intensity-modulated light into analog or digital audio for transmission or recording. Fiber optic microphones possess high dynamic and frequency range, similar to the best high fidelity conventional microphones. Fiber optic microphones do not react to or influence any electrical, magnetic, electrostatic or radioactive fields (this is called EMI/RFI immunity). The fiber optic microphone design is therefore ideal for use in areas where conventional microphones are ineffective or dangerous, such as inside industrial turbines or in magnetic resonance imaging (MRI) equipment environments. Fiber optic microphones are robust, resistant to environmental changes in heat and moisture, and can be produced for any directionality or impedance matching. The distance between the microphone's light source and its photo detector may be up to several kilometers without need for any preamplifier and/or other electrical device, making fiber optic microphones suitable for industrial and surveillance acoustic monitoring. Fiber optic microphones are used in very specific application areas such as for infrasound monitoring and noise-canceling. They have proven especially useful in medical applications, such as allowing radiologists, staff and patients within the powerful and noisy magnetic field to converse normally, inside the MRI suites as well as in remote control rooms.[11]) Other uses include industrial equipment monitoring and sensing, audio calibration and measurement, highfidelity recording and law enforcement. Laser microphone Main article: Laser microphone Laser microphones are often portrayed in movies as spy gadgets, because they can be used to pick up sound at a distance from the microphone equipment. A laser beam is aimed at the surface of a window or other plane surface that is affected by sound. The vibrations of this surface change the angle at which the beam is reflected, and the motion of the laser spot from the returning beam is detected and converted to an audio signal. In a more robust and expensive implementation, the returned light is split and fed to an interferometer, which detects movement of the surface by changes in the optical path length of the reflected beam. The former implementation is a tabletop experiment; the latter requires an extremely stable laser and precise optics. A new type of laser microphone is a device that uses a laser beam and smoke or vapor to detect sound vibrations in free air. On 25 August 2009, U.S. patent 7,580,533 issued for a Particulate Flow Detection Microphone based on a laser-photocell pair with a moving stream of smoke or vapor in the laser beam's path. Sound pressure waves cause disturbances in the smoke that in turn cause variations in the amount of laser light reaching the photo detector. A prototype of the device was demonstrated at the 127th Audio Engineering Society convention in New York City from 9 through 12 October 2009. Liquid microphone Main article: Water microphone Early microphones did not produce intelligible speech, until Alexander Graham Bell made improvements including a variable resistance microphone/transmitter. Bell's liquid transmitter consisted of a metal cup filled with water with a small amount of sulfuric acid added. A sound wave caused the diaphragm to move, forcing a needle to move up and down in the water. The electrical resistance between the wire and the cup was then inversely proportional to the size of the water meniscus around the submerged needle. Elisha Gray filed a caveat for a version using a brass rod instead of the needle. Other minor variations and improvements were made to the liquid microphone by Majoranna, Chambers, Vanni, Sykes, and Elisha Gray, and one version was patented by Reginald Fessenden in 1903. These were the first working microphones, but they were not practical for commercial application. The famous first phone conversation between Bell and Watson took place using a liquid microphone. MEMS microphone Main article: Microelectromechanical systems The MEMS (MicroElectrical-Mechanical System) microphone is also called a microphone chip or silicon microphone. The pressure-sensitive diaphragm is etched directly into a silicon chip by MEMS techniques, and is usually accompanied with integrated preamplifier. Most MEMS microphones are variants of the condenser microphone design. Often MEMS microphones have built in analog-to-digital converter (ADC) circuits on the same CMOS chip making the chip a digital microphone and so more readily integrated with modern digital products. Major manufacturers producing MEMS silicon microphones are Wolfson Microelectronics (WM7xxx), Analog Devices, Akustica (AKU200x), Infineon (SMM310 product), Knowles Electronics, Memstech (MSMx), NXP Semiconductors, Sonion MEMS, AAC Acoustic Technologies,[12] and Omron.[13] Speakers as microphones A loudspeaker, a transducer that turns an electrical signal into sound waves, is the functional opposite of a microphone. Since a conventional speaker is constructed much like a dynamic microphone (with a diaphragm, coil and magnet), speakers can actually work "in reverse" as microphones. The result, though, is a microphone with poor quality, limited frequency response (particularly at the high end), and poor sensitivity. In practical use, speakers are sometimes used as microphones in applications where high quality and sensitivity are not needed such as intercoms, walkie-talkies or Video game voice chat peripherals, or when conventional microphones are in short supply. However, there is at least one other practical application of this principle: Using a medium-size woofer placed closely in front of a "kick" (bass drum) in a drum set to act as a microphone. The use of relatively large speakers to transduce low frequency sound sources, especially in music production, is becoming fairly common. A product example of this type of device is the Yamaha Subkick, a 6.5-inch (170 mm) woofer shock-mounted it into a 10" drum shell used in front of kick drums. Since a relatively massive membrane is unable to transduce high frequencies, placing a speaker in front of a kick drum is often ideal for reducing cymbal and snare bleed into the kick drum sound. Less commonly, microphones themselves can be used as speakers, almost always as tweeters. Microphones, however, are not designed to handle the power that speaker components are routinely required to cope with. One instance of such an application was the STC microphone-derived 4001 super-tweeter, which was successfully used in a number of high quality loudspeaker systems from the late 1960s to the mid-70s. Capsule design and directivity The inner elements of a microphone are the primary source of differences in directivity. A pressure microphone uses a diaphragm between a fixed internal volume of air and the environment, and responds uniformly to pressure from all directions, so it is said to be omnidirectional. A pressure-gradient microphone uses a diaphragm that is at least partially open on both sides. The pressure difference between the two sides produces its directional characteristics. Other elements such as the external shape of the microphone and external devices such as interference tubes can also alter a microphone's directional response. A pure pressuregradient microphone is equally sensitive to sounds arriving from front or back, but insensitive to sounds arriving from the side because sound arriving at the front and back at the same time creates no gradient between the two. The characteristic directional pattern of a pure pressuregradient microphone is like a figure-8. Other polar patterns are derived by creating a capsule that combines these two effects in different ways. The cardioid, for instance, features a partially closed backside, so its response is a combination of pressure and pressure-gradient characteristics.[14] Microphone polar patterns (Microphone facing top of page in diagram, parallel to page): Omnidirectional Subcardioid Cardioid Supercardioid Bi-directional or Figure of 8 Hypercardioid Shotgun A microphone's directionality or polar pattern indicates how sensitive it is to sounds arriving at different angles about its central axis. The polar patterns illustrated above represent the locus of points that produce the same signal level output in the microphone if a given sound pressure level (SPL) is generated from that point. How the physical body of the microphone is oriented relative to the diagrams depends on the microphone design. For large-membrane microphones such as in the Oktava (pictured above), the upward direction in the polar diagram is usually perpendicular to the microphone body, commonly known as "side fire" or "side address". For small diaphragm microphones such as the Shure (also pictured above), it usually extends from the axis of the microphone commonly known as "end fire" or "top/end address". Some microphone designs combine several principles in creating the desired polar pattern. This ranges from shielding (meaning diffraction/dissipation/absorption) by the housing itself to electronically combining dual membranes. Omnidirectional An omnidirectional (or nondirectional) microphone's response is generally considered to be a perfect sphere in three dimensions. In the real world, this is not the case. As with directional microphones, the polar pattern for an "omnidirectional" microphone is a function of frequency. The body of the microphone is not infinitely small and, as a consequence, it tends to get in its own way with respect to sounds arriving from the rear, causing a slight flattening of the polar response. This flattening increases as the diameter of the microphone (assuming it's cylindrical) reaches the wavelength of the frequency in question. Therefore, the smallest diameter microphone gives the best omnidirectional characteristics at high frequencies. The wavelength of sound at 10 kHz is little over an inch (3.4 cm) so the smallest measuring microphones are often 1/4" (6 mm) in diameter, which practically eliminates directionality even up to the highest frequencies. Omnidirectional microphones, unlike cardioids, do not employ resonant cavities as delays, and so can be considered the "purest" microphones in terms of low coloration; they add very little to the original sound. Being pressure-sensitive they can also have a very flat low-frequency response down to 20 Hz or below. Pressure-sensitive microphones also respond much less to wind noise and plosives than directional (velocity sensitive) microphones. An example of a nondirectional microphone is the round black eight ball.[15] Unidirectional A unidirectional microphone is sensitive to sounds from only one direction. The diagram above illustrates a number of these patterns. The microphone faces upwards in each diagram. The sound intensity for a particular frequency is plotted for angles radially from 0 to 360°. (Professional diagrams show these scales and include multiple plots at different frequencies. The diagrams given here provide only an overview of typical pattern shapes, and their names.) Cardioid US664A University Sound Dynamic Supercardioid Microphone The most common unidirectional microphone is a cardioid microphone, so named because the sensitivity pattern is heart-shaped. A hyper-cardioid microphone is similar but with a tighter area of front sensitivity and a smaller lobe of rear sensitivity. A super-cardioid microphone is similar to a hyper-cardioid, except there is more front pickup and less rear pickup. These three patterns are commonly used as vocal or speech microphones, since they are good at rejecting sounds from other directions. A cardioid microphone is effectively a superposition of an omnidirectional and a figure-8 microphone; for sound waves coming from the back, the negative signal from the figure-8 cancels the positive signal from the omnidirectional element, whereas for sound waves coming from the front, the two add to each other. A hypercardioid microphone is similar, but with a slightly larger figure-8 contribution. Since pressure gradient transducer microphones are directional, putting them very close to the sound source (at distances of a few centimeters) results in a bass boost. This is known as the proximity effect.[16] Bi-directional "Figure 8" or bi-directional microphones receive sound equally from both the front and back of the element. Most ribbon microphones are of this pattern. In principle they do not respond to sound pressure at all, only to the gradient between front and back; since sound arriving from the side reaches front and back equally there is no gradient and therefore no sensitivity to sound from that direction. While omnidirectional microphones are scalar transducers responding to pressure from any direction, bi-directional microphones are vector transducers responding to the gradient along an axis normal to the plane of the diaphragm. As a result, output polarity is inverted for sounds arriving from the back side. Shotgun An Audio-Technica shotgun microphone Shotgun microphones are the most highly directional. They have small lobes of sensitivity to the left, right, and rear but are significantly less sensitive to the side and rear than other directional microphones. This results from placing the element at the back end of a tube with slots cut along the side; wave cancellation eliminates much of the off-axis sound. Due to the narrowness of their sensitivity area, shotgun microphones are commonly used on television and film sets, in stadiums, and for field recording of wildlife. Boundary or "PZM" Several approaches have been developed for effectively using a microphone in less-than-ideal acoustic spaces, which often suffer from excessive reflections from one or more of the surfaces (boundaries) that make up the space. If the microphone is placed in, or very close to, one of these boundaries, the reflections from that surface are not sensed by the microphone. Initially this was done by placing an ordinary microphone adjacent to the surface, sometimes in a block of acoustically transparent foam. Sound engineers Ed Long and Ron Wickersham developed the concept of placing the diaphgram parallel to and facing the boundary.[17] While the patent has expired, "Pressure Zone Microphone" and "PZM" are still active trademarks of Crown International, and the generic term "boundary microphone" is preferred. While a boundary microphone was initially implemented using an omnidirectional element, it is also possible to mount a directional microphone close enough to the surface to gain some of the benefits of this technique while retaining the directional properties of the element. Crown's trademark on this approach is "Phase Coherent Cardioid" or "PCC," but there are other makers who employ this technique as well. Application-specific designs A lavalier microphone is made for hands-free operation. These small microphones are worn on the body. Originally, they were held in place with a lanyard worn around the neck, but more often they are fastened to clothing with a clip, pin, tape or magnet. The lavalier cord may be hidden by clothes and either run to an RF transmitter in a pocket or clipped to a belt (for mobile use), or run directly to the mixer (for stationary applications). A wireless microphone transmits the audio as a radio or optical signal rather than via a cable. It usually sends its signal using a small FM radio transmitter to a nearby receiver connected to the sound system, but it can also use infrared waves if the transmitter and receiver are within sight of each other. A contact microphone picks up vibrations directly from a solid surface or object, as opposed to sound vibrations carried through air. One use for this is to detect sounds of a very low level, such as those from small objects or insects. The microphone commonly consists of a magnetic (moving coil) transducer, contact plate and contact pin. The contact plate is placed directly on the vibrating part of a musical instrument or other surface, and the contact pin transfers vibrations to the coil. Contact microphones have been used to pick up the sound of a snail's heartbeat and the footsteps of ants. A portable version of this microphone has recently been developed. A throat microphone is a variant of the contact microphone that picks up speech directly from a person's throat, which it is strapped to. This lets the device be used in areas with ambient sounds that would otherwise make the speaker inaudible. A parabolic microphone uses a parabolic reflector to collect and focus sound waves onto a microphone receiver, in much the same way that a parabolic antenna (e.g. satellite dish) does with radio waves. Typical uses of this microphone, which has unusually focused front sensitivity and can pick up sounds from many meters away, include nature recording, outdoor sporting events, eavesdropping, law enforcement, and even espionage. Parabolic microphones are not typically used for standard recording applications, because they tend to have poor low-frequency response as a side effect of their design. A stereo microphone integrates two microphones in one unit to produce a stereophonic signal. A stereo microphone is often used for broadcast applications or field recording where it would be impractical to configure two separate condenser microphones in a classic X-Y configuration (see microphone practice) for stereophonic recording. Some such microphones have an adjustable angle of coverage between the two channels. A noise-canceling microphone is a highly directional design intended for noisy environments. One such use is in aircraft cockpits where they are normally installed as boom microphones on headsets. Another use is in live event support on loud concert stages for vocalists involved with live performances. Many noise-canceling microphones combine signals received from two diaphragms that are in opposite electrical polarity or are processed electronically. In dual diaphragm designs, the main diaphragm is mounted closest to the intended source and the second is positioned farther away from the source so that it can pick up environmental sounds to be subtracted from the main diaphragm's signal. After the two signals have been combined, sounds other than the intended source are greatly reduced, substantially increasing intelligibility. Other noise-canceling designs use one diaphragm that is affected by ports open to the sides and rear of the microphone, with the sum being a 16 dB rejection of sounds that are farther away. One noisecanceling headset design using a single diaphragm has been used prominently by vocal artists such as Garth Brooks and Janet Jackson.[18] A few noise-canceling microphones are throat microphones. Connectors Electronic symbol for a microphone The most common connectors used by microphones are: Male XLR connector on professional microphones ¼ inch (sometimes referred to as 6.3 mm) jack plug also known as 1/4 inch TRS connector on less expensive consumer microphones. Many consumer microphones use an unbalanced 1/4 inch phone jack. Harmonica microphones commonly use a high impedance 1/4 inch TS connection to be run through guitar amplifiers. 3.5 mm (sometimes referred to as 1/8 inch mini) stereo (wired as mono) mini phone plug on very inexpensive and computer microphones A microphone with a USB connector, made by Blue Microphones Some microphones use other connectors, such as a 5-pin XLR, or mini XLR for connection to portable equipment. Some lavalier (or 'lapel', from the days of attaching the microphone to the news reporters suit lapel) microphones use a proprietary connector for connection to a wireless transmitter. Since 2005, professional-quality microphones with USB connections have begun to appear, designed for direct recording into computer-based software. Impedance-matching Microphones have an electrical characteristic called impedance, measured in ohms (Ω), that depends on the design. Typically, the rated impedance is stated.[19] Low impedance is considered under 600 Ω. Medium impedance is considered between 600 Ω and 10 kΩ. High impedance is above 10 kΩ. Owing to their built-in amplifier, condenser microphones typically have an output impedance between 50 and 200 Ω.[20] The output of a given microphone delivers the same power whether it is low or high impedance. If a microphone is made in high and low impedance versions, the high impedance version has a higher output voltage for a given sound pressure input, and is suitable for use with vacuum-tube guitar amplifiers, for instance, which have a high input impedance and require a relatively high signal input voltage to overcome the tubes' inherent noise. Most professional microphones are low impedance, about 200 Ω or lower. Professional vacuum-tube sound equipment incorporates a transformer that steps up the impedance of the microphone circuit to the high impedance and voltage needed to drive the input tube; the impedance conversion inherently creates voltage gain as well. External matching transformers are also available that can be used in-line between a low impedance microphone and a high impedance input. Low-impedance microphones are preferred over high impedance for two reasons: one is that using a high-impedance microphone with a long cable results in high frequency signal loss due to cable capacitance, which forms a low-pass filter with the microphone output impedance. The other is that long high-impedance cables tend to pick up more hum (and possibly radio-frequency interference (RFI) as well). Nothing is damaged if the impedance between microphone and other equipment is mismatched; the worst that happens is a reduction in signal or change in frequency response. Most microphones are designed not to have their impedance matched by the load they are connected to.[21] Doing so can alter their frequency response and cause distortion, especially at high sound pressure levels. Certain ribbon and dynamic microphones are exceptions, due to the designers' assumption of a certain load impedance being part of the internal electro-acoustical damping circuit of the microphone.[22][dubious – discuss] Digital microphone interface Neumann D-01 digital microphone and Neumann DMI-8 8-channel USB Digital Microphone Interface The AES 42 standard, published by the Audio Engineering Society, defines a digital interface for microphones. Microphones conforming to this standard directly output a digital audio stream through an XLR or XLD male connector, rather than producing an analog output. Digital microphones may be used either with new equipment with appropriate input connections that conform to the AES 42 standard, or else via a suitable interface box. Studio-quality microphones that operate in accordance with the AES 42 standard are now available from a number of microphone manufacturers. Measurements and specifications A comparison of the far field on-axis frequency response of the Oktava 319 and the Shure SM58 Because of differences in their construction, microphones have their own characteristic responses to sound. This difference in response produces non-uniform phase and frequency responses. In addition, microphones are not uniformly sensitive to sound pressure, and can accept differing levels without distorting. Although for scientific applications microphones with a more uniform response are desirable, this is often not the case for music recording, as the non-uniform response of a microphone can produce a desirable coloration of the sound. There is an international standard for microphone specifications,[19] but few manufacturers adhere to it. As a result, comparison of published data from different manufacturers is difficult because different measurement techniques are used. The Microphone Data Website has collated the technical specifications complete with pictures, response curves and technical data from the microphone manufacturers for every currently listed microphone, and even a few obsolete models, and shows the data for them all in one common format for ease of comparison.[1]. Caution should be used in drawing any solid conclusions from this or any other published data, however, unless it is known that the manufacturer has supplied specifications in accordance with IEC 60268-4. A frequency response diagram plots the microphone sensitivity in decibels over a range of frequencies (typically 20 Hz to 20 kHz), generally for perfectly on-axis sound (sound arriving at 0° to the capsule). Frequency response may be less informatively stated textually like so: "30 Hz–16 kHz ±3 dB". This is interpreted as meaning a nearly flat, linear, plot between the stated frequencies, with variations in amplitude of no more than plus or minus 3 dB. However, one cannot determine from this information how smooth the variations are, nor in what parts of the spectrum they occur. Note that commonly made statements such as "20 Hz–20 kHz" are meaningless without a decibel measure of tolerance. Directional microphones' frequency response varies greatly with distance from the sound source, and with the geometry of the sound source. IEC 60268-4 specifies that frequency response should be measured in plane progressive wave conditions (very far away from the source) but this is seldom practical. Close talking microphones may be measured with different sound sources and distances, but there is no standard and therefore no way to compare data from different models unless the measurement technique is described. The self-noise or equivalent noise level is the sound level that creates the same output voltage as the microphone does in the absence of sound. This represents the lowest point of the microphone's dynamic range, and is particularly important should you wish to record sounds that are quiet. The measure is often stated in dB(A), which is the equivalent loudness of the noise on a decibel scale frequency-weighted for how the ear hears, for example: "15 dBA SPL" (SPL means sound pressure level relative to 20 micropascals). The lower the number the better. Some microphone manufacturers state the noise level using ITU-R 468 noise weighting, which more accurately represents the way we hear noise, but gives a figure some 11–14 dB higher. A quiet microphone typically measures 20 dBA SPL or 32 dB SPL 468-weighted. Very quiet microphones have existed for years for special applications, such the Brüel & Kjaer 4179, with a noise level around 0 dB SPL. Recently some microphones with low noise specifications have been introduced in the studio/entertainment market, such as models from Neumann and Røde that advertise noise levels between 5–7 dBA. Typically this is achieved by altering the frequency response of the capsule and electronics to result in lower noise within the A-weighting curve while broadband noise may be increased. The maximum SPL the microphone can accept is measured for particular values of total harmonic distortion (THD), typically 0.5%. This amount of distortion is generally inaudible, so one can safely use the microphone at this SPL without harming the recording. Example: "142 dB SPL peak (at 0.5% THD)". The higher the value, the better, although microphones with a very high maximum SPL also have a higher self-noise. The clipping level is an important indicator of maximum usable level, as the 1% THD figure usually quoted under max SPL is really a very mild level of distortion, quite inaudible especially on brief high peaks. Clipping is much more audible. For some microphones the clipping level may be much higher than the max SPL. The dynamic range of a microphone is the difference in SPL between the noise floor and the maximum SPL. If stated on its own, for example "120 dB", it conveys significantly less information than having the self-noise and maximum SPL figures individually. Sensitivity indicates how well the microphone converts acoustic pressure to output voltage. A high sensitivity microphone creates more voltage and so needs less amplification at the mixer or recording device. This is a practical concern but is not directly an indication of the mic's quality, and in fact the term sensitivity is something of a misnomer, 'transduction gain' being perhaps more meaningful, (or just "output level") because true sensitivity is generally set by the noise floor, and too much "sensitivity" in terms of output level compromises the clipping level. There are two common measures. The (preferred) international standard is made in millivolts per pascal at 1 kHz. A higher value indicates greater sensitivity. The older American method is referred to a 1 V/Pa standard and measured in plain decibels, resulting in a negative value. Again, a higher value indicates greater sensitivity, so −60 dB is more sensitive than −70 dB. Measurement microphones Some microphones are intended for testing speakers, measuring noise levels and otherwise quantifying an acoustic experience. These are calibrated transducers and are usually supplied with a calibration certificate that states absolute sensitivity against frequency. The quality of measurement microphones is often referred to using the designations "Class 1," "Type 2" etc., which are references not to microphone specifications but to sound level meters.[23] A more comprehensive standard[24] for the description of measurement microphone performance was recently adopted. Measurement microphones are generally scalar sensors of pressure; they exhibit an omnidirectional response, limited only by the scattering profile of their physical dimensions. Sound intensity or sound power measurements require pressure-gradient measurements, which are typically made using arrays of at least two microphones, or with hot-wire anemometers. Microphone calibration Main article: Measurement microphone calibration To take a scientific measurement with a microphone, its precise sensitivity must be known (in volts per pascal). Since this may change over the lifetime of the device, it is necessary to regularly calibrate measurement microphones. This service is offered by some microphone manufacturers and by independent certified testing labs. All microphone calibration is ultimately traceable to primary standards at a national measurement institute such as NPL in the UK, PTB in Germany and NIST in the USA, which most commonly calibrate using the reciprocity primary standard. Measurement microphones calibrated using this method can then be used to calibrate other microphones using comparison calibration techniques. Depending on the application, measurement microphones must be tested periodically (every year or several months, typically) and after any potentially damaging event, such as being dropped (most such mikes come in foam-padded cases to reduce this risk) or exposed to sounds beyond the acceptable level. Microphone array and array microphones Main article: Microphone array A microphone array is any number of microphones operating in tandem. There are many applications: Systems for extracting voice input from ambient noise (notably telephones, speech recognition systems, hearing aids) Surround sound and related technologies Locating objects by sound: acoustic source localization, e.g. military use to locate the source(s) of artillery fire. Aircraft location and tracking. High fidelity original recordings 3D spatial beamforming for localized acoustic detection of subcutaneous sounds Typically, an array is made up of omnidirectional microphones distributed about the perimeter of a space, linked to a computer that records and interprets the results into a coherent form. Microphone windscreens Various microphone covers Windscreens[note 1] are used to protect microphones that would otherwise be buffeted by wind or vocal plosives from consonants such as "P", "B", etc. Most microphones have an integral windscreen built around the microphone diaphragm. A screen of plastic, wire mesh or a metal cage is held at a distance from the microphone diaphragm, to shield it. This cage provides a first line of defense against the mechanical impact of objects or wind. Some microphones, such as the Shure SM58, may have an additional layer of foam inside the cage to further enhance the protective properties of the shield. One disadvantage of all windscreen types is that the microphone's high frequency response is attenuated by a small amount, depending on the density of the protective layer. Beyond integral microphone windscreens, there are three broad classes of additional wind protection. Microphone covers Microphone covers are often made of soft open-cell polyester or polyurethane foam because of the inexpensive, disposable nature of the foam. Optional windscreens are often available from the manufacturer and third parties. A visible example of an optional accessory windscreen is the A2WS from Shure, one of which is fitted over each of the two Shure SM57 microphones used on the United States president's lectern.[25] One disadvantage of polyurethane foam microphone covers is that they can deteriorate over time. Windscreens also tend to collect dirt and moisture in their open cells and must be cleaned to prevent high frequency loss, bad odor and unhealthy conditions for the person using the microphone. On the other hand, a major advantage of concert vocalist windscreens is that one can quickly change to a clean windscreen between users, reducing the chance of transferring germs. Windscreens of various colors can be used to distinguish one microphone from another on a busy, active stage. Pop filter by Gauge Precision Instruments Pop filters Pop filters or pop screens are used in controlled studio environments to minimize plosives when recording. A typical pop filter is composed of one or more layers of acoustically transparent gauze-like material, such as woven nylon (e.g. pantyhose) stretched over a circular frame and a clamp and a flexible mounting bracket to attach to the microphone stand. The pop shield is placed between the vocalist and the microphone. The closer a vocalist brings his or her lips to the microphone, the greater the requirement for a Pop filter. Singers can be trained either to soften their plosives or direct the air blast away from the microphone, in which cases they don't need a pop filter. Pop filters also keep spittle off the microphone. Most condenser microphones can be damaged by spittle. Blimps Two recordings being made—A blimp is being used on the left. An open-cell foam windscreen is being used on the right. a 'dead cat' and a 'dead kitten' windscreens. The dead kitten covers a stereo mic for a DSLR camera. The difference in name is due to the size of the fur. Blimps (also known as Zeppelins) are large, hollow windscreens used to surround microphones for outdoor location audio, such as nature recording, electronic news gathering, and for film and video shoots. They can cut wind noise by as much as 25 dB, especially low-frequency noise. The blimp is essentially a hollow cage or basket with acoustically transparent material stretched over the outer frame. The blimp works by creating a volume of still air around the microphone. The microphone is often further isolated from the blimp by an elastic suspension inside the basket. This reduces wind vibrations and handling noise transmitted from the cage. To extend the range of wind speed conditions in which the blimp remains effective, many have the option of a secondary cover over the outer shell. This is usually an acoustically transparent, synthetic fur material with long, soft hairs. Common and slang names for this include "dead cat" or "windmuff". The hairs deaden the noise caused by the shock of wind hitting the blimp. A synthetic fur cover can reduce wind noise by an additional 10 dB.[26] Amplifier Generally, an amplifier or simply amp, is a device for increasing the power of a signal. In popular use, the term usually describes an electronic amplifier, in which the input "signal" is usually a voltage or a current. In audio applications, amplifiers drive the loudspeakers used in PA systems to make the human voice louder or play recorded music. Amplifiers may be classified according to the input (source) they are designed to amplify (such as a guitar amplifier, to perform with an electric guitar), the device they are intended to drive (such as a headphone amplifier), the frequency range of the signals (Audio, IF, RF, and VHF amplifiers, for example), whether they invert the signal (inverting amplifiers and non-inverting amplifiers), or the type of device used in the amplification (valve or tube amplifiers, FET amplifiers, etc.). A related device that emphasizes conversion of signals of one type to another (for example, a light signal in photons to a DC signal in amperes) is a transducer, a transformer, or a sensor. However, none of these amplify power. Figures of merit The quality of an amplifier can be characterized by a number of specifications, listed below. [edit] Gain The gain of an amplifier is the ratio of output to input power or amplitude, and is usually measured in decibels. (When measured in decibels it is logarithmically related to the power ratio: G(dB)=10 log(Pout /(Pin)). RF amplifiers are often specified in terms of the maximum power gain obtainable, while the voltage gain of audio amplifiers and instrumentation amplifiers will be more often specified (since the amplifier's input impedance will often be much higher than the source impedance, and the load impedance higher than the amplifier's output impedance). Example: an audio amplifier with a gain given as 20 dB will have a voltage gain of ten (but a power gain of 100 would only occur in the unlikely event the input and output impedances were identical). If two equivalent amplifiers are being compared, the amplifier with higher gain settings would be more sensitive as it would take less input signal to produce a given amount of power.[1] [edit] Bandwidth The bandwidth of an amplifier is the range of frequencies for which the amplifier gives "satisfactory performance". The definition of "satisfactory performance" may be different for different applications. However, a common and well-accepted metric is the half power points (i.e. frequency where the power goes down by half its peak value) on the output vs. frequency curve. Therefore bandwidth can be defined as the difference between the lower and upper half power points. This is therefore also known as the −3 dB bandwidth. Bandwidths (otherwise called "frequency responses") for other response tolerances are sometimes quoted (−1 dB, −6 dB etc.) or "plus or minus 1dB" (roughly the sound level difference people usually can detect). The gain of a good quality full-range audio amplifier will be essentially flat between 20 Hz to about 20 kHz (the range of normal human hearing). In ultra high fidelity amplifier design, the amp's frequency response should extend considerably beyond this (one or more octaves either side) and might have −3 dB points < 10 Hz and > 65 kHz. Professional touring amplifiers often have input and/or output filtering to sharply limit frequency response beyond 20 Hz-20 kHz; too much of the amplifier's potential output power would otherwise be wasted on infrasonic and ultrasonic frequencies, and the danger of AM radio interference would increase. Modern switching amplifiers need steep low pass filtering at the output to get rid of high frequency switching noise and harmonics. [edit] Efficiency Efficiency is a measure of how much of the power source is usefully applied to the amplifier's output. Class A amplifiers are very inefficient, in the range of 10–20% with a max efficiency of 25% for direct coupling of the output. Inductive coupling of the output can raise their efficiency to a maximum of 50%. Drain efficiency is the ratio of output RF power to input DC power when primary input DC power has been fed to the drain of an FET. Based on this definition, the drain efficiency cannot exceed 25% for a class A amplifier that is supplied drain bias current through resistors (because RF signal has its zero level at about 50% of the input DC). Manufacturers specify much higher drain efficiencies, and designers are able to obtain higher efficiencies by providing current to the drain of the transistor through an inductor or a transformer winding. In this case the RF zero level is near the DC rail and will swing both above and below the rail during operation. While the voltage level is above the DC rail current is supplied by the inductor. Class B amplifiers have a very high efficiency but are impractical for audio work because of high levels of distortion (See: Crossover distortion). In practical design, the result of a tradeoff is the class AB design. Modern Class AB amplifiers commonly have peak efficiencies between 30– 55% in audio systems and 50-70% in radio frequency systems with a theoretical maximum of 78.5%. Commercially available Class D switching amplifiers have reported efficiencies as high as 90%. Amplifiers of Class C-F are usually known to be very high efficiency amplifiers. RCA manufactured an AM broadcast transmitter employing a single class-C low mu triode with an RF efficiency in the 90% range. More efficient amplifiers run cooler, and often do not need any cooling fans even in multikilowatt designs. The reason for this is that the loss of efficiency produces heat as a by-product of the energy lost during the conversion of power. In more efficient amplifiers there is less loss of energy so in turn less heat. In RF linear Power Amplifiers, such as cellular base stations and broadcast transmitters, special design techniques can be used to improve efficiency. Doherty designs, which use a second output stage as a "peak" amplifier, can lift efficiency from the typical 15% up to 30-35% in a narrow bandwidth. Envelope Tracking designs are able to achieve efficiencies of up to 60%, by modulating the supply voltage to the amplifier in line with the envelope of the signal. [edit] Linearity An ideal amplifier would be a totally linear device, but real amplifiers are only linear within limits. When the signal drive to the amplifier is increased, the output also increases until a point is reached where some part of the amplifier becomes saturated and cannot produce any more output; this is called clipping, and results in distortion. In most amplifiers a reduction in gain takes place before hard clipping occurs; the result is a compression effect, which (if the amplifier is an audio amplifier) sounds much less unpleasant to the ear. For these amplifiers, the 1 dB compression point is defined as the input power (or output power) where the gain is 1 dB less than the small signal gain. Sometimes this nonlinearity is deliberately designed in to reduce the audible unpleasantness of hard clipping under overload. Ill effects of nonlinearity can be reduced with negative feedback. Linearization is an emergent field, and there are many techniques, such as feedforward, predistortion, postdistortion, in order to avoid the undesired effects of the non-linearities. [edit] Noise This is a measure of how much noise is introduced in the amplification process. Noise is an undesirable but inevitable product of the electronic devices and components; also, much noise results from intentional economies of manufacture and design time. The metric for noise performance of a circuit is noise figure or noise factor. Noise figure is a comparison between the output signal to noise ratio and the thermal noise of the input signal. [edit] Output dynamic range Output dynamic range is the range, usually given in dB, between the smallest and largest useful output levels. The lowest useful level is limited by output noise, while the largest is limited most often by distortion. The ratio of these two is quoted as the amplifier dynamic range. More precisely, if S = maximal allowed signal power and N = noise power, the dynamic range DR is DR = (S + N ) /N.[2] In many switched mode amplifiers, dynamic range is limited by the minimum output step size. [edit] Slew rate Slew rate is the maximum rate of change of the output, usually quoted in volts per second (or microsecond). Many amplifiers are ultimately slew rate limited (typically by the impedance of a drive current having to overcome capacitive effects at some point in the circuit), which sometimes limits the full power bandwidth to frequencies well below the amplifier's small-signal frequency response. [edit] Rise time The rise time, tr, of an amplifier is the time taken for the output to change from 10% to 90% of its final level when driven by a step input. For a Gaussian response system (or a simple RC roll off), the rise time is approximated by: tr * BW = 0.35, where tr is rise time in seconds and BW is bandwidth in Hz. [edit] Settling time and ringing The time taken for the output to settle to within a certain percentage of the final value (for instance 0.1%) is called the settling time, and is usually specified for oscilloscope vertical amplifiers and high accuracy measurement systems. Ringing refers to an output variation that cycles above and below an amplifier's final value and leads to a delay in reaching a stable output. Ringing is the result of overshoot caused by an underdamped circuit. [edit] Overshoot In response to a step input, the overshoot is the amount the output exceeds its final, steady-state value. [edit] Stability Stability is an issue in all amplifiers with feedback, whether that feedback is added intentionally or results unintentionally. It is especially an issue when applied over multiple amplifying stages. Stability is a major concern in RF and microwave amplifiers. The degree of an amplifier's stability can be quantified by a so-called stability factor. There are several different stability factors, such as the Stern stability factor and the Linvil stability factor, which specify a condition that must be met for the absolute stability of an amplifier in terms of its two-port parameters. [edit] Electronic amplifiers Main article: Electronic amplifier There are many types of electronic amplifiers, commonly used in radio and television transmitters and receivers, high-fidelity ("hi-fi") stereo equipment, microcomputers and other electronic digital equipment, and guitar and other instrument amplifiers. Critical components include active devices, such as vacuum tubes or transistors. [edit] Other amplifier types [edit] Carbon microphone One of the first devices used to amplify signals was the carbon microphone (effectively a soundcontrolled variable resistor). By channeling a large electric current through the compressed carbon granules in the microphone, a small sound signal could produce a much larger electric signal. The carbon microphone was extremely important in early telecommunications; analog telephones in fact work without the use of any other amplifier. Before the invention of electronic amplifiers, mechanically coupled carbon microphones were also used as amplifiers in telephone repeaters for long distance service. [edit] Magnetic amplifier Main article: magnetic amplifier A magnetic amplifier is a transformer-like device that makes use of the saturation of magnetic materials to produce amplification. It is a non-electronic electrical amplifier with no moving parts. The bandwidth of magnetic amplifiers extends to the hundreds of kilohertz. [edit] Rotating electrical machinery amplifier A Ward Leonard control is a rotating machine like an electrical generator that provides amplification of electrical signals by the conversion of mechanical energy to electrical energy. Changes in generator field current result in larger changes in the output current of the generator, providing gain. This class of device was used for smooth control of large motors, primarily for elevators and naval guns. Field modulation of a very high speed AC generator was also used for some early AM radio transmissions.[3] See Alexanderson alternator. [edit] Johnsen-Rahbek effect amplifier The earliest form of audio power amplifier was Edison's "electromotograph" loud-speaking telephone, which used a wetted rotating chalk cylinder in contact with a stationary contact. The friction between cylinder and contact varied with the current, providing gain. Edison discovered this effect in 1874, but the theory behind the Johnsen-Rahbek effect was not understood until the semiconductor era. [edit] Mechanical amplifiers Mechanical amplifiers were used in the pre-electronic era in specialized applications. Early autopilot units designed by Elmer Ambrose Sperry incorporated a mechanical amplifier using belts wrapped around rotating drums; a slight increase in the tension of the belt caused the drum to move the belt. A paired, opposing set of such drives made up a single amplifier. This amplified small gyro errors into signals large enough to move aircraft control surfaces. A similar mechanism was used in the Vannevar Bush differential analyzer. The electrostatic drum amplifier used a band wrapped partway around a rotating drum, and fixed at its anchored end to a spring. The other end connected to a speaker cone. The input signal was transformed up to high voltage, and added to a high voltage dc supply line. This voltage was connected between drum and belt. Thus the input signal varied the electric field between belt and drum, and thus the friction between them, and thus the amount of lateral movement of the belt and thus speaker cone. Other variations on the theme also existed at one time. [edit] Optical amplifiers Main article: Optical amplifier Optical amplifiers amplify light through the process of stimulated emission. See Laser and Maser. [edit] Miscellaneous types There are also mechanical amplifiers, such as the automotive servo used in braking. Relays can be included under the above definition of amplifiers, although their transfer function is not linear (that is, they are either open or closed). Also purely mechanical manifestations of such digital amplifiers can be built (for theoretical, instructional purposes, or for entertainment), see e.g. domino computer. Another type of amplifier is the fluidic amplifier, based on the fluidic triode Loudspeaker loudspeaker (or "speaker") is an electroacoustic transducer that produces sound in response to an electrical audio signal input. Non-electrical loudspeakers were developed as accessories to telephone systems, but electronic amplification by vacuum tube made loudspeakers more generally useful. The most common form of loudspeaker uses a paper cone supporting a voice coil electromagnet acting on a permanent magnet, but many other types exist. Where accurate reproduction of sound is required, multiple loudspeakers may be used, each reproducing a part of the audible frequency range. Miniature loudspeakers are found in devices such as radio and TV receivers, and many forms of music players. Larger loudspeaker systems are used for music, sound reinforcement in theatres and concerts, and in public address systems. Audio Mixer In professional audio, a mixing console, or audio mixer, also called a sound board, mixing desk, or mixer is an electronic device for combining (also called "mixing"), routing, and changing the level, timbre and/or dynamics of audio signals. A mixer can mix analog or digital signals, depending on the type of mixer. The modified signals (voltages or digital samples) are summed to produce the combined output signals. Mixing consoles are used in many applications, including recording studios, public address systems, sound reinforcement systems, broadcasting, television, and film post-production. An example of a simple application would be to enable the signals that originated from two separate microphones (each being used by vocalists singing a duet, perhaps) to be heard through one set of speakers simultaneously. When used for live performances, the signal produced by the mixer will usually be sent directly to an amplifier, unless that particular mixer is "powered" or it is being connected to powered speakers. Structure Yamaha 2403 audio mixing console in a 'live' mixing application A typical analog mixing board has three sections: Channel inputs Master controls Audio level metering The channel input strips are usually a bank of identical monaural or stereo input channels. The master control section has sub-group faders, master faders, master auxiliary mixing bus level controls and auxiliary return level controls. In addition it may have solo monitoring controls, a stage talk-back microphone control, muting controls and an output matrix mixer. On smaller mixers the inputs are on the left of the mixing board and the master controls are on the right. In larger mixers, the master controls are in the center with inputs on both sides. The audio level meters may be above the input and master sections or they may be integrated into the input and master sections themselves Digital Audio Digital audio is sound reproduction using pulse-code modulation and digital signals. Digital audio systems include analog-to-digital conversion (ADC), digital-to-analog conversion (DAC), digital storage, processing and transmission components. A primary benefit of digital audio is in its convenience of storage, transmission and retrieval. Overview of digital audio Sampling and 4-bit quantization of an analog signal (red) using Pulse-code modulation Digital audio has emerged because of its usefulness in the recording, manipulation, massproduction, and distribution of sound. Modern distribution of music across the Internet via online stores depends on digital recording and digital compression algorithms. Distribution of audio as data files rather than as physical objects has significantly reduced the cost of distribution. In an analog audio system, sounds begin as physical waveforms in the air, are transformed into an electrical representation of the waveform, via a transducer (for example, a microphone), and are stored or transmitted. To be re-created into sound, the process is reversed, through amplification and then conversion back into physical waveforms via a loudspeaker. Although its nature may change, analog audio's fundamental wave-like characteristics remain the same during its storage, transformation, duplication, and amplification. Analog audio signals are susceptible to noise and distortion, unavoidable due to the innate characteristics of electronic circuits and associated devices. In the case of purely analog recording and reproduction, numerous opportunities for the introduction of noise and distortion exist throughout the entire process. When audio is digitized, distortion and noise are introduced only by the stages that precede conversion to digital format, and by the stages that follow conversion back to analog. The digital audio chain begins when an analog audio signal is first sampled, and then (for pulsecode modulation, the usual form of digital audio) it is converted into binary signals—‘on/off’ pulses—which are stored as binary electronic, magnetic, or optical signals, rather than as continuous time, continuous level electronic or electromechanical signals. This signal may then be further encoded to allow correction of any errors that might occur in the storage or transmission of the signal, however this encoding is for error correction, and is not strictly part of the digital audio process. This "channel coding" is essential to the ability of broadcast or recorded digital system to avoid loss of bit accuracy. The discrete time and level of the binary signal allow a decoder to recreate the analog signal upon replay. An example of a channel code is Eight to Fourteen Bit Modulation as used in the audio Compact Disc (CD). [edit] Conversion process The lifecycle of sound from its source, through an ADC, digital processing, a DAC, and finally as sound again. A digital audio system starts with an ADC that converts an analog signal to a digital signal. [note 1] The ADC runs at a sampling rate and converts at a known bit resolution. For example, CD audio has a sampling rate of 44.1 kHz (44,100 samples per second) and 16-bit resolution for each channel. For stereo there are two channels: 'left' and 'right'. If the analog signal is not already bandlimited then an anti-aliasing filter is necessary before conversion, to prevent aliasing in the digital signal. (Aliasing occurs when frequencies above the Nyquist frequency have not been band limited, and instead appear as audible artifacts in the lower frequencies). The digital audio signal may be stored or transmitted. Digital audio storage can be on a CD, a digital audio player, a hard drive, USB flash drive, CompactFlash, or any other digital data storage device. The digital signal may then be altered in a process which is called digital signal processing where it may be filtered or have effects applied. Audio data compression techniques — such as MP3, Advanced Audio Coding, Ogg Vorbis, or FLAC — are commonly employed to reduce the file size. Digital audio can be streamed to other devices. The last step is for digital audio to be converted back to an analog signal with a DAC. Like ADCs, DACs run at a specific sampling rate and bit resolution but through the processes of oversampling, upsampling, and downsampling, this sampling rate may not be the same as the initial sampling rate. [edit] History of digital audio use in commercial recording Pulse-code modulation was invented by British scientist Alec Reeves in 1937[1] and was used in telecommunications applications long before its first use in commercial broadcast and recording. Commercial digital recording was pioneered in Japan by NHK, and Nippon Columbia (a.k.a. Denon) in the 1960s. The first commercial digital recordings were released in 1971.[2] The BBC also began experimenting with digital audio in the 1960s. By the early 1970s they had developed a 2-channel recorder and in 1972 they deployed a digital audio transmission system linking their broadcast center to their remote transmitters.[2] The first 16-bit PCM recording in the United States was made by Thomas Stockham at the Santa Fe Opera in 1976 on a Soundstream recorder. In 1978, an improved version of the Soundstream system was used by Telarc to produce several classical recordings. At the same time 3M was well along in development of their digital multitrack recorder based on BBC technology. The first all-digital album recorded on this machine was Ry Cooder's "Bop 'Til You Drop" which was released in 1979. In a crash program started in 1978, British record label Decca developed their own 2-track digital audio recorders. Decca released the first European digital recording in 1979.[2] Helped along by introduction of popular digital multitrack recorders from Sony and Mitsubishi in the early 1980s, digital recording was soon embraced by the major record companies. With the introduction of the CD by Sony and Philips in 1982, digital audio was embraced by consumers as well.[2] [edit] Digital audio technologies Digital audio broadcasting Digital Audio Broadcasting (DAB) HD Radio Digital Radio Mondiale (DRM) In-band on-channel (IBOC) Storage technologies: Digital audio player Digital Audio Tape (DAT) Compact Disc (CD) Hard disk recorder DVD Audio MiniDisc Super Audio CD Various audio file formats Synthesizers A synthesizer (often abbreviated "synth") is an electronic instrument capable of producing sounds by generating electrical signals of different frequencies. These electrical signals are played through a loudspeaker or set of headphones. Synthesizers can usually produce a wide range of sounds, which may either imitate other instruments ("imitative synthesis") or generate new timbres. Synthesizers use a number of different technologies or programmed algorithms, each with their own strengths and weaknesses. Among the most popular waveform synthesis techniques are subtractive synthesis, additive synthesis, wavetable synthesis, frequency modulation synthesis, phase distortion synthesis, physical modeling synthesis and sample-based synthesis. Other sound synthesis methods, like subharmonic synthesis or granular synthesis, are not commonly found in hardware music synthesizers. Synthesizers are often controlled with a piano-style keyboard, leading such instruments to be referred to simply as "keyboards". Several other forms of controller have been devised to resemble violins, guitars (see guitar synthesizer) and wind-instruments. Synthesizers without controllers are often called "modules", and they can be controlled using MIDI or CV/Gate methods MIDI MIDI ( /ˈmɪdi/; Musical Instrument Digital Interface) is an industry-standard protocol that enables electronic musical instruments (synthesizers, drum machines), computers and other electronic equipment (MIDI controllers, sound cards, samplers) to communicate and synchronize with each other. Unlike analog devices, MIDI does not transmit an audio signal: it sends event messages about musical notation, pitch and intensity, control signals for parameters such as volume, vibrato and panning, cues, and clock signals to set the tempo. As an electronic protocol, it is notable for its widespread adoption throughout the music industry. MIDI protocol was defined in 1982.[1] All MIDI-compatible controllers, musical instruments, and MIDI-compatible software follow the same MIDI 1.0 specification, and thus interpret any given MIDI message the same way, and so can communicate with and understand each other. MIDI composition and arrangement takes advantage of MIDI 1.0 and General MIDI (GM) technology to allow musical data files to be shared among many different devices due to some incompatibility with various electronic instruments by using a standard, portable set of commands and parameters. Because the music is stored as instructions rather than recorded audio waveforms, the data size of the files is quite small by comparison. Basics of Staff Notation To better understand how to read music, maybe it is best to first ask ourselves: What is music exactly? Well, according to the 1976 edition (okay so I need to update my book collection!) of Funk & Wagnalls Standard Desk Dictionary the definition is: mu.sic (myoo'zik) n. 1. The art of producing significant arrangements of sounds, usually with reference to rhythm, pitch and tone colour. 3. A succession or combination of notes, especially if pleasing to the ear. Man!, don't you just hate it when you look up a definition and you need to look up words the definition uses? Well, I'll try to save you the trouble this time. pitch is the frequency at which a note vibrates, I'll explain this shortly. Tone colour is the type of sound, for example an overdriven electric guitar has a very rough aggressive tone while a flute usually has a soft mellow tone (unless the flute player really sucks I suppose). Rhythm is a measure of the the time frame you play the notes in, but I will explain that later too. For now, let's just say that music is the art of producing significant arrangements of sounds, usually for the purpose of causing emotional responses in people (usually, you want people to like what they hear unless of course you are trying to be the latest punk band and want people to be offended by your sound! To each his own I guess...). Okay, now back to what we set out to do in the first place, teach you how to read music... Sound and Pitch in Music Now that we've established that music is made up of sounds I will explain what a sound actually is: All sounds are caused by the vibrations of air molecules. These waves ("sound waves") of vibrations in air molecules originate from some kind of vibrating object, perhaps a musical instrument or a person's vocal chords. In music we refer to the frequency (how many times the molecules vibrate per second) a note vibrates at as the pitch of the note. In most contemporary sheet music you will see the music will be written on either the treble clef staff: Or the bass clef staff: As the notes are written closer to the top of these clefs there pitch increases giving them a higher, lighter sound. Conversely, as notes are written closer to the bottom of the clefs the pitch decreases giving them a lower, darker sound. The treble clef contains notes that are higher in pitch than the bass clef and the bass clef contains notes that are lower in pitch than the treble clef. For this reason for some instruments that have a wide range of notes, the piano in particular, you may see these two staffs combined as follows: The next image may help you visualize how notes are placed on the staffs in relation to their pitch. It is a picture of a piano keyboard with the clefs and notes written over top: Click here to listen to sound of notes in picture from left to right Notice that as you go from the lower pitch notes on the left of the piano to the higher pitch notes on the right side of the piano the notes are written on the staffs in ascending order. As you can see from the diagram above we sometimes write notes that are below or above the lines on the staff, these notes appear on extra small lines called ledger lines. You may also notice that there is one note (middle C) which can be written as either one ledger line above the bass clef or as one ledger line below the treble clef. The diagram above shows all of the white notes on the piano written on the staffs, but you are probably wondering about the black notes, how are they written? Well, this can be answered by viewing the diagram below: Click here to compare regular and accidental notes in picture (Note: The rhythm in the sound file is slightly different to than the rhythm shown in the picture.) In music there are notes that we sometimes come across called "Accidentals". So what exactly are these accidentals, you may be asking, the notes I accidentally play by mistake? No, although some musicians might try to use that as an excuse, accidentals are actually notes that are called for you to play in a piece of music which are not in the general key that most of the song is written in. When you encounter a note in music that has a to the left of it you play the note immediately left of it on the keyboard. If you encounter a note that has a immediately to the right of it on the keyboard. in front of it you play the note Rhythm and Note Durations There are many different durations of notes, typically you will see the following basic note durations in today's contemporary music: Whole Note Half Note Quarter Note Eigth Note Sixteenth Note The majority of the contemporary rock and pop music you hear on the radio these days is written in the 4/4 time signature: The top number tells us how many of the specified notes are in a bar and the bottom number tells us what duration (ie: how long) that specified note is. For example in 4/4 Time the top number tells us there are 4 notes in a bar and the bottom number tells us that each note is 1/4 of the length of the bar, or more simply put a quarter note. Therefore, we can tell that a song written with a 4/4 time signature is made up of bars (musical units a song is divided up into) which contain 4 quarter note long beats. The following picture may help in visualizing this: Click here to listen (Note: I have added a drum click to empasize the beat and will also do so in some later examples. The drum will be played on every beat with an accent on beat one.) Sound Card A sound card (also known as an audio card) is an internal computer expansion card that facilitates the input and output of audio signals to and from a computer under control of computer programs. The term sound card is also applied to external audio interfaces that use software to generate sound, as opposed to using hardware inside the PC. Typical uses of sound cards include providing the audio component for multimedia applications such as music composition, editing video or audio, presentation, education and entertainment (games) and video projection. Many computers have sound capabilities built in, while others require additional expansion cards to provide for audio capability. [edit] Sound channels and polyphony 8-channel DAC Cirrus Logic CS4382 placed on Sound Blaster X-Fi Fatal1ty. An important sound card characteristic is polyphony, which refers to its ability to process and output multiple independent voices or sounds simultaneously. These distinct channels are seen as the number of audio outputs, which may correspond to a speaker configuration such as 2.0 (stereo), 2.1 (stereo and sub woofer), 5.1 (surround), or other configuration. Sometimes, the terms voice and channel are used interchangeably to indicate the degree of polyphony, not the output speaker configuration. For example, many older sound chips could accommodate three voices, but only one audio channel (i.e., a single mono output) for output, requiring all voices to be mixed together. Later cards, such as the AdLib sound card, had a 9-voice polyphony combined in 1 mono output channel. For some years, most PC sound cards have had multiple FM synthesis voices (typically 9 or 16) which were usually used for MIDI music. The full capabilities of advanced cards aren't often completely used; only one (mono) or two (stereo) voice(s) and channel(s) are usually dedicated to playback of digital sound samples, and playing back more than one digital sound sample usually requires a software downmix at a fixed sampling rate. Modern low-cost integrated soundcards (i.e., those built into motherboards) such as audio codecs like those meeting the AC'97 standard and even some lower-cost expansion sound cards still work this way. These devices may provide more than two sound output channels (typically 5.1 or 7.1 surround sound), but they usually have no actual hardware polyphony for either sound effects or MIDI reproduction – these tasks are performed entirely in software. This is similar to the way inexpensive softmodems perform modem tasks in software rather than in hardware. Also, in the early days of wavetable synthesis, some sound card manufacturers advertised polyphony solely on the MIDI capabilities alone. In this case, the card's output channel is irrelevant (and typically, the card is only capable of two channels of digital sound). Instead, the polyphony measurement solely applies to the amount of MIDI instruments the sound card is capable of producing at one given time. Today, a sound card providing actual hardware polyphony, regardless of the number of output channels, is typically referred to as a "hardware audio accelerator", although actual voice polyphony is not the sole (or even a necessary) prerequisite, with other aspects such as hardware acceleration of 3D sound, positional audio and real-time DSP effects being more important. Since digital sound playback has become available and provided better performance than synthesis, modern soundcards with hardware polyphony don't actually use DACs with as many channels as voices. Instead, they perform voice mixing and effects processing in hardware (eventually performing digital filtering and conversions to and from the frequency domain for applying certain effects) inside a dedicated DSP. The final playback stage is performed by an external (in reference to the DSP chip(s)) DAC with significantly fewer channels than voices (e.g., 8 channels for 7.1 audio, which can be divided among 32, 64 or even 128 voices). [edit] Color codes Connectors on the sound cards are colour coded as per the PC System Design Guide. They will also have symbols with arrows, holes and soundwaves that are associated with each jack position, the meaning of each is given below: Colour Pink Function Analog microphone audio input. Light blue Analog line level audio input. Connector 3.5 mm TRS A microphone 3.5 mm TRS An arrow going into a circle Arrow going out one side of a circle into a wave Lime green Analog line level audio output for the main stereo signal (front speakers or headphones). 3.5 mm TRS Brown/Dark Analog line level audio output for a special panning,'Right-to-left speaker'. 3.5 mm TRS Black Analog line level audio output for surround speakers, typically rear stereo. 3.5 mm TRS Orange symbol Analog line level audio output for center channel 3.5 mm speaker and subwoofer TRS Gold/Grey Game port / MIDI 15 pin D Arrow going out both sides into waves [edit] History of sound cards for the IBM PC architecture The AdLib Music Synthesizer Card, was one of the first sound cards circa 1990. Note the manual volume adjustment knob. ISA-8 bus. Sound card Mozart 16 for ISA-16 bus. A Turtle Beach sound card. PCI bus. Echo Digital Audio's Indigo IO — PCMCIA card 24-bit 96 kHz stereo in/out sound card. Sound cards for computers compatible with the IBM PC were very uncommon until 1988, which left the single internal PC speaker as the only way early PC software could produce sound and music. The speaker hardware was typically limited to square waves, which fit the common nickname of "beeper". The resulting sound was generally described as "beeps and boops". Several companies, most notably Access Software, developed techniques for digital sound reproduction over the PC speaker; the resulting audio, while baldly functional, suffered from distorted output and low volume, and usually required all other processing to be stopped while sounds were played. Other home computer models of the 1980s included hardware support for digital sound playback, or music synthesis (or both), leaving the IBM PC at a disadvantage to them when it came to multimedia applications such as music composition or gaming. It is important to note that the initial design and marketing focuses of sound cards for the IBM PC platform were not based on gaming, but rather on specific audio applications such as music composition (AdLib Personal Music System, Creative Music System, IBM Music Feature Card) or on speech synthesis (Digispeech DS201, Covox Speech Thing, Street Electronics Echo). Not until Sierra and other game companies became involved in 1988 was there a switch toward gaming. [edit] Hardware manufacturers One of the first manufacturers of sound cards for the IBM PC was AdLib, who produced a card based on the Yamaha YM3812 sound chip, also known as the OPL2. The AdLib had two modes: A 9-voice mode where each voice could be fully programmed, and a less frequently used "percussion" mode with 3 regular voices producing 5 independent percussion-only voices for a total of 11. (The percussion mode was considered inflexible by most developers; it was used mostly by AdLib's own composition software.) Creative Labs also marketed a sound card about the same time called the Creative Music System. Although the C/MS had twelve voices to AdLib's nine, and was a stereo card while the AdLib was mono, the basic technology behind it was based on the Philips SAA 1099 chip which was essentially a square-wave generator. It sounded much like twelve simultaneous PC speakers would have, and failed to sell well, even after Creative renamed it the Game Blaster a year later, and marketed it through Radio Shack in the US. The Game Blaster retailed for under $100 and included the hit game Silpheed. A large change in the IBM PC compatible sound card market happened with Creative Labs' introduced the Sound Blaster card. The Sound Blaster cloned the AdLib, and added a sound coprocessor for recording and play back of digital audio (likely to have been an Intel microcontroller relabeled by Creative). It was incorrectly called a "DSP" (to suggest it was a digital signal processor), a game port for adding a joystick, and capability to interface to MIDI equipment (using the game port and a special cable). With more features at nearly the same price, and compatibility as well, most buyers chose the Sound Blaster. It eventually outsold the AdLib and dominated the market. The Sound Blaster line of cards, together with the first inexpensive CD-ROM drives and evolving video technology, ushered in a new era of multimedia computer applications that could play back CD audio, add recorded dialogue to computer games, or even reproduce motion video (albeit at much lower resolutions and quality in early days). The widespread decision to support the Sound Blaster design in multimedia and entertainment titles meant that future sound cards such as Media Vision's Pro Audio Spectrum and the Gravis Ultrasound had to be Sound Blaster compatible if they were to sell well. Until the early 2000s (by which the AC'97 audio standard became more widespread and eventually usurped the SoundBlaster as a standard due to its low cost and integration into many motherboards), Sound Blaster compatibility is a standard that many other sound cards still support to maintain compatibility with many games and applications released. [edit] Industry adoption When game company Sierra On-Line opted to support add-on music hardware (instead of builtin hardware such as the PC speaker and built-in sound capabilities of the IBM PCjr and Tandy 1000), what could be done with sound and music on the IBM PC changed dramatically. Two of the companies Sierra partnered with were Roland and Adlib, opting to produce in-game music for King's Quest 4 that supported the Roland MT-32 and Adlib Music Synthesizer. The MT-32 had superior output quality, due in part to its method of sound synthesis as well as built-in reverb. Since it was the most sophisticated synthesizer they supported, Sierra chose to use most of the MT-32's custom features and unconventional instrument patches, producing background sound effects (e.g., chirping birds, clopping horse hooves, etc.) before the Sound Blaster brought playing real audio clips to the PC entertainment world. Many game companies also supported the MT-32, but supported the Adlib card as an alternative because of the latter's higher market base. The adoption of the MT-32 led the way for the creation of the MPU-401/Roland Sound Canvas and General MIDI standards as the most common means of playing in-game music until the mid1990s. [edit] Feature evolution Early ISA bus soundcards were half-duplex, meaning they couldn't record and play digitized sound simultaneously, mostly due to inferior card hardware (e.g., DSPs). Later, ISA cards like the SoundBlaster AWE series and Plug-and-play Soundblaster clones eventually became fullduplex and supported simultaneous recording and playback, but at the expense of using up two IRQ and DMA channels instead of one, making them no different from having two half-duplex sound cards in terms of configuration. Towards the end of the ISA bus' life, ISA soundcards started taking advantage of IRQ sharing, thus reducing the IRQs needed to one, but still needed two DMA channels. Many PCI bus cards do not have these limitations and are mostly fullduplex. It should also be noted that many modern PCI bus cards also do not require free DMA channels to operate. Also, throughout the years, soundcards have evolved in terms of digital audio sampling rate (starting from 8-bit 11.025 kHz, to 32-bit, 192 kHz that the latest solutions support). Along the way, some cards started offering wavetable synthesis, which provides superior MIDI synthesis quality relative to the earlier OPL-based solutions, which uses FM-synthesis. Also, some higher end cards started having their own RAM and processor for user-definable sound samples and MIDI instruments as well as to offload audio processing from the CPU. For years, soundcards had only one or two channels of digital sound (most notably the Sound Blaster series and their compatibles) with the exception of the E-MU card family, which had hardware support for up to 32 independent channels of digital audio. Early games and MODplayers needing more channels than a card could support had to resort to mixing multiple channels in software. Even today, the tendency is still to mix multiple sound streams in software, except in products specifically intended for gamers or professional musicians, with a sensible difference in price from "software based" products. Also, in the early era of wavetable synthesis, soundcard companies would also sometimes boast about the card's polyphony capabilities in terms of MIDI synthesis. In this case polyphony solely refers to the amount of MIDI notes the card is capable of synthesizing simultaneously at one given time and not the amount of digital audio streams the card is capable of handling. In regards to physical sound output, the number of physical sound channels has also increased. The first soundcard solutions were mono. Stereo sound was introduced in the early 90s, and quadraphonic sound came in 1989. This was shortly followed by 5.1 channel audio. The latest soundcards support up to 8 physical audio channels in the 7.1 speaker setup. Audio File formats and CODECs An audio file format is a file format for storing digital audio data on a computer system. This data can be stored uncompressed, or compressed to reduce the file size. It can be a raw bitstream, but it is usually a container format or an audio data format with defined storage layer Types of formats It is important to distinguish between a file format and an audio codec. A codec performs the encoding and decoding of the raw audio data while the data itself is stored in a file with a specific audio file format. Most of the publicly documented audio file formats can be created with one of two or more encoders or codecs.[citation needed] Although most audio file formats support only one type of audio data (created with an audio coder), a multimedia container format (as Matroska or AVI) may support multiple types of audio and video data. There are three major groups of audio file formats: Uncompressed audio formats, such as WAV, AIFF, AU or raw header-less PCM; Formats with lossless compression, such as FLAC, Monkey's Audio (filename extension APE), WavPack (filename extension WV), TTA, ATRAC Advanced Lossless, Apple Lossless (filename extension m4a), MPEG-4 SLS, MPEG-4 ALS, MPEG-4 DST, Windows Media Audio Lossless (WMA Lossless), and Shorten (SHN). Formats with lossy compression, such as MP3, Vorbis, Musepack, AAC, ATRAC and Windows Media Audio Lossy (WMA lossy)). [edit] Uncompressed audio formats There is one major uncompressed audio format, PCM, which is usually stored in a .wav file on Windows or in a .aiff file on Mac OS. The AIFF format is based on the Interchange File Format (IFF). The WAV format is based on the Resource Interchange File Format (RIFF), which is similar to IFF. WAV and AIFF are flexible file formats designed to store more or less any combination of sampling rates or bitrates. This makes them suitable file formats for storing and archiving an original recording. BWF (Broadcast Wave Format) is a standard audio format created by the European Broadcasting Union as a successor to WAV. BWF allows metadata to be stored in the file. See European Broadcasting Union: Specification of the Broadcast Wave Format (EBU Technical document 3285, July 1997). This is the primary recording format used in many professional audio workstations in the television and film industry. BWF files include a standardized timestamp reference which allows for easy synchronization with a separate picture element. Stand-alone, file based, multi-track recorders from Sound Devices,[1] Zaxcom,[2] HHB USA,[3] Fostex, and Aaton[4] all use BWF as their preferred format. The .cda (Compact Disk Audio Track) is a small file that serves as a shortcut to the audio data for a track on a music CD. It does not contain audio data and is therefore not considered to be a proper audio file format. [edit] Lossless compressed audio formats A lossless compressed format stores data in less space by eliminating unnecessary data. It requires more processing power both to compress the data and to uncompress for playback.[citation needed] Uncompressed audio formats encode both sound and silence with the same number of bits per unit of time. Encoding an uncompressed minute of absolute silence produces a file of the same size as encoding an uncompressed minute of music. In a lossless compressed format, however, the music would occupy a smaller portion of the file and the silence would take up almost no space at all. Lossless compression formats enable the original uncompressed data to be recreated exactly. They include the common[5] FLAC, WavPack, Monkey's Audio, ALAC (Apple Lossless). They provide a compression ratio of about 2:1 (i.e. their files take up half the space of the originals). Development in lossless compression formats aims to reduce processing time while maintaining a good compression ratio. [edit] Lossy compressed audio formats Lossy compression enables even greater reductions in file size by removing some of the data. A variety of techniques are used, mainly by exploiting psychoacoustics, to remove data with minimal reduction in the quality of reproduction. For many everyday listening situations, the loss in data (and thus quality) is imperceptible. The popular MP3 format is probably the best-known example, but AAC format is another common one. Most formats offer a range of degrees of compression, generally measured in bit rate. The lower the rate, the smaller the file and the greater the quality loss. Audio Recording Systems Sound recording and reproduction is an electrical or mechanical inscription and re-creation of sound waves, such as spoken voice, singing, instrumental music, or sound effects. The two main classes of sound recording technology are analog recording and digital recording. Acoustic analog recording is achieved by a small microphone diaphragm that can detect changes in atmospheric pressure (acoustic sound waves) and record them as a graphic representation of the sound waves on a medium such as a phonograph (in which a stylus senses grooves on a record). In magnetic tape recording, the sound waves vibrate the microphone diaphragm and are converted into a varying electric current, which is then converted to a varying magnetic field by an electromagnet, which makes a representation of the sound as magnetized areas on a plastic tape with a magnetic coating on it. Analog sound reproduction is the reverse process, with a bigger loudspeaker diaphragm causing changes to atmospheric pressure to form acoustic sound waves. Electronically generated sound waves may also be recorded directly from devices such as an electric guitar pickup or a synthesizer, without the use of acoustics in the recording process other than the need for musicians to hear how well they are playing during recording sessions. Digital recording and reproduction converts the analog sound signal picked up by the microphone to a digital form by a process of digitization, allowing it to be stored and transmitted by a wider variety of media. Digital recording stores audio as a series of binary numbers representing samples of the amplitude of the audio signal at equal time intervals, at a sample rate high enough to convey all sounds capable of being heard. Digital recordings are considered higher quality than analog recordings not necessarily because they have higher fidelity (wider frequency response or dynamic range), but because the digital format can prevent much loss of quality found in analog recording due to noise and electromagnetic interference in playback, and mechanical deterioration or damage to the storage medium. A digital audio signal must be reconverted to analog form during playback before it is applied to a loudspeaker or earphones. Audio and Multimedia Multimedia is media and content that uses a combination of different content forms. The term can be used as a noun (a medium with multiple content forms) or as an adjective describing a medium as having multiple content forms. The term is used in contrast to media which use only rudimentary computer display such as text-only, or traditional forms of printed or hand-produced material. Multimedia includes a combination of text, audio, still images, animation, video, or interactivity content forms. Examples of individual content forms combined in multimedia: Text Audio Still Images Animation Video Footage Interactivity Multimedia is usually recorded and played, displayed or accessed by information content processing devices, such as computerized and electronic devices, but can also be part of a live performance. Multimedia (as an adjective) also describes electronic media devices used to store and experience multimedia content. Multimedia is distinguished from mixed media in fine art; by including audio, for example, it has a broader scope. The term "rich media" is synonymous for interactive multimedia. Hypermedia can be considered one particular multimedia application. Voice Recognition and Response Interactive voice response (IVR) is a technology that allows a computer to interact with humans through the use of voice and DTMF keypad inputs. In telecommunications, IVR allows customers to interact with a company’s database via a telephone keypad or by speech recognition, after which they can service their own inquiries by following the IVR dialogue. IVR systems can respond with prerecorded or dynamically generated audio to further direct users on how to proceed. IVR applications can be used to control almost any function where the interface can be broken down into a series of simple interactions. IVR systems deployed in the network are sized to handle large call volumes. IVR technology is also being introduced into automobile systems for hands-free operation. Current deployment in automobiles revolves around satellite navigation, audio and mobile phone systems. It has become common in industries that have recently entered the telecommunications industry to refer to an automated attendant as an IVR. The terms, however, are distinct and mean different things to traditional telecommunications professionals, whereas emerging telephony and VoIP professionals often use the term IVR as a catch-all to signify any kind of telephony menu, even a basic automated attendant.[citation needed] The term voice response unit (VRU), is sometimes used as well Audio Processing Software. Typical Audio Editing Applications Software audio editing for studios and professional journalists. Edit sound files to broadcast over the internet with the BroadWave Streaming Audio Server Normalizing the level of audio files during mastering before burning to CD. Editing mp3 files for your iPod, PSP or other portable device. As a music editor (includes ringtones creator formats). Music editing and recording to produce mp3 files. Voice editing for multimedia productions (use with our Video Editor). Restoration of audio files including removing excess noise such as hiss and hums.