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Sound

|Sound measurements |

|Sound pressure p |

|Sound pressure level (SPL) |

|Particle velocity v |

|Particle velocity level (SVL) |

|   (Sound velocity level) |

|Particle displacement ξ |

|Sound intensity I |

|Sound intensity level (SIL) |

|Sound power Pac |

|Sound power level (SWL) |

|Sound energy density E |

|Sound energy flux q |

|Acoustic impedance Z |

|Speed of sound c |

Sound is what can be perceived by a living organism through its sense of hearing.[1] Physically, sound is vibrational mechanical energy that propagates through matter as a wave.

For humans, hearing is limited to frequencies between about 20 Hz and 20000 Hz, with the upper limit generally decreasing with age. Other species may have a different range of hearing.[2] As a signal perceived by one of the major senses, sound is used by many species for detecting danger, navigation, predation, and communication. In Earth's atmosphere, water, and soil 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 became highly evolved to produce song and (in humans) speech. Furthermore, humans have developed culture and technology (such as music, telephony and radio) that allows them to generate, record, transmit, and broadcast sounds.

The mechanical vibrations that can be interpreted as sound can travel through all forms of matter: gases, liquids, solids, and plasmas. However, sound cannot propagate through vacuum. The matter that supports the sound is called the medium. 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 and transverse waves. Sound is further characterized by the generic properties of waves, which are frequency, wavelength, period, amplitude, intensity, speed, and direction (sometimes speed and direction are combined as a velocity vector, or wavelength and direction are combined as a wave vector). Transverse waves, also known as shear waves, have an additional property of polarization. Sound characteristics can depend on the type of sound waves (longitudinal versus transverse) as well as on the physical properties of the transmission medium.

Sound propagates as waves of alternating pressure deviations from the equilibrium pressure (or, for transverse waves in solids, as waves of alternating shear stress), causing local regions of compression and rarefaction. Matter in the medium is periodically displaced by the wave, and thus oscillates. The energy carried by the sound wave is split equally between the potential energy of the extra compression of the matter and the kinetic energy of the oscillations of the medium. The scientific study of the propagation, absorption, and reflection of sound waves is called acoustics.

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.

Speed of sound

Main article: Speed of sound

The speed of sound depends on the medium through which the waves are passing, and is often quoted as 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 air at sea level, the speed of sound is approximately 343 m/s, in water 1482 m/s (both at 20 °C, or 68 °F), and in steel about 5960 m/s.[3] The speed of sound is also slightly sensitive (a second-order 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).

Sound pressure level

Main article: Sound pressure

Sound pressure is defined as the difference between the actual pressure (at a given point and a given time) in the medium and the average, or equilibrium, pressure of the medium at that location. 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 such average is taken to obtain a root mean square (RMS) value. For example, 1 Pa RMS sound pressure in atmospheric air implies that the actual pressure in the sound wave oscillates between (1 atm [pic]Pa) and (1 atm [pic]Pa), that is between 101323.6 and 101326.4 Pa. Such a tiny (relative to atmospheric) variation in air pressure at an audio frequency will be perceived as quite a deafening sound, and can cause hearing damage, according to the table below.

As the human ear can detect sounds with a very 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

[pic]

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.

Since the human ear does not have a flat spectral response, sound pressures are often frequency weighted so that the measured level will match 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 A-weighted sound pressure levels are labeled dBA. C-weighting is used to measure peak levels.

Examples of sound pressure and sound pressure levels

|Source of sound |RMS sound pressure |sound pressure level |

|  |Pa |dB re 20 µPa |

|immediate soft tissue damage |50000 |approx. 185 |

|rocket launch equipment acoustic tests | |approx. 165 |

|threshold of pain |100 |134 |

|hearing damage during short-term effect |20 |approx. 120 |

|jet engine, 100 m distant |6–200 |110–140 |

|jack hammer, 1 m distant / discotheque |2 |approx. 100 |

|hearing damage from long-term exposure |0.6 |approx. 85 |

|traffic noise on major road, 10 m distant |0.2–0.6 |80–90 |

|moving passenger car, 10 m distant |0.02–0.2 |60–80 |

|TV set – typical home level, 1 m distant |0.02 |approx. 60 |

|normal talking, 1 m distant |0.002–0.02 |40–60 |

|very calm room |0.0002–0.0006 |20–30 |

|quiet rustling leaves, calm human breathing |0.00006 |10 |

|auditory threshold at 2 kHz – undamaged human ears |0.00002 |0 |

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.

Frequency

Frequency is a measure of the number of occurrences of a repeating event per unit time. It is also referred to as temporal frequency.

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| |

Definition and units

For cyclical processes, such as rotation, oscillations, or waves, it is defined as a number of cycles, or periods, per unit time. In physics and engineering disciplines, such as optics, acoustics, and radio, frequency is usually denoted by a Latin letter f or by a Greek letter ν (nu).

The number of wavelengths per second of a particular radiation.

In SI system, the unit of frequency is hertz (Hz), named after the German physicist Heinrich Hertz. For example, 1 Hz means that an event repeats once per second, 2 Hz is twice per second, and so on [1]. This unit was originally called a cycle per second (cps), which is still sometimes used. Heart rate and musical tempo are measured in beats per minute (BPM). Frequency of rotation is often expressed as a number of revolutions per minute (rpm). BPM and rpm values must be divided by 60 to obtain the corresponding value in Hz: thus, 60 BPM translates into 1 Hz.

A related measure of frequency, called angular frequency ω, is often introduced. It is defined as the rate of change in the orientation angle (during rotation), or in the phase of a sinusoidal waveform (e.g. in oscillations and waves): ω = 2πf. Angular frequency is measured in radians per second (s-1).

Measurement

By counting

To calculate the frequency of the event, the number of occurrences of the event within a fixed time interval are counted, and then divided by the length of the time interval.

To calculate the frequency of an event in experimental work however (for example, calculating the frequency of an oscillating pendulum) it is crucial that the time taken for a fixed number of occurrences is recorded, rather than the number of occurrences within a fixed time. This is because your random error is significantly increased performing the experiment the other way around. It [the frequency] is still calculated by dividing the number of occurrences by the time interval, however, the number of occurrences is fixed, not the time interval.

An alternative method to calculate frequency is to measure the time between two consecutive occurrences of the event (the period) and then compute the frequency f as the reciprocal of this time:

[pic]

where

T= Period

A more accurate measurement takes many cycles into account and averages the period between each.

By stroboscope effect, or frequency beats

In case when the frequency is so high that counting is difficult or impossible with the available means, another method is used, based on a source (such as a laser, a tuning fork, or a waveform generator) of a known reference frequency f0, that must be tunable or very close to the measured frequency f. Both the observed frequency and the reference frequency are simultaneously produced, and frequency beats are observed at a much lower frequency Δf, which can be measured by counting. This is sometimes referred to as a stroboscope effect. The unknown frequency is then found from [pic].

Frequency of waves

Frequency has an inverse relationship to the concept of wavelength, simply, frequency is inversely proportional to wavelength λ. The frequency f is equal to the speed v of the wave divided by the wavelength λ (lambda) of the wave:

[pic]

In the special case of electromagnetic waves moving through a vacuum, then v = c, where c is the speed of light in a vacuum, and this expression becomes:

[pic]

When waves travel from one medium to another, their frequency remains exactly the same — only their wavelength and speed change.

Apart from being modified by the Doppler effect or any other nonlinear process, frequency is an invariant quantity in the universe. That is, it cannot be changed by any linearly physical process unlike velocity of propagation or wavelength.

Frequency of sound

Sound is a wave associated with the transmission of mechanical energy through a supporting medium. It can be shown experimentally that sound cannot travel through a vacuum. The energy available in a sound wave disturbs the medium in a periodic manner. Periodicity is important if a sound wave is to carry information. In air, the disturbance propagates as the successive compression and decompression (the latter sometimes called rarefaction) of small regions in the medium. If we generate a pure note and place a detector (our ear, for example) at a point in the surrounding medium, a distance from the source, the number of compression-decompression sequences arriving at the detector during a unit time interval is called the frequency. The time interval between successive maximal compressions is called the period. The product of the frequency and the wavelength is the velocity.

Examples

• In music and acoustics, the frequency of the standard pitch A above middle C on a piano is usually defined as 440 Hz, that is, 440 cycles per second (Listen (help·info)) and known as concert pitch, to which an orchestra tunes.

• A baby can hear tones with oscillations up to approximately 20,000 Hz, but these frequencies become more difficult to hear as people age.

• In Europe, Africa, Australia, Southern South America, most of Asia, and in Russia, the frequency of the alternating current in household electrical outlets is 50 Hz (close to the tone G), however, in North America and Northern South America, the frequency of the alternating current is 60 Hz (between the tones B♭ and B — that is, a minor third above the European frequency). The frequency of the 'hum' in an audio recording can show where the recording was made — in countries utilizing the European, or the American grid frequency.

The seventh octave is the last octave at the top of a piano.Using middle C (C4) as a guide, the next higher C is C5 or tenor C. The next C is C6 or soprano high C. The next C, C7 or double high C, is again one octave higher. C7 is eight notes away from the last note on the 88-key piano: C8. C7 is also the highest note on most musical keyboards. The seventh octave is the range of notes between C7 and C8. Only a small percentage of coloratura sopranos are capable of singing in this octave. While notes in the sixth octave, between soprano high C and C7, can have enough color to sound flutey or canary-like (which give the flageolet register its name), the squeaky, whistly tones in the seventh octave help give the whistle register its name. The piercing qualities of notes in this octave help give the whistle register its name. Examples of pop singers capable of this vocal altitude in performance are Mariah Carey, Minnie Riperton, Tanya Blount, Sebastian Vilas and Adam Lopez.

Octave.

|Perfect octave |

|Inverse |unison |

|Name |

|Other names |- |

|Abbreviation |P8 |

|Size |

|Semitones |12 |

|Interval class |0 |

|Just interval |2:1 |

|Cents |

|Equal temperament |1200 |

|Just intonation |1200 |

In music, an octave (sometimes abbreviated 8ve or P8) is the interval between one musical note and another with half or double its frequency.

Examples

[pic]

An example of an octave, from G4 to G5

For example, if one note has a frequency of 400 Hz, the note an octave above it is at 800 Hz, and the note an octave below is at 200 Hz. The ratio of frequencies of two notes an octave apart is therefore 2:1. Further octaves of a note occur at 2n times the frequency of that note (where n is an integer), such as 2, 4, 8, 16, etc. and the reciprocal of that series. For example, 50 Hz and 400 Hz are one and two octaves away from 100 Hz because they are [pic](or 2 − 1) and 4 (or 22) times the frequency, respectively. However, 300 Hz is not a whole number octave above 100 Hz, despite being a harmonic of 100 Hz.

Musical relevance

After the unison, the octave is the simplest interval in music. The human ear tends to hear both notes as being essentially "the same". For this reason, notes an octave apart are given the same note name in the Western system of music notation—the name of a note an octave above A is also A. This is called octave equivalency, and is closely related to harmonics. This is similar to enharmonic equivalency, and less so transpositional equivalency and, less still, inversional equivalency, the latter of which is generally used only in counterpoint, musical set theory, or atonal theory. Thus all C#s, or all 1s (if C=0), in any octave are part of the same pitch class. Octave equivalency is a part of most musics, but is far from universal in "primitive" and early music (e.g., Nettl, 1956; Sachs & Kunst, 1962). However, monkeys experience octave equivalency, and its biological basis apparently is an octave mapping of neurons in the auditory thalamus of the mammalian brain [1] and the perception of octave equivalency in self-organizing neural networks can form through exposure to pitched notes, without any tutoring, this being derived from the acoustical structure of those notes (Bharucha 2003, cited in Fineberg 2006).

While octaves commonly refer to the perfect octave (P8), the interval of an octave in music theory encompasses chromatic alterations within the pitch class, meaning that G natural to G# (13 semitones higher) is an augmented octave (A8), and G natural to G-flat (11 semitones higher) is a diminished octave (d8). The use of such intervals is rare, as there is frequently a more preferable enharmonic notation available, but these categories of octaves must be acknowledged in any full understanding of the role and meaning of octaves more generally in music.

Electrical relevance

In electronics design, an amplifier or filter may be stated to have a frequency response of ±6dB per octave over a particular frequency range, which signifies that the power gain changes by ±6 decibels (a factor of four in power), or more precisely 6.0206 decibels when the frequency changes by a factor of 2. This response is equivalent to ±20dB per decade (a change in frequency by a factor of 10).

Example

A magnitude of 400 at 4 kH decreases as frequency increases at -2 dB/octave. What is the magnitude at 13 kH. The number of octaves = log base 2 of 13/4. 20 * log(230) / # octaves = -2 db/octave.

Other uses of term

As well as being used to describe the relationship between two notes, the word is also used when speaking of a range of notes that fall between a pair an octave apart. In the diatonic scale, and the other standard heptatonic scales of Western music, this is 8 notes if one counts both ends, hence the name "octave", from the Latin octavus, from octo (meaning "eight"). In the chromatic scale, this is 13 notes counting both ends, although traditionally, one speaks of 12 notes of the chromatic scale, since there are 12 intervals. Other scales may have a different number of notes covering the range of an octave, such as the Arabic classical scale with 17, 19, or even 24 notes, but the word "octave" is still used.

In terms of playing an instrument, "octave" may also mean a special effect involving playing two notes that are an octave apart at the same time. This effect may have to be created by the musician. However, some instruments are purposely tuned or designed to produce this effect, for example, the twelve-string guitar and the octave harmonica.

In most Western music, the octave is divided into 12 semitones (see musical tuning). These semitones are usually equally spaced out in a method known as equal temperament.

Many times singers will be described as having a four-octave range or a five-octave range. This is technically a misnomer, and is described here: five-octave vocal range. It is important to remember when hearing this description that a piano has 7 1/3 octaves total.

Many of the dual toned sirens manufactured by the Sentry Siren Company use an octave ratio on their sirens, usually 16/8, which produces a 2/1 octave.

Notation

[pic]

[pic]

An example of the same two notes expressed regularly, in an 8va bracket, and in a 15ma bracket.

The notation 8va is sometimes seen in sheet music, meaning "play this an octave higher than written." 8va stands for ottava, the Italian word for octave. Sometimes 8va will also be used to indicate a passage is to be played an octave lower, although the similar notation 8vb (ottava bassa) is more common. Similarly, 15ma (quindicesima) means "play two octaves higher than written" and 15mb (quindicesima bassa) means "play two octaves lower than written." Col 8 or c. 8va stands for coll'ottava and means "play the notes in the passage together with the notes in the notated octaves". Any of these directions can be cancelled with the word loco, but often a dashed line or bracket indicates the extent of the music affected.

For music-theoretical purposes (not on sheet music), octave can be abbreviated as P8 (which is an abbreviation for Perfect Eighth, the interval between 12 semitones or an octave).

Amplitude

The amplitude is a nonnegative scalar measure of a wave's magnitude of oscillation, that is, the magnitude of the maximum disturbance in the medium during one wave cycle.

[pic]

The displacement y is the amplitude of the wave

Sometimes this distance is called the peak amplitude, distinguishing it from another concept of amplitude, used especially in electrical engineering: the RMS or root mean square amplitude, defined as the square root of the temporal mean of the square of the vertical distance of this graph from the horizontal axis. The use of peak amplitude is unambiguous for symmetric, periodic waves, like a sine wave, a square wave, or a triangular wave. For an asymmetric wave (periodic pulses in one direction, for example), the peak amplitude becomes ambiguous because the value obtained is different depending on whether the maximum positive signal is measured relative to the mean, the maximum negative signal is measured relative to the mean, or the maximum positive signal is measured relative the maximum negative signal (the peak-to-peak amplitude) and then divided by two.

For complex waveforms, especially non-repeating signals like noise, the RMS amplitude is usually used because it is unambiguous and because it has physical significance. For example, the average power transmitted by an acoustic or electromagnetic wave or by an electrical signal is proportional to the square of the RMS amplitude (and not, in general, to the square of the peak amplitude).

[pic]

[pic]

Different amplitude measurements of a sine wave

There are a few ways to formalize amplitude:

In the simple wave equation

[pic]

A is the amplitude of the wave.

The units of the amplitude depend on the type of wave.

For waves on a string, or in medium such as water, the amplitude is a displacement.

The amplitude of sound waves and audio signals (also referred to as Volume) conventionally refers to the amplitude of the air pressure in the wave, but sometimes the amplitude of the displacement (movements of the air or the diaphragm of a speaker) is described. The logarithm of the amplitude squared is usually quoted in dB, so a null amplitude corresponds to -∞ dB. Loudness is related to amplitude and intensity and is one of most salient qualities of a sound, although in general sounds can be recognized independently of amplitude.

For electromagnetic radiation, the amplitude corresponds to the electric field of the wave. The square of the amplitude is proportional to the intensity of the wave.

The amplitude may be constant (in which case the wave is a continuous wave) or may vary with time and/or position. The form of the variation of amplitude is called the envelope of the wave.

Pulse amplitude

In telecommunication, pulse amplitude is the magnitude of a pulse parameter, such as the field intensity, voltage level, current level, or power level.

Note 1: Pulse amplitude is measured with respect to a specified reference and therefore should be modified by qualifiers, such as "average", "instantaneous", "peak", or "root-mean-square."

Note 2: Pulse amplitude also applies to the amplitude of frequency- and phase-modulated waveform envelopes.

Timbre

In music, timbre, or sometimes timber (pronounced /ˈtam-bər'/, ˈtim-bər' like timber, or ˈtam(brə),[1] from Fr. timbre) is the quality of a musical note or sound that distinguishes different types of sound production, such as voices or musical instruments. The physical characteristics of sound that mediate the perception of timbre include spectrum and envelope. Timbre is also known in psychoacoustics as sound quality or sound color.

For example, timbre is what, with a little practice, people use to distinguish the saxophone from the trumpet in a jazz group, even if both instruments are playing notes at the same pitch and amplitude. Timbre has been called "the psychoacoustician's multidimensional wastebasket category" [2] as it can denote many apparently unrelated aspects of a sound.

History

The Chinese developed a sophisticated understanding of the musical quality of timbre during the Song Dynasty[citation needed]. They discovered that the timbre of string instruments could be changed depending on how the strings were touched. Strings could be plucked, brushed, hit, scraped, or rubbed to produce different sounds.The Chinese composed music on the Qin, a long, wooden board with strings. Their Qin songs emphasized the timbre, and the changes in sound could be heard throughout the song.

Synonyms

Tone quality is used as a synonym for timbre.

Tone color is also often used as a synonym. People who experience synesthesia may see certain colors when they hear particular instruments. Helmholtz used the German Klangfarbe (tone color), and Tyndall proposed an English translation, clangtint. But both terms were disapproved of by Alexander Ellis who also discredits register and color for their pre-existing English meanings (Erickson 1975, p.7).

Colors of the optical spectrum are not generally explicitly associated with particular sounds. Rather, the sound of an instrument may be described with words like "warm" or "harsh" or other terms, perhaps suggesting that tone color has more in common with the sense of touch than of sight. However, color is often used to describe different types of noise such as pink or white. Noise color is determined by mixing together parts of the visible light spectrum that correspond to the audible sound spectrum. A 20 hertz tone is subsonic and a 20000 hertz tone is ultrasonic, so pink noise is pink because it contains loud low-frequency noise mixed with quieter broadband noise.

American Standards Association definition

The American Standards Association defines timbre as "[...] that attribute of sensation in terms of which a listener can judge that two sounds having the same loudness and pitch are dissimilar". A note to the 1960 definition (p.45) adds that "timbre depends primarily upon the spectrum of the stimulus, but it also depends upon the waveform, the sound pressure, the frequency location of the spectrum, and the temporal characteristics of the stimulus."

Attributes

J.F. Schouten (1968, p.42) describes the "elusive attributes of timbre" as "determined by at least five major acoustic parameters" which Robert Erickson (1975) finds "scaled to the concerns of much contemporary music":

1. The range between tonal and noiselike character.

2. The spectral envelope.

3. The time envelope in terms of rise, duration, and decay.

4. The changes both of spectral envelope (formant-glide) and fundamental frequency (micro-intonation).

5. The prefix, an onset of a sound quite dissimilar to the ensuing lasting vibration.

Spectra

The richness of a sound or note produced by a musical instrument is sometimes described in terms of a sum of a number of distinct frequencies. The lowest frequency is called the fundamental frequency and the pitch it produces is used to name the note. For example, in western music, instruments are normally tuned to A = 440 Hz. Other significant frequencies are called overtones of the fundamental frequency, which may include harmonics and partials. Harmonics are whole number multiples of the fundamental frequency — ×2, ×3, ×4, etc. Partials are other overtones. Most western instruments produce harmonic sounds, but many instruments produce partials and inharmonic tones, such as cymbals and other non-pitched instruments.

When the orchestral tuning note is played, the sound is a combination of 440 Hz, 880 Hz, 1320 Hz, 1760 Hz and so on. The balance of the amplitudes of the different frequencies is responsible for the characteristic sound of each instrument.

The fundamental is not necessarily the strongest component of the overall sound. But it is implied by the existence of the harmonic series — the A above would be distinguishable from the one an octave below (220 Hz, 440 Hz, 660 Hz, 880 Hz) by the presence of the third harmonic, even if the fundamental were indistinct. Similarly, a pitch is often inferred from non-harmonic spectra, supposedly through a mapping process, an attempt to find the closest harmonic fit.

It is possible to add artificial 'subharmonics' to the sound using electronic effects but, again, this does not affect the naming of the note.

William Sethares (2004) wrote that just intonation and the western equal tempered scale derive from the harmonic spectra/timbre of most western instruments. Similarly the specific inharmonic timbre of Thai metallophones would produce the seven-tone near-equal temperament they do indeed employ. The five-note sometimes near-equal tempered slendro scale provides the most consonance in the combination of the inharmonic spectra of Balinese metallophones with harmonic instruments such as the stringed rebab.

Envelope

The timbre of a sound is also greatly affected by the following aspects of its envelope: attack time and characteristics, decay, sustain, release and transients. Thus these are all common controls on synthesizers. For instance, if one takes away the attack from the sound of a piano or trumpet, it becomes more difficult to identify the sound correctly, since the sound of the hammer hitting the strings or the first blat of the player's lips are highly characteristic of those instruments. The envelope is the overall amplitude structure of a sound, so called because the sound just "fits" inside its envelope: what this means should be clear from a time-domain display of almost any interesting sound, zoomed out enough that the entire waveform is visible.

In music

Timbre is often cited as one of the fundamental aspects of music. Formally, timbre and other factors are usually secondary to pitch. "To a marked degree the music of Debussy elevates timbre to an unprecedented structural status; already in L'Apres-midi d'un Faune the color of flute and harp functions referentially," according to Jim Samson (1977). Surpassing Debussy is Klangfarbenmelodie and surpassing that the use of sound masses.

Erickson (ibid, p.6) gives a table of subjective experiences and related physical phenomena based on Schouten's five attributes:

|Subjective |Objective |

|Tonal character, usually pitched |Periodic sound |

|Noisy, with or without some tonal character, including |Noise, including random pulses characterized by the rustle time (the mean |

|rustle noise |interval between pulses) |

|Coloration |Spectral envelope |

|Beginning/ending |Physical rise and decay time |

|Coloration glide or formant glide |Change of spectral envelope |

|Microintonation |Small change (one up and down) in frequency |

|Vibrato |Frequency modulation |

|Tremolo |Amplitude modulation |

|Attack |Prefix |

|Final sound |Suffix |

Often listeners are able to identify the kind of instrument even across "conditions of changing pitch and loudness, in different environments and with different players." In the case of the clarinet, an acoustic analysis of the waveforms shows they are irregular enough to suggest three instruments rather than one. David Luce (1963, p.17) suggests that this implies "certain strong regularities in the acoustic waveform of the above instruments must exist which are invariant with respect to the above variables." However, Robert Erickson argues that there are few regularities and they do not explain our "powers of recognition and identification." He suggests the borrowing from studies of vision and visual perception the concept of subjective constancy. (Erickson 1975, p.11)

Spelling

Though timber is accepted, the more common spelling is timbre to distinguish the word from timber ("wood").

Microphone

A microphone, sometimes referred to as a mike or mic (both pronounced /ˈmaɪk/), is an acoustic to electric transducer or sensor that converts sound into an electrical signal.

[pic]

A Neumann U87 capacitor microphone

Microphones are used in many applications such as telephones, tape recorders, hearing aids, motion picture production, live and recorded audio engineering, in radio and television broadcasting and in computers for recording voice, VoIP, and for non-acoustic purposes such as ultrasonic checking.

History

Several early inventors built primitive microphones (then called transmitters) prior to Alexander Bell, but the first commercially practical microphone was the carbon microphone conceived in October 1876 by Thomas Edison. Many early developments in microphone design took place at Bell Laboratories.

Principle of operation

[pic]

[pic]

Edmund Lowe away from the mic

A microphone is a device made to capture waves in air, water (hydrophone) or hard material and translate them into an electrical signal. The most common method is via a thin membrane producing some proportional electrical signal. Most microphones in use today for audio use electromagnetic generation (dynamic microphones), capacitance change (condenser microphones) or piezoelectric generation to produce the signal from mechanical vibration.

Microphone varieties

Condenser, capacitor or electrostatic microphones

[pic]

[pic]

Inside the Oktava 319 condenser microphone.

Technology

In a condenser microphone, also known as a capacitor microphone, the diaphragm acts as one plate of a capacitor, and the vibrations produce changes in the distance between the plates.

There are two methods of extracting an audio output from the transducer thus formed. They are known as DC biased and RF (or HF) condenser microphones.

DC-biased microphone operating principle

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:

[pic]

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.)

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 and the value of the bias resistor form a filter which is highpass for the audio signal, and lowpass for the bias voltage. Note that the time constant of a RC circuit equals the product of the resistance and capacitance.

Within the time-frame of the capacitance change (on the order of 100 μs), the charge thus appears practically constant and the voltage across the capacitor adjusts itself 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.

[pic]

[pic]

An Oktava condenser microphone.

RF condenser microphone operating principle

In a DC-biased condenser microphone, a high capsule polarisation voltage is necessary. In contrast, RF condenser microphones use a comparatively low RF voltage, generated by a low-noise oscillator. The oscillator is frequency modulated by the capacitance changes produced by the sound waves moving the capsule diaphragm. Demodulation yields a low-noise audio frequency signal with a very low source impedance. This technique achieves better low frequency response - in fact it will theoretically operate down to DC.

The RF biasing process results in a lower electrical impedance capsule, a useful byproduct of which is that RF condenser microphones can be operated in damp weather conditions which would effectively short out a DC biased microphone. The Sennheiser "MKH" series of microphones use the RF biased technique.

Usage

Condenser microphones span the range from cheap throw-aways to high-fidelity quality instruments. They generally produce a high-quality audio signal and are now the popular choice in laboratory and studio recording applications. They require a power source, provided either from microphone inputs as phantom power or from a small battery. Power is necessary for establishing the capacitor plate voltage, and is also needed for internal amplification of the signal to a useful output level. Condenser microphones are also available with two diaphragms, the signals from which can be electrically connected such as to provide a range of polar patterns (see below), such as cardioid, omnidirectional and figure-eight. It is also possible to vary the pattern smoothly with some microphones, for example the Røde NT2000.

Electret condenser microphones

An electret microphone is a relatively new type of capacitor microphone invented at Bell laboratories in 1962 by Gerhard Sessler and Jim West[1]. 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. 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 low-cost and considered low quality, the best ones can now rival capacitor microphones in every respect and can even offer the long-term stability and ultra-flat response needed for a measuring microphone. Unlike other capacitor microphones, they require no polarizing voltage, but normally contain an integrated preamplifier which does require power (often incorrectly called polarizing power or bias). This preamp is frequently phantom powered in sound reinforcement and studio applications. While few electret microphones rival the best DC-polarized units in terms of noise level, this is not due to any inherent limitation of the electret. Rather, mass production techniques needed to produce electrets cheaply don't lend themselves to the precision needed to produce the highest quality microphones.

Dynamic microphones

Dynamic microphones work via electromagnetic induction. They are robust, relatively inexpensive and resistant to moisture, and for this reason they are widely used on-stage by singers. There are two basic types: the moving coil microphone and the ribbon microphone.

Moving coil microphones

[pic]

[pic]

The Shure SM57 and Beta 57A dynamic microphones

Technology

The dynamic principle is exactly the same 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 will 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. AKG D112 is for example designed for bass content rather than treble. In audio engineering several kinds of microphones are often used at the same time to get the best result.

Ribbon microphones

In ribbon microphones a thin, usually corrugated metal ribbon is 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 bidirectional (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 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. Older ribbon microphones, some of which still give very high quality sound reproduction, and were once valued for this reason, but a good low-frequency response could only be obtained only if the ribbon is suspended very loosely, and this made them fragile. Modern ribbon materials have now been introduced that eliminate those concerns. 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. (There are some new modern ribbon microphone designs which incorporate a preamplifier and therefore do require phantom power, also there are new ribbon materials available that are immune to wind blasts and phantom power.)

Carbon microphones

A carbon microphone, formerly used in telephone handsets, is a capsule containing carbon granules pressed between two metal plates. 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 voltage across the two plates, and hence 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.

Unlike other microphone types, the carbon microphone can also be used as a type of amplifier, using a small amount of sound energy to produce 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 microphones

Technology

A crystal 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 Rochelle salt (potassium sodium tartrate), which is a piezoelectric crystal that works as a transducer, both as a microphone and as a slimline loudspeaker component.

Usage

Crystal microphones used to be 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.

Piezo 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 piezos 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.

Laser microphones

Usage

Laser microphones are very rare and expensive, and are most commonly portrayed in movies as spying devices.

Liquid microphones

Main article: Water microphone

Technology

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 even patented by Reginald Fessenden in 1903.

Usage

These were the first working microphones, but they were not practical for commercial application and are utterly obsolete now. It was with a liquid microphone that the famous first phone conversation between Bell and Watson took place. Other inventors, especially Thomas Edison, soon devised superior microphones.

MEMS microphones

The MEMS microphone is also called a microphone chip or silicon microphone. The pressure-sensitive diaphragm is etched directly on a silicon chip by MEMS (MicroElectrical-Mechanical Systems) techniques[citation needed], and is usually accompanied with integrated preamplifier. Most MEMS microphones are modern embodiments of the standard condenser microphone. Often MEMS mics have a built in ADC on the same CMOS chip making the chip a digital microphone and easily integrated into modern digital products. Major manufacturers using MEMS manufacturing for silicon microphones are Akustica (AKU200x), Infineon (SMM310 product), Knowles Electronics and Sonion MEMS.

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 such applications as intercoms or walkie-talkies, where high quality and sensitivity are not needed. 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. 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.

Capsule design and directivity

The shape of the microphone defines its directivity. Inner elements are of major importance and concerns the structural shape of the capsule, outer elements may be the interference tube.

A pressure gradient microphone is a microphone in which both sides of the diaphragm are exposed to the incident sound and the microphone is therefore responsive to the pressure differential (gradient) between the two sides of the membrane. Sound incident parallel to the plane of the diaphragm produces no pressure differential, giving pressure-gradient microphones their characteristic figure-eight directional patterns.

The capsule of a pressure microphone however is closed on one side, which results in an omnidirectional pattern.

Microphone polar patterns

Regarding directionality, omnidirectional microphones are pressure transducers, whereas all others are pressure gradient transducers or a combination between the two.

Common polar patterns for microphones (Microphone facing top of page in diagram, parallel to page):

|[pic] |[pic] |[pic] |[pic] |

|Omnidirectional |Subcardioid |Cardioid |Supercardioid |

|[pic] |[pic] |[pic] | |

|Hypercardioid |Bi-directional |Shotgun | |

A microphone's directionality or polar pattern indicates how sensitive it is to sounds arriving at different angles about its central axis. The above polar patterns represent the locus of points that produce the same signal level output in the microphone if a given sound pressure level 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". 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".

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 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 will give 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 than directional (velocity sensitive) microphones.

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. These diagrams just provide an overview of the typical shapes and their names.)

Cardioids

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[pic]

US664A University Sound Dymamic Supercardioid Microphone

The most common unidirectional microphone is a cardioid microphone, so named because the sensitivity pattern is heart-shaped (see cardioid). A hyper-cardioid is similar but with a tighter area of front sensitivity and a tiny 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.

Bi-directional

Figure 8 or bi-directional microphones receive sound from both the front and back of the element. Most ribbon microphones are of this pattern.

Shotgun

[pic]

[pic]

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 more sensitive to the front. This results from placing the element inside a tube with slots cut along the side; wave-cancellation eliminates most of the off-axis noise. Shotgun microphones are commonly used on TV and film sets, and for field recording of wildlife.

An omnidirectional microphone is a pressure transducer; the output voltage is proportional to the air pressure at a given time.

On the other hand, a figure-8 pattern is a pressure gradient transducer; A sound wave arriving from the back will lead to a signal with a polarity opposite to that of an identical sound wave from the front. Moreover, shorter wavelengths (higher frequencies) are picked up more effectively than lower frequencies.

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, at distances of a few centimeters of the sound source results in a bass boost. This is known as the proximity effect[2]

Application-specific microphone designs

A lavalier microphone is made for hands-free operation. These small microphones are worn on the body and held in place either with a lanyard worn around the neck or a clip fastened to clothing. The 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 is one which does not use a cable. It usually transmits its signal using a small FM radio transmitter to a nearby receiver connected to the sound system, but it can also use infrared light if the transmitter and receiver are within sight of each other.

A contact microphone is designed to pick 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 against the object from which vibrations are to be picked up; the contact pin transfers these vibrations to the coil of the transducer. 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, used to pick up speech directly from the throat, around which it is strapped. This allows the device to 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.

Connectivity

[pic]

[pic]

Electronic symbol for a microphone.

Connectors

The most common connectors used by microphones are:

• Male XLR connector on professional microphones

• ¼ inch mono phone plug on less expensive consumer microphones

• 3.5 mm (Commonly referred to as 1/8 inch mini) stereo (wired as mono) mini phone plug on very inexpensive and computer microphones

Some microphones use other connectors, such as 1/4 inch TRS (tip ring sleeve), 5-pin XLR, or stereo mini phone plug (1/8 inch TRS) on some stereo microphones. Some lavalier 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 studios.

Impedance matching

Microphones have an electrical characteristic called impedance, measured in ohms (Ω), that depends on the design. Typically, the rated impedance is stated.[3] Low impedance is considered under 600 Ω. Medium impedance is considered between 600 Ω and 10 kΩ. High impedance is above 10 kΩ.

Most professional microphones are low impedance, about 200 Ω or lower. Low-impedance microphones are preferred over high impedance for two reasons: one is that using a high-impedance microphone with a long cable will result in loss of high frequency signal due to the capacitance of the cable; the other is that long high-impedance cables tend to pick up more hum (and possibly radio-frequency interference (RFI) as well). However, some equipment, such as vacuum tube guitar amplifiers, has an input impedance that is inherently high, requiring the use of a high impedance microphone or a matching transformer. Nothing will be damaged if the impedance between microphone and other equipment is mismatched; the worst that will happen is a reduction in signal or change in frequency response.

To get the best sound in most cases, the impedance of the microphone must be distinctly lower (by a factor of at least five) than that of the equipment to which it is connected. Most microphones are designed not to have their impedance "matched" by the load to which they are connected; doing so can alter their frequency response and cause distortion, especially at high sound pressure levels. There are transformers (confusingly called matching transformers) that adapt impedances for special cases such as connecting microphones to DI units or connecting low-impedance microphones to the high-impedance inputs of certain amplifiers, but microphone connections generally follow the principle of bridging (voltage transfer), not matching (power transfer). In general, any XLR microphone can usually be connected to any mixer with XLR microphone inputs, and any plug microphone can usually be connected to any jack that is marked as a microphone input, but not to a line input. This is because the signal level of a microphone is typically 40-60 dB lower (a factor of 100 to 1000) than a line input. Microphone inputs include the necessary amplification circuitry to deal with these very low level signals. The exception to these comments is in the case of certain ribbon and dynamic microphones which are most linear when operated into a load of known impedance [4]

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 male connector, rather than producing an analog output. Digital microphones may be used either with new equipment which has the appropriate input connections conforming to the AES 42 standard, or else by use of a suitable interface box. Studio-quality microphones which operate in accordance with the AES 42 standard are now appearing from a number of microphone manufacturers.

Measurements and specifications

[pic]

[pic]

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,[5] 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 at least 0–20 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 a (mostly) 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 to 14 dB higher. A quiet microphone will measure typically 20 dBA SPL or 32 dB SPL 468-weighted. The state of the art has recently improved with the NT1-A microphone from Røde, which has a noise level of 5dBA.

The maximum SPL (sound pressure level) the microphone can accept is measured for particular values of total harmonic distortion (THD), typically 0.5%. This is generally inaudible, so one can safely use the microphone at this level 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 perhaps a better 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. Harmonic distortion from microphones is usually of low-order (mostly third harmonic) type, and hence not very audible even at 3-5%. Clipping, on the other hand, usually caused by the diaphragm reaching its absolute displacement limit (or by the preamplifier), will produce a very harsh sound on peaks, and should be avoided if at all possible. 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 will need 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 will generally be set by the noise floor, and too much "sensitivity" in terms of output level will compromise 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 use as standard measuring microphones for the testing of speakers and checking noise levels etc. These are calibrated transducers and will usually be supplied with a calibration certificate stating absolute sensitivity against frequency.

Microphone calibration techniques

[edit] Pistonphone apparatus

A pistonphone is an acoustical calibrator (sound source) using a closed coupler to generate a precise sound pressure for the calibration of instrumentation microphones. The principle relies on a piston mechanically driven to move at a specified rate on a fixed volume of air to which the microphone under test is exposed. The air is assumed to be compressed adiabatically and the SPL in the chamber can be calculated from the adiabatic gas law, which requires that the product of the pressure P with V raised to the power gamma be constant; here gamma is the ratio of the specific heat of air at constant pressure to its specific heat at constant volume. The pistonphone method only works at low frequencies, but it can be accurate and yields an easily calculable sound pressure level. The standard test frequency is usually around 250 Hz.

Reciprocal method

This method relies on the reciprocity of one or more microphones in a group of 3 to be calibrated. It can still be used when only one of the microphones is reciprocal (exhibits equal response when used as a microphone or as a loudspeaker).

Microphone array and array microphones

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

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

Windscreens are used to protect microphones that would otherwise be buffeted by wind or vocal plosives (from consonants such as "P", "B", etc.). Windscreens are often made of soft open-cell polyester or polyurethane foam because of the inexpensive, disposable nature of the foam. Finer windscreens are made of thin plastic screening held out at a distance from the diaphragm by a framework or cage fitted to the microphone body. Pop filters or pop screens are used in controlled studio environments to keep plosives down when recording. Foam windscreens are integral to some microphone designs such as the Shure SM58 which has a thin foam layer just inside the wire mesh ball enclosing the diaphragm. Optional windscreens are often available from the manufacturer and third parties. A very visible example of optional accessory windscreen is the A2WS from Shure, one of which is fitted over each of the two SM57s used on the United States Presidential lectern.[6]

Large, hollow "blimp" or "zeppelin" windscreens are used to surround boom microphones for 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. A refinement of the blimp windscreen is the addition of a synthetic furry cover which can cut wind noise down by a further 12 dB.[7]

Vocalists often use windscreens on handheld microphones to cut plosive breath noise that involve sharp outward airflow from the mouth. The necessity of a windscreen increases the closer a vocalist brings the microphone to their lips. Singers can be trained to soften their plosives, in which case they don't need a windscreen for any reason other than wind.

Windscreens are used extensively in outdoor concert sound and location recording where wind is an unpredictable factor.

Highly directional microphones benefit the most from windscreens, more so than omnidirectional mics which aren't as vulnerable to wind noise.

One disadvantage of windscreens is that the microphone's high frequency response is attenuated by a small amount relative to how dense the protective layer is. Another disadvantage is that windscreens are often fragile, lightweight and/or small, making it easy to damage or lose them. Poorly fitted windscreens can slip to expose microphone porting to wind action and can fall off completely. The polyurethane foam deteriorates over time, requiring replacement in older microphones undergoing refurbishment. Windscreens collect dirt and moisture in their open cells and must be cleaned from time to time to prevent high frequency loss, bad odor and unhealthy conditions for the artist. On the other hand, a major advantage of concert vocalist windscreens is that one can quickly change to a clean windscreen between artists, reducing the chance of transferring germs. Windscreens of various colors can be used to distinguish one microphone from another on a busy, active stage.

Audio connector

[pic]

[pic]

RCA connectors are commonly used for home stereo and video equipment

Audio

Audio connectors are electrical connectors designed and used for audio frequencies. They can be analogue or digital. Common audio connectors include:

• Single-conductor connectors:

o Banana connectors

o Five-way binding posts and banana plugs for loudspeakers

o Fahnestock clips on early breadboard radio receivers.

• Multi-conductor connectors:

o DB25 is for multi-track recording and other multi-channel audio, analog or digital

o DIN connectors and mini-DIN connectors

o RCA connectors, also known as phono connectors or phono plugs, used for analog or digital audio or analog video

o Speakon connectors by Neutrik for loudspeakers

o TRS connectors (tip-ring-sleeve jack plugs), including the original 6.35mm (quarter inch) jack and the more recent 3.5mm (miniature or 1/8th inch) and 2.5mm (subminiature) jacks, all in both mono and stereo (or balanced) versions.

o XLR connectors, also known as Cannon plugs, used for analog or digital balanced audio with a balanced line

• Digital audio interfaces and interconnects:

o ADAT interface (DB25)

o AES/EBU interface, normally with XLR connectors

o S/PDIF, either over electrical coaxial cable (with RCA jacks) or optical fiber (TOSLINK).

Colour codes

|white RCA/TS |analogue audio, left channel; |

| |also mono (RCA/TS), stereo (TRS only), |

| |or undefined/other |

|black RCA/TS/TRS | |

|grey RCA/TS/TRS | |

|red RCA/TS |analogue audio, right channel |

|orange RCA |SPDIF digital audio |

For computers:

|green TRS 3.5mm |stereo output, front channels |

|black TRS 3.5mm |stereo output, rear channels |

|grey TRS 3.5mm |stereo output, side channels |

|gold TRS 3.5mm |dual output, center and subwoofer |

|blue TRS 3.5mm |stereo input, line level |

|pink TS 3.5mm |mono microphone input |

There are exceptions to the above:

• Hosa cables use grey and orange for left and right analogue channels.

• RadioShack cables sometimes use grey and black for left and right.

• Older sound cards had non-standard colour codes until after PC99, prior to that there were no colours at all.

XLR connector

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[pic]

XLR3 cable connectors, female on left and male on right

The XLR connector is an electrical connector design. XLR plugs and sockets are used mostly in professional audio and video electronics cabling applications. Home audio and video electronics normally use RCA connectors.

In reference to its original manufacturer, Cannon (now part of ITT), the connector is colloquially known as a cannon plug or canon. Originally the "Cannon X" series, subsequent versions added a Latch ("Cannon XL") and then a Rubber compound surrounding the contacts, which led to the abbreviation XLR.[1] Many companies now make XLRs. The initials "XLR" have nothing to do with the pinout of the connector. XLR connectors can have other numbers of pins besides three.

They are superficially similar to the older, smaller, and less rugged DIN connector range, but are not physically compatible with them.

Patterns of XLR connector

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Variety of male and female XLR connectors with different numbers of pins

The most common is the 3-pin XLR3, used almost universally as a balanced audio connector for high quality microphones and connections between equipment. XLR4 (with four pins) is used for ClearCom and Telex intercom headsets and handsets, some DC power connections and the older AMX analog lighting control. XLR5 is the standard connector for DMX512 digital lighting control and is also used for dual-element microphones and dual-channel intercom headsets. XLR6 is used for dual channel intercom beltpacks.

Many other types exist, with various pin numbers. Most notable are two now obsolete 3-pin patterns manufactured by ITT Cannon. The power Cannon (also called the XLR-LNE connector) had shrouded pins and red insulation, it was intended as a mains power connector, but has been superseded by the IEC mains connector and increasingly, more recently, the PowerCon connector developed by Neutrik.

The loudspeaker Cannon had blue or white insulation (depending on its gender), was intended for connections between audio power amplifiers and loudspeakers. At one time XLR3 connectors were also used extensively on loudspeaker cables, as when first introduced they represented a new standard of ruggedness, and economic alternatives were not readily available. The convention was that a 2-conductor loudspeaker cable had XLR3F connectors on both ends, to distinguish it from a 3-conductor shielded signal level cable which has an XLR3F at one end and an XLR3M at the other. Either pin 2 or 3 was live, depending on the manufacturer, with pin 1 always the 'earthy' return. This usage is now both obsolete and dangerous to equipment but is still sometimes encountered, especially on older equipment. For example, some loudspeakers have a built-in XLR3M as an input connector. This use was superseded in professional audio applications by the Neutrik Speakon connector.

The female XLR connectors are designed to first connect pin 1 (the earth pin), before the other pins make contact, when a male XLR connector is inserted. With the ground connection established before the signal lines are connected, the insertion (and removal) of XLR connectors in live equipment is possible without picking up external signals (as it usually happens with, for example, RCA connectors).

Lighting control for entertainment applications is widely connected using five pin XLRs. While only three pins are used to carry the DMX512 signal, the design allows expansion with the remaining two pins considered for use with Remote Device Management (RDM) and Architecture for Control Networks (ACN) and also prevents users from confusing lighting with common XLR3 audio cables. Unfortunately, five pin XLRs still allow the use of lower-grade (non-110 Ohm) microphone cable for transmission of signals. Some manufacturers of DJ lighting and professional lighting are still using three-pin connectors as their standard. Manufacturers such as Leviton and Lightronics have even established new protocols not compatible with DMX512 that use three pin XLR to control lighting devices (primarily dimmers made by the same manufacturer). Non-DMX512 protocols using three pins are not generally accepted as a professional standard and are used primarily to promote consumers to buy multiple products from the same company. Any protocol using control or management on pins 4 & 5 is against the stated use in the USITT DMX512 standard, and all of its later revisions. Their stated use is for a second universe of DMX512 (thereby allowing two universes to pass down one cable. i.e. 1024 channels). WARNING: Any use, other than for the transmission of DMX512, of pins 4&5 on a DMX512 line may destroy connected equipment.

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|XLR-LNE 2-pin socket and plug, originally used for |Male and female XLR4 panel |XLR5 Socket |Female XLR6 panel |

|mains power connections |connectors | |connector |

XLR3 connectors

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Left to right: Cannon XLR3-12C (line), Switchcraft X3F (line), Neutrik NC3MP panel, Neutrik NC3FP panel

EIA Standard RS-297-A describes the use of the XLR3 for balanced audio signal level applications:

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|Pin |Function |

|1 |Chassis ground (cable shield) |

|2 |Normal polarity ("hot") |

|3 |Inverted polarity ("cold") |

• When looking at a female connector, the top left hole is 2, top right is 1, and bottom is 3.

• When looking at a male connector, the top left pin is 1, top right is 2, and bottom is 3.

Some audio equipment manufacturers reverse the use of pin 2 (properly the normal input) and pin 3 (inverting input). This reflects their own previous usage before any standard existed. Pin 1 is always ground, and many connectors connect it internally to the connector shell or case.

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XLR and 1/4" TRS combo jack.

Note that neither the standards nor manufacturers agree on the best way to handle the usage of pin 1 at both ends of a cable, particularly with respect to the cable shield, the connector's shell, signal ground, and a third cable wire connected to pin 1 — which may (or may not) be connected to the shield. Comments on AES48

An XLR3M (male) connector is used for an output and an XLR3F (female) for an input. Thus a microphone will have a built-in XLR3M connector, and signal cables such as microphone cables will each have an XLR3F at one end and an XLR3M at the other. At the stage box end of a multicore cable, the inputs to the mixing desk will be XLR3F connectors, while the returns to the stage will be XLR3M connectors. Similarly, on a mixing desk, the microphone inputs will be XLR3F connectors, and any balanced outputs XLR3M connectors.

Neutrik also offers several models of "combo" jacks that accept both XLR and 1/4" TS or T

RCA connector

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RCA Plugs for composite video and stereo audio

An RCA jack, also referred to as a phono connector or CINCH/AV connector, is a type of electrical connector that is commonly used in the audio/video market. The name "RCA" derives from the Radio Corporation of America, which introduced the design by the early 1940s to allow phonograph players to be connected to amplifiers.

For many other applications it began to replace the older jack plugs used in the audio world when component high fidelity started becoming popular in the 1950s.

The corresponding plug is called an RCA plug or a phono plug. The latter is often confused with a phone plug which refers to a TRS connector.

Uses

In the most normal usage, cables have a standard plug on each end, consisting of a central male connector, surrounded by a ring. The ring is often segmented for flexibility. Devices mount the jack, consisting of a central hole with a ring of metal around it. The ring is slightly smaller in diameter and longer than the ring on the plug, allowing the plug's ring to fit tightly over it. The jack has a small area between the outer and inner rings which is filled with an insulator, typically plastic (very early versions, or those made for use as RF connectors used ceramic).

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Audio grade RCA connectors.

As with many other connectors, the RCA has been adopted for other uses than originally intended, including as a power connector, an RF connector, and as a connector for loudspeaker cables. Its use as a connector for composite video signals is extremely common, but provides poor impedance matching. RCA connectors and cable are also commonly used to carry SPDIF-formatted digital audio, with plugs colored orange to differentiate them from other typical connections.

Connections are made by pushing the cable's plug into the jack on the device. The signal-carrying pin protrudes from the plug, and often comes into contact with the socket before the grounded rings meet, resulting in loud hum or buzz if the audio components are powered while making connections. Continuous noise can occur if the plug partially falls out of the jack, breaking ground connection but not the signal. Some variants of the plug, especially cheaper versions, also give very poor grip and contact between the ground sheaths due to their lack of flexibility.

They are often color coded, yellow for composite video, red for the right channel and white or black for the left channel of stereo audio. This trio (or pair) of jacks can be found on the back of almost all audio and video equipment. At least one set is usually found on the front panel of modern TV sets, to facilitate connection of camcorders, digital cameras, and video gaming consoles. Although nearly all audio-visual connectors, including audio, composite and component video, and S/PDIF audio can use identical 75 Ω cables, sales of special-purpose cables for each use have proliferated. Varying cable quality means that a cheap line-level audio cable might not successfully transfer component video or digital audio signals.[citation needed]

The male plug has a center pin which is 3.70 mm in diameter, and is surrounded by an outer shell which is 8.25 mm in diameter

Disadvantages

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"Bullet plug" variation. Notice the hollow center conductor and the single pin point for the return signal.

One problem with the RCA jack system is that each signal requires its own wire. Even in a simple case of attaching a cassette deck one may need four of them, two for input, two for output. In any common setup this quickly leads to cable spaghetti, which is made worse if one considers more complex signals like component video (a total of three for video and two for analog audio or one for digital coaxial audio). There have been numerous attempts to use combined connectors in both the audio and video world, but none of these have ever become universal—with the exception of the SCART connector, which has become very successful in Europe. For a time the 5-pin DIN plug was popular for bi-directional stereo connection between A/V equipment, but it has been entirely displaced by the phono connector on modern consumer devices, despite the fact that it takes four phono jacks to replace it. Nearly all modern TV sets, VCRs, and DVD players sold in Europe have SCART sockets, and in many cases they have no RCA sockets at all. However, RCA-to-SCART adapters are easily available, as SCART cables can also carry composite video and stereo audio, among other signals. For the purposes of consumer digital AV connections, HDMI is largely replacing RCA jacks as, like SCART, it has the ability to carry several different types of signals in the one connector.

Origin

The word phono is an abbreviation of the word phonograph, because this connector was originally created to allow the connection of a phonograph turntable to a radio receiver, utilizing the radio as an amplifier. This setup was present in most radios manufactured in the 1930s onward by the Radio Corporation of America (RCA), who later marketed a special turntable for 45 RPM records.

Color coding in consumer equipment

Plugs and sockets on consumer equipment are conventionally color-coded to aid correct connections. The standard[1] colors for the various signals are shown below.

Note: in stereo audio applications there are combinations of either Black+Red or White+Red RCA connectors - In both cases, Red denotes Right. Purple may also be substituted by Black.

|Analog audio |Left/Mono |White |  |

| |Right |Red |  |

| |Center |Green |  |

| |Left surround |Blue |  |

| |Right surround |Gray |  |

| |Left back surround |Brown |  |

| |Right back surround |Tan |  |

| |Subwoofer |Purple |  |

|Digital audio |S/PDIF |Orange |  |

|Composite analog video |Composite |Yellow |  |

|Component analog video (YPbPr) |Y |Green |  |

| |Pb |Blue |  |

| |Pr |Red |  |

|Component analog video/VGA (RGB/HV) |R |Red |  |

| |G |Green |  |

| |B |Blue |  |

| |H/Horizontal sync |Yellow |  |

| |V/Vertical sync |White |  |

TRS connector

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TRS connector

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"Triple contact plug" as described in 1907.

A TRS connector, also called a jack plug (UK) or phone plug (U.S.), is a common audio connector. It is cylindrical in shape, typically with three contacts, although sometimes with two (a TS connector) or four (a TRRS connector). It was invented for use in telephone switchboards in the 19th century and is still widely used, both in its original quarter-inch (6.3 mm) size and in miniaturized versions. The connector's name is an acronym derived from the names of three conducting parts of the plug: Tip, Ring, and Sleeve[1] – hence, TRS.

In the U. K., the terms jack plug and jack socket are commonly used for the respectively male and female TRS connectors.[2]

In the U. S., a female connector is called a jack. The terms phone plug and phone jack are commonly used to refer to TRS connectors,[3] but are also sometimes used colloquially to refer to telephone plugs and the corresponding jacks that connect wired telephones to wall outlets. The similar terms phono plug and phono jack normally refer to RCA connectors. To unambiguously refer to the connectors described here, the diameter or other qualifier is often added, e.g. 1/4-inch phone plug, 3.5 mm phone jack, or stereo phone plug, for the three-contact version.

The initial application for the TRS connector was in telephone equipment, which explains why, to this day, it is often termed a "phone plug," even though its use in telephony applications ended many decades ago. The connector's association with stereo headphones possibly helped maintain this term.

Modern connectors

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2.5 mm (3/32") mono (TS), 3.5 mm (1/8") mono and stereo (TRS), and 6.3 mm (1/4") stereo jack plugs

Modern TS and TRS connectors are available in three standard sizes. The original 1/4" (6.35 mm) version dates from 1878, for use in manual telephone exchanges—making it possibly the oldest electrical connector standard still in use. The 3.5 mm or miniature and 2.5 mm or subminiature sizes were originally designed as two-conductor connectors for earpieces on transistor radios. The 3.5 mm and 2.5 mm sizes are also referred to as 1/8" and 3/32" respectively in the United States, though those dimensions are only approximations. All three sizes are now readily available in two-conductor (mono) and three-conductor (stereo or tip ring sleeve) versions.

Four and five conductor versions of the 3.5 mm plug are used for certain applications. A four conductor version is becoming a de facto standard output connector for compact camcorders, providing stereo sound plus a video signal. This interface is also seen on some laptop computers. Proprietary interfaces using both four and five conductor versions exist, such as the audio connector on the first four generations of iPod MP3 players (the 5th generation player now uses a standard 3 conductor cable), where the extra conductors were used to supply power for accessories. There is also an optical connector used for TOSLINK (mainly on things like portable equipment; hi-fi separates and similar tend to use the standard square connector) that is the same size as a 3.5 mm jack. Sockets exist that can make either an optical connection to such a plug or an electrical connection to a stereo jack plug, such as the headphone jacks on many laptops.

A three or four conductor version of the 2.5 mm plug is widely used on cell phone handsfree headsets, providing mono (three conductor) or stereo (four conductor) sound and a microphone input. It should be noted that the use of common stereo headphones with the 2.5 mm plug are often not compatible with this type of socket.

Although relatively unknown in modern electronics, the professional audio world and the telecommunication industry rely heavily on tiny telephone (TT) connectors which use mid-size phone plugs with a 4.4 mm (0.173-inch) diameter shaft. In the telecom world, this is known as a "bantam" plug. Due to their compactness and reliability, TTs are often used for professional console and outboard patchbays in studios and live sound applications, in which a single patch panel may require hundreds of patch points in a limited space. The TRS versions of TT connectors are capable of handling balanced line signals and are preferred in pro audio installations

Both two-conductor and three-conductor versions of the three standard sizes are readily available in male (plug) and female (socket or simply "jack") line versions, and panel-mounting female versions. Panel-mounting male versions of these also exist but are rare, as they are vulnerable to mechanical damage and therefore unreliable. Female line versions are also notoriously unreliable and are avoided by many users.

The most common arrangement remains to have the male plug on the cable, and the female socket mounted in a piece of equipment, which was the original intention of the design. A considerable variety of line plugs and panel sockets is available, including plugs suiting various cable sizes, right angle plugs, and both plugs and sockets in a variety of price ranges and with current capacities up to about 15 amperes for the 1/4" version.

Non-standard sizes, both diameters and lengths, are also available from some manufacturers, and are used when it is desired to restrict the availability of matching connectors.

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A dual 310 patch cable, two pin jack plug

• A two-pin version, known to the telecom industry as a "310 connector" consists of two TRS 6.3 mm jack plugs at a centre spacing of 1". The socket versions of these can be used with normal jack plugs provided the plug bodies are not too large, but the plug version will only mate with two jack sockets at 1" centre spacing, or with line sockets, again with sufficiently small bodies. These connectors are still widley used today in telephone company central offices on "DSX" patch panels for DS1 circuits. A similar type of 3.5 mm connector is often used in the armrests of aircraft, as part of the on-board entertainment system. Plugging a stereo plug into one of the two mono jacks typically results in the audio coming into only one ear. Adaptors are available.

• A short-barrelled version also exists, once used on high-impedance mono headphones, and in particular those used in World War II aircraft. It is physically possible to use a normal plug in a short socket, but a short plug will neither lock into a normal socket nor complete the tip circuit. These are still manufactured but are now regarded as a non-standard size.

Mono and stereo compatibility

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Old profile jack plugs. The leftmost plug has three conductors; the others have two.

At the top is a three-conductor jack from the same era.

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Modern profile 2-conductor 1/4" jack plugs.

In the original application in manual telephone exchanges, many different configurations of 1/4" jack plug were used, some accommodating five or more conductors, with several tip profiles. Of these many varieties, only the two-conductor version with a rounded tip profile was compatible between different manufacturers, and this was the design that was at first adopted for use with microphones, electric guitars, headphones, loudspeakers, and many other items of audio equipment.

When a three-conductor version of the 1/4" jack was introduced for use with stereo headphones, it was given a sharper tip profile in order to make it possible to manufacture jacks (sockets) that would accept only stereo plugs, to avoid short-circuiting the right channel amplifier. This attempt has long been abandoned, and now the normal convention is that all plugs fit all sockets of the same size, regardless of whether they are mono or stereo. Most 1/4" plugs, mono or stereo, now have the profile of the original stereo plug, although a few rounded mono plugs are also still produced. The profiles of stereo miniature and subminiature plugs have always been identical to the mono plugs of the same size.

The results of this physical compatibility are:

• If a two-conductor plug of the same size is connected to a three-conductor socket, the result is that the ring (right channel) of the socket is grounded. This property is deliberately used in several applications, see "tip ring sleeve", below. However, grounding one channel may also be dangerous to the equipment if the result is to short circuit the output of the right channel amplifier. In any case, any signal from the right channel is naturally lost.

• If a three-conductor plug is connected to a two-conductor socket, normally the result is to leave the ring of the plug unconnected (open circuit). In the days of valves ("tubes" in the U.S.) this was also potentially dangerous to equipment but most solid state devices tolerate this condition well. A stereo socket could be wired as a mono socket to ground the ring in this situation, but the more conventional wiring in this case is to leave the ring unconnected, exactly simulating a mono socket.

Uses

Some common uses of jack plugs and their matching sockets are:

• Headphone and earphone jacks on a wide range of equipment. 1/4 in. plugs are common on standalone equipment, while 3.5 mm plugs are nearly universal for portable audio equipment. 2.5 mm plugs are not as common, but are sometimes used on communication equipment such as two-way radios and mobile phones.

• Microphone inputs on tape and cassette recorders, sometimes with remote control switching on the ring.

• Patching points on a wide range of equipment.

• Personal computer sound cards. Stereo 3.5 mm jacks are used for:

o Line in (stereo)

o Line out (stereo)

o Headphones/loudspeaker out (stereo)

o Microphone input (mono, sometimes with 5v power available on the ring)

• Electric guitars. Almost all electric guitars use a ¼ in mono jack (socket) as their output connector. Some makes (such as Shergold) use a stereo jack instead for stereo output, or a second stereo jack, in addition to a mono jack (as with Rickenbacker).

• Instrument amplifiers for guitars, basses and similar amplified musical instruments. ¼ in jacks are overwhelmingly the most common connectors for:

o Inputs. A shielded cable with a mono ¼ in jack plug on each end is commonly called a guitar cord or a patching cord, the first name reflecting this usage, the second the history of the jack plug's development for use in manual telephone exchanges.

o Loudspeaker outputs, especially on low-end equipment. Speakon connectors are generally considered superior and so are usually preferred on higher-end equipment, although it is not uncommon to find both provided for compatibility. Heavy-duty ¼ in loudspeaker jacks are rated at 15 A maximum which limits them to applications involving less than 1800 watts. ¼ in loudspeaker jacks commonly aren't rigged to lock the plug in place and will short out the amplifier's output circuitry if connected or disconnected when the amplifier is live.

o Line outputs.

o Foot switches and effects pedals. Stereo plugs are used for double switches (for example by Fender). There is little compatibility between makers.

o Effects loops, which are normally wired as patch points.

• Electronic keyboards use jacks for a similar range of uses to guitars and amplifiers, and in addition

o Sustain pedals.

o Expression pedals.

• Electronic drums use jacks to connect sensor pads to the synthesizer module or MIDI encoder. In this usage, a change in voltage on the wire indicates a drum stroke.

• Some compact and/or economy model audio mixing desks use stereo jacks for balanced microphone inputs.

• The majority of professional audio equipment uses mono jacks as the standard unbalanced input or output connector, often providing a ¼ in unbalanced line connector alongside (or in a few cases in the middle of!) and as an alternative to an XLR balanced line connector.

• Modular synthesizers commonly use monophonic cables for creating patches.

• ¼ in connectors are widely used to connect external processing devices to mixing consoles' insert points (see Insert (effects processing)). TRS or TS connectors might be used in pairs as separate Send and Return jacks or a single TRS jack might be employed for both Send and Return in which case the signals are unbalanced. The single unbalanced combination Send/Return TRS insert jack saves both panel space and component complexity. Note that mixing console insert points can also be XLR, RCA or Bantam TT (tiny telephone) jacks, depending on the make and model.

• Some small electronic devices such as audio cassette players, especially in the cheaper price brackets, use a two-conductor 3.5 mm or 2.5 mm jack as a DC power connector.

• Some photographic studio strobe lights have ¼ in or 3.5 mm jacks for the flash synchronization input. A camera's electrical flash output (PC socket or hot shoe adapter) is cabled to the strobe light's sync input jacks. Some examples: Calumet Travelite, and Speedotron use a ¼ in mono jack as the sync input; White Lightning uses ¼ in stereo jacks; Pocket Wizard (radio trigger) and Alien Bees use 3.5 mm mono jacks.

• Some cameras (for example, Canon, Sigma, and Pentax DSLRs) use the 2.5mm stereo jack for the connector for the remote shutter release (and focus activation); examples are Canon's RS-60E3 remote switch and Sigma's CR-21 wired remote control.

• Some miniaturized electronic devices use 2.5 or 3.5 mm jack plugs as serial port connectors for data transfer and unit programming. This technique is particularly common on graphing calculators, such as the TI-83 series, and some types of amateur and two-way radio, though in some more modern equipment USB mini-B connectors are provided in addition to or instead of jack connectors. The second-generation iPod Shuffle from Apple has a single TRS jack which serves as headphone, USB, or power supply, depending on the connected plug.

• On CCTV cameras and video encoders, mono audio in (originating from a microphone in or near the camera) and mono audio out (destined to a speaker in or near the camera) are provided on a single three-conductor connector, where one signal is on the tip conductor and the other is on the ring conductor.[4]

Switch contacts

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A jack plug breaks the contact of a normally closed switch.

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Miniature jack plugs and jacks. All are 3.5 mm except the gold-plated plug, which is 2.5 mm. All the jacks are two-conductor (TS). The tan-colored jacks have a normally-closed switch.

Panel-mounting jacks are often provided with switch contacts. Most commonly, a mono jack is provided with a single normally closed (NC) contact, which is connected to the tip (live) connection when no plug is in the socket, and disconnected when a plug is inserted. Stereo sockets commonly provide two such NC contacts, one for the tip (left channel live) and one for the ring or collar (right channel live). Some designs of jack also have such a connection on the sleeve, as this contact is usually ground it is not much use for signal switching but could be used to indicate to electronic circuitry that the socket was in use.

Less commonly, some jacks are provided with normally open (NO) or change-over contacts, and/or the switch contacts may be isolated from the connector.

The original purpose of these contacts was for switching in telephone exchanges, for which there were many patterns. Two sets of change-over contacts, isolated from the connector contacts, were common. The more recent pattern of one NC contact for each signal path, internally attached to the connector contact, stems from their use as headphone jacks. In many amplifiers and equipment containing them, such as electronic organs, a headphone jack is provided that disconnects the loudspeakers when in use. This is done by means of these switch contacts. In other equipment, a dummy load is provided when the headphones are not connected. This is also easily provided by means of these NC contacts.

Other uses for these contacts have been found. One is to interrupt a signal path to enable other circuitry to be inserted. This is done by using one NC contact of a stereo jack to connect the tip and ring together when no plug is inserted. The tip is then made the output, and the ring the input (or vice versa), thus forming a patch point.

Another use is to provide alternative mono or stereo output facilities on some guitars and electronic organs. This is achieved by using two mono jacks, one for left channel and one for right, and wiring the NC contact on the right channel jack to connect the two connector tips together when the right channel output is not in use. This then mixes the signals so that the left channel jack doubles as a mono output.

Where a 3.5 mm or 2.5 mm jack is used as a DC power inlet connector, a switch contact may be used to disconnect an internal battery whenever an external power supply is connected, to prevent incorrect recharging of the battery.

A three-conductor signal input socket is used on some battery-powered guitar effects pedals to eliminate the need for a separate power switch. When the user plugs in a two-conductor guitar or microphone lead, the resulting short-circuit between earth and ring connects an internal battery to the unit's circuitry, ensuring that it powers up or down automatically whenever a signal lead is inserted or removed. A side effect is the risk of inadvertently discharging the battery if the lead is not removed after use, for example if equipment is left connected overnight.

Tip/ring/sleeve terminology

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1. Sleeve: usually ground

2. Ring: Right-hand channel for stereo signals, negative phase for balanced mono signals, power supply for power-requiring mono signal sources

3. Tip: Left-hand channel for stereo signals, positive phase for balanced mono signals, signal line for unbalanced mono signals

4. Insulating rings

In twisted pair wiring to this day, the non-inverting and/or "live" (or "hot") wire of each pair is known as the ring, while the inverting and/or "earthy" (or "neutral") wire is known as the tip, inherited from the traditional connection via the TRS connector in telephone systems. If the pair is shielded, or if the pair is accompanied by a dedicated earth wire, this third conductor is known as the sleeve. This usage corresponds to the connection to a three-connector jack plug in a manual telephone exchange. This appears to have originated with the use of TRS jacks by switchboard operators with the tip and ring wires attached to the corresponding parts of the jack. Originally, the hot and ground were reversed, but often the metallic desktops of the switch boards were scarred by the discharge from the tips and the system was reversed to the present usage.

The term tip ring sleeve is more common in some English-speaking countries than others. Outside of the USA the term stereo jack plug is probably more common, even for connectors not used for stereo. The modern profile three-conductor jack plug was originally designed for stereo signal connections, with left channel on the tip, right on the ring and common return on the body or sleeve. The term TRS is particularly appropriate to distinguish these three-conductor (stereo) plugs used in other than stereo applications.

| |Unbalanced Output |Unbalanced Input |Unbalanced Insert |Balanced |Stereo |

|Tip |Signal |Signal |Send or Return signal |Positive/"Hot" |Left channel |

|Ring |Ground or No Connection |Ground or No Connection |Return or Send signal |Negative/"Cold" |Right channel |

|Sleeve |Ground |Ground |Ground |Ground |Ground |

Note that early QSC amplifiers used a Tip Negative, Ring Positive input jack wiring scheme. [5]

Whirlwind Line Balancer/Splitters do not use the Sleeve as a conductor on their unbalanced ¼ in TRS input. Tip and Ring are wired to the transformer's two terminals; Sleeve is not connected. [6]

Usage

Audio

When a TRS is used to make a balanced connection, the two active conductors are both used for a monaural signal. The ring, used for the right channel in stereo systems, is used instead for the inverting input. This is a common use in small audio mixing desks, where space is a premium and they offer a more compact alternative to XLR connectors. Another advantage offered by TRS connectors used for balanced microphone inputs is that a standard unbalanced signal lead using a mono jack plug can simply be plugged into such as input. The ring (right channel) contact then makes contact with the plug body, correctly grounding the inverting input.

The disadvantage of using TRS jacks for balanced audio connections is that the ground mates last and the socket grounds the plug tip and ring when inserting or pulling out the plug. This causes bursts of hum, cracks and pops and may stress some outputs as they will be short circuited briefly, or longer if the plug is left half in. Professional audio equipment uses XLR connectors which mate the ground signal on pin 1 first.

TRS connectors are also commonly used as unbalanced audio patch points (or insert points, or simply inserts), with the output on many mixers found on the tip (left channel) and the input on the ring (right channel). This is often expressed as "tip send, ring return." Other mixers have unbalanced insert points with "ring send, tip return." One advantage of this system is that the switch contact in the panel socket, originally designed for other purposes, can be used to close the circuit when the patch point is not in use. Another is that if the "tip send" patch point is used as an output only, use of a mono jack plug correctly grounds the input. In the same fashion, use of a "tip return" insert style allows a mono jack plug to bring an unbalanced signal directly into the circuit, correctly grounding the output. Combining Send and Return functions via single 6.35 mm TRS connectors in this way is seen in very many professional and semi-professional audio mixing desks, due to the halving of space needed for insert jack fields which would otherwise require two jacks, one for Send and one for Return. The tradeoff is that unbalanced signals are more prone to buzz, hum and outside interference.

In some TRS inserts, the concept is extended by using specially designed TRS jacks that will accept a mono jack plug partly inserted ("to the first click") and will then connect the tip to the signal path without breaking it. Most standard TRS jacks can also be used in this way with varying success, but neither the switch contact nor the tip contact can be relied upon unless the internal contacts have been designed with extra strength for holding the plug tip in place. Even with stronger contacts, an accidental mechanical movement of the inserted plug can interrupt signal within the circuit. For maximum reliability, any usage involving "first click" or "half-click" will instead rewire the plug to short Tip and Ring together and then insert this modified plug all the way into the jack.

The TRS Tip Return, Ring Send unbalanced insert configuration is mostly found on older mixers. This allowed for the insert jack to serve as a standard-wired mono line input that would bypass the mic preamp (and likely a resistive pad, as well as other circuitry, depending on the design), and thus improve sound quality. However tip send has become the generally accepted standard for mixer inserts since the early-to-mid 1990s. The TRS Ring Send configuration is still found on some compressor sidechain input jacks such as dbx 166XL.

In some very compact equipment, 3.5 mm TRS jacks are used as patch points.

Some sound recording devices use a TRS as a mono microphone input, using the tip as the signal path and the ring to connect a standby switch on the microphone.

Computer sound

Personal computer sound cards from Creative Labs, Sound Blaster or compatible to these use a 3.5 mm TRS as a mono microphone input, and deliver a 5 V polarising voltage on the ring to power electret microphones from the card manufacturer. Sometimes called phantom power, this is not a suitable power source for microphones designed for true phantom power and is better called bias voltage. Compatibility between different manufacturers is unreliable.

Normally, 3.5 mm 3-conductor sockets are used in computer soundcards for stereo output. Thus, for a soundcard with 5.1 output, there will be 3 sockets to accommodate 6 channels - front left & right, rear left & right, and center & subwoofer. But the 6.1 and 7.1 channel soundcards from Creative Labs are equipped with 1 and 2 sockets of 3.5 mm 4-conductor sockets respectively. This is to accommodate rear-center (6.1) or side left & right (7.1) channels without additional sockets on the sound card. But speaker have normal 3-conductor sockets. In Creative's documentation, the word "pole" is used instead of "conductor".

The Apple PlainTalk microphone jack used on some older Macintosh systems is designed to accept an extended 3.5 mm TRS; in this case, the tip carries power for a preamplifier inside the microphone. If a PlainTalk-compatible microphone is not available, the jack can accept a line-level sound input, though it cannot accept a standard microphone without a preamp.

Nowadays, all of Apple's computers have combination electric/optical 3.5 mm TRS jacks for both input and output. This allows for conventional stereo input and output with electrical connections, or 5.1 digital input and output with a mini-Toslink cable.

Plug-in power

Recording equipment

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Stereo devices which use "plug-in power": the electret capsules are wired in this way

Many small video cameras, laptops, Minidisc recorders and other consumer devices use a 3.5 mm microphone connector for attaching a (mono/stereo) microphone to the system. These fall into three categories:

• Devices (usually of the "toy" variety), which use an un-powered microphone: usually a cheap dynamic or piezo microphone. The microphone generates its own voltage, and does not require power.

• Devices (usually very expensive recorders, for hi-fi or broadcast use) which use a self-powered microphone: usually an expensive dynamic microphone with internal battery-powered amplifier.

• Devices (most consumer equipment) which use a "plug-in powered" microphone: an electret microphone containing an internal FET amplifier. These provide a good quality signal, in a very small microphone. However, the internal FET requires a DC power supply, which is provided as a bias voltage.

Plug-in power is supplied on the same line as the audio signal, using an RC filter. The DC bias voltage supplies the FET amplifier (at a low current), while the capacitor decouples the DC supply from the AC input to the recorder. Typically, V=1.5 V, R=1 kΩ, C=47 µF.

If a recorder provides plug-in power, and the microphone does not need it, everything will usually work ok, although the sound quality may be lower than expected. In the converse case (recorder provides no power; microphone requires power), no sound will be recorded. Neither misconfiguration will damage consumer hardware, but it could destroy a broadcast-type microphone.

Aircraft headsets

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Aviation plug type U-174/U, commonly used on military aircraft and civil helicopters.

Commercial and general aviation civil airplane headset plugs are similar, but with a difference. A standard 1/4-inch monaural plug, type PJ-055, is used for headphones, paired with special tip-ring-sleeve, 0.206 inch diameter plug, type PJ-068, for the microphone. The extra connection in the microphone plug is used by an optional push-to-talk switch.

Military aircraft and civil helicopters have another type similar to a standard 1/4-inch stereo plug, but with a 0.281-inch diameter short shaft with an extra sleeve, known by the designation U-174/U. This provides four connections in one plug, allowing for a pair of monaural headphones, a microphone, a push-to-talk switch and a common ground conductor.

Some mobile phones such as the Nokia N95, the Apple iPhone and the HP IPAQ 500 Voice Messenger also use a similarly-wired plug for their headphone/microphone set.

Configurations and schematic symbols

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These examples are meant to illustrate each possible component of such jacks, but many other configurations using these basic components are available. All examples in the above figure are oriented so the plug 'enters' from the right.

A. A simple two-conductor jack. The connection to the sleeve is the rectangle towards the right, and the connection to the tip is the line with the notch. Wiring connections are illustrated as white circles.

B. A three-conductor, or TRS, jack. The upper connector is the tip, as it is farther away from the sleeve. The sleeve is shown connected directly to the chassis, a very common configuration. This is the typical configuration for a balanced connection. Some jacks have metal mounting connections (which would make this connection) and some have plastic, to isolate the sleeve from the chassis, and provide a separate sleeve connection point, as in A.

C. This three-conductor jack has two isolated SPDT switches. They are activated by a plug going into the jack, which disconnects one throw and connects the other. The white arrowheads indicate a mechanical connection, while the black arrowheads indicate an electrical connection. This would be useful for a device that turns on when a plug is inserted, and off otherwise, with the power routed through the switches.

D. This three-conductor jack has two normally closed switches connected to the contacts themselves. This would be useful for a patch point, for instance, or for allowing another signal to feed the line until a plug is inserted. The switches open when a plug is inserted. A common use for this style of connector is a stereo headphone jack that shuts off the default output (speakers) when the connector is plugged in.

Color Codes

These codes were standardized by Microsoft and Intel in 1999 for computers as part of the PC99 standard. See: PCxx Standards.

|green TRS 3.5mm |stereo output, front channels |

|black TRS 3.5mm |stereo output, rear channels |

|grey TRS 3.5mm |stereo output, side channels |

|gold TRS 3.5mm |dual output, center and subwoofer |

|blue TRS 3.5mm |stereo input, line level |

|pink TS 3.5mm |mono microphone input |

Mixing console

In professional audio, a mixing console, digital mixing console, mixing desk (Brit.), or audio mixer, also called a sound board or soundboard, 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.

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Structure

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Yamaha 2403 audio mixing console in a 'live' mixing application

The input strip is usually separated into these sections:

• Input Jacks / Microphone preamps

• Basic input controls

• Channel EQ

• Routing Section including Direct Outs, Aux-sends, Panning control and Subgroup assignments

• Input Faders

• Subgroup faders

• Output controls including Master level controls, EQ and/or Matrix routing

On the Yamaha Console to the right, these sections are color coded for quick identification by the operator.

Each signal that is input into the mixer has its own channel. Depending on the specific mixer, each channel is stereo or monaural. On most mixers, each channel has an XLR input, and many have RCA or quarter-inch Jack plug line inputs.

Basic input controls

Below each input, there are usually several rotary controls (knobs, pots). The first is typically a trim or gain control. The inputs buffer the signal from the external device and this controls the amount of amplification or attenuation needed to bring the signal to a nominal level for processing. This stage is where most noise or interference is picked up, due to the high gains involved (around +50 dB, for a microphone). Balanced inputs and connectors, such as XLR or Tip-Ring-Sleeve (TRS) quarter-inch connectors, reduce interference problems.

There may be insert points after the buffer/gain stage, which send to and return from external processors which should only affect the signal of that particular channel. Insert points are most commonly used with effects that control a signal's amplitude, such as noise gates, expanders, and compressors.

Auxiliary send routing

The Auxiliary send routes a split of the incoming signal to an auxiliary bus which can then be used with external devices. Auxiliary sends can either be pre-fader or post-fader, in that the level of a pre-fade send is set by the Auxiliary send control, whereas post-fade sends depend on the position of the channel fader as well. Auxiliary sends can be used to send the signal to an external processor such as a reverb, which can then be routed back through another channel or designated auxiliary returns on the mixer. These will normally be post-fader. Pre-fade auxiliary sends can be used to provide a monitor mix to musicians onstage, this mix is thus independent of the main mix.

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Mixing desk used for live performances.

Channel EQ

Further channel controls affect the equalization of the signal by separately attenuating or boosting a range of frequencies (e.g., bass, midrange, and treble frequencies). Most large mixing consoles (24 channels and larger) usually have sweep equalization in one or more bands of its parametric equalizer on each channel, where the frequency and affected bandwidth of equalization can be selected. Smaller mixing consoles have few or no equalization control. Care must be taken not to add too much EQ to a signal that is already close to clipping; additional energy will overdrive the channel. Some mixers have a general equalization control (either graphic or parametric) at the output.

Subgroup and mix routing

Each channel on a mixer has an audio taper pot, or potentiometer, controlled by a sliding volume control (fader), that allows adjustment of the level, or amplitude, of that channel in the final mix. A typical mixing console has many rows of these sliding volume controls. Each control adjusts only its respective channel (or one half of a stereo channel); therefore, it only affects the level of the signal from one microphone or other audio device. The signals are summed to create the main mix, or combined on a bus as a submix, a group of channels that are then added to get the final mix (for instance, many drum mics could be grouped into a bus, and then the proportion of drums in the final mix can be controlled with one bus fader).

There may also be insert points for a certain bus, or even the entire mix.

Master output controls

Subgroup and main output fader controls are often found together on the right hand side of the mixer or, on larger consoles, in a center section flanked by banks of input channels. Matrix routing is often contained in this master section, as are headphone and local loudspeaker monitoring controls. Talkback controls allow conversation with the artist through their wedges, headphones or IEMs. A test tone generator might be located in the master output section. Aux returns such as those signals returning from outboard reverb devices are often in the master section.

Metering

Finally, there are usually one or more VU or peak meters to indicate the levels for each channel, or for the master outputs, and to indicate whether the console levels are overmodulating or clipping the signal. Most mixers have at least one additional output, besides the main mix. These are either individual bus outputs, or auxiliary outputs, used, for instance, to output a different mix to on-stage monitors. The operator can vary the mix (or levels of each channel) for each output.

As audio is heard in a logarithmic fashion (both amplitude and frequency), mixing console controls and displays are almost always in decibels, a logarithmic measurement system. This is also why special audio taper pots or circuits are needed. Since it is a relative measurement, and not a unit itself (like a percentage), the meters must be referenced to a nominal level. The "professional" nominal level is considered to be +4 dBu. The "consumer grade" level is −10 dBV.

Hardware routing and patching

For convenience, some mixing console racks contain a patch bay or patch panel. These may be more useful for those not using a computer with several plugins on their software.

Most, but not all, audio mixers can

• add external effects.

• use monaural signals to produce stereo sound by adjusting the position of each signal on the sound stage (pan and balance controls).

• provide phantom power (typically 48 volts) required by some microphones.

• create an audible tone via an oscillator, usually at 440 Hz, 1 kHz, or 2 kHz

Some mixers can

• add effects internally.

• interface with computers or other recording equipment (to control the mixer with computer presets, for instance).

• be powered by batteries.

Digital vs. Analog

Digital mixing console sales have increased dramatically since their introduction in the 1990s. Yamaha sold more than 1000 PM5D mixers by July, 2005,[1] and other manufacturers are seeing increasing sales of their digital products. Digital mixers are more versatile than analog ones and offer many new features, such as the ability to save multiple mute groups, multiple VCA groups and channel settings into a scene and reconfigure signal routing at the touch of a button. The faders can be "swapped" or "flipped" to show aux send levels; a feature very useful in mixing artist's monitors. In addition, digital consoles often include a range of special effects such as parametric EQ, compression, gating, reverb, automatic feedback reduction, tap delay and straight delay. Some products are expandable via third-party software features (called plugins) that add further reverb, compression, delay and tone-shaping tools. Several digital mixers include spectrograph and real time analyzer functions. A few incorporate loudspeaker management tools such as crossover filtering and limiting. Digital signal processing can perform automatic mixing for some simple applications, such as courtrooms, conferences and panel discussions, but at this time no digital mixer in live audio includes automixing.

Digital mixers can be designed to be quieter than most analog mixers, as digital mixers often incorporate very low threshold noise gates to stop inactive mix bus background hiss from summing with active signals. Digital circuitry is more resistant to outside interference from radio transmitters such as walkie-talkies and cell phones.

Propagation delay

Digital mixers have an unavoidable amount of latency or propagation delay, ranging from 1.5 milliseconds to as much as 10 ms, depending on the model of digital mixer and what functions are engaged. This small amount of delay isn't a problem for loudspeakers aimed at the audience or even monitor wedges aimed at the artist, but can be disorienting and unpleasant for IEMs (In ear monitors) where the artist hears their voice acoustically in their head and electronically amplified in their ears but delayed by a couple of milliseconds.

Every analog to digital conversion and digital to analog conversion within a digital mixer entails propagation delay. Audio inserts to favorite external analog processors make for almost double the usual delay. Further delay can be traced to format conversions such as from ADAT to AES3 and from normal digital signal processing steps.

Within a digital mixer there can be differing amounts of latency, depending on the routing and on how much DSP is in use. Assigning a signal to two parallel paths with significantly different processing on each path can result in extreme comb filtering when recombined. Some digital mixers incorporate internal methods of latency correction so that such problems are avoided.

Ease of use

Analog consoles remain popular due to their continuing to have one knob, fader or button per function, a reassuring feature for the user. This takes up more physical space but allows more rapid response to changing performance conditions. Most digital mixers take advantage of the technology to reduce the physical space requirements of their product, entailing compromises in user interface such as a single shared channel adjustment area that is selectable for only one channel at a time. Additionally, most digital mixers have virtual pages or layers which change the fader banks into separate controls for additional inputs or for adjusting equalization or aux send levels. This layering can be confusing for operators.

Analog consoles make for simpler understanding of hardware routing. Many digital mixers allow internal reassignment of inputs so that convenient groupings of inputs appear near each other at the fader bank, a feature that can be disorienting for persons having to make a hardware patch change.

On the other hand, many digital mixers allow for extremely easy building of a mix from saved data. USB flash drives and other storage methods are employed to bring past performance data to a new venue in highly portable manner. At the new venue, the traveling mix technician simply plugs the collected data into the venue's digital mixer and quickly makes small adjustments to the local input and output patch layout, allowing for full show readiness in very short order.

Some digital mixers allow offline editing of the mix, a feature that lets the traveling technician use a laptop to make anticipated changes to the show while en route, further shortening the time it takes for the sound system to be ready for the artist.

Sound quality

Both digital and analog mixers rely on analog mic preamps, a high-gain circuit that is the origin of much of the perceived character of sound quality in an audio mixer. In this respect, both formats are on par with each other. In a digital mixer, the mic preamp is followed by an ADC which quantizes the audio stream. Ideally, this process is carefully engineered to deal gracefully with overloading and clipping while delivering an accurate digital stream over the linear dynamic range. Further processing and mixing of digital streams within a mixer need to avoid clipping and truncation if maximum audio quality is desired.

Analog mixers, too, must deal gracefully with overloading and clipping at the mic preamp and as well as avoiding overloading of mix buses. Background hiss in an analog mixer is always present, though good gain stage management minimizes its audibility. Idle subgroups left "up" in a mix will add their background hiss to the main outputs; many digital mixers avoid this problem by low-level gating.

Many electronic design elements combine to affect perceived sound quality, making the global "analog mixer vs. digital mixer" question difficult to answer. Controlled ABX double-blind listening tests haven't been published at this date; no conclusive answer can be reached. Experienced live sound professionals agree that microphones and loudspeakers (with their innate higher distortion levels) are a much greater source of coloration of sound than the choice of mixer. The mix style of the person mixing is also more important than the make and model of audio console. Analog and digital mixers both have been associated with extremely high-quality concert performances and studio recordings.

Remote control

Analog mixing in live sound has had the option since the 1990s of using wired remote controls for certain digital processes such as monitor wedge equalization and parameter changes in outboard reverb devices. That concept has expanded until wired and wireless remote controls are being seen in relation to entire digital mixing platforms. It's possible to set up a sound system and mix via wireless (or wired) laptop, touchscreen or tablet, especially if the performance requires no unpredictable fast responses to multiple changing conditions on stage. Computer networks can connect digital system elements for expanded monitoring and control, allowing the system technician to make adjustments to distant devices during the performance. The use of remote control technology can be utilized to reduce "seat-kills", allowing more paying customers into the performance space.

Virtual mixing

Increasingly, the mixing process can be performed on screen, using computer software and associated input, output and recording hardware. The traditional large control surface of the mixing console is not utilized, saving space at the engineer's mix position. Some virtual mixing (such as the Gamble DCX[2]) uses digital controls of analog audio circuitry, but most virtual mixers are fully digital so as to save cost and physical space. In the virtual studio, there is either no normal mixer fader bank at all or there is a compact group of motorized faders designed to fit into a small space and connected to the computer via USB or Firewire. Many project studios use such a space-efficient solution, as the mixing room at other times can serve as business office, media archival, etc.

Applications

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A Behringer EuroRack UB1002FX in a DJ setup

Dub producers/engineers such as Lee 'Scratch' Perry were perhaps the first musicians to use a mixing board as a musical instrument.

Public address systems will use a mixing console to set microphones for different speakers to the correct level, and can add in recorded sounds into the mix. A major requirement is to minimise audio feedback.

Most bands will use a mixing console to combine musical instruments and vocals to the correct level.

Radio broadcasts use a mixing desk to select audio from different sources, such as CD players, telephones, remote feeds, or prerecorded advertisements.

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