Sound
Sound is what can be perceived by a living organism through its
Sound measurements
sense of hearing.[1] Physically, sound is vibrational mechanical Sound pressure p
energy that propagates through matter as a wave.
Sound pressure level (SPL)
For humans, hearing is limited to frequencies between about 20 Hz Particle velocity v
and 20000 Hz, with the upper limit generally decreasing with age. Particle velocity level (SVL)
Other species may have a different range of hearing.[2] As a signal (Sound velocity level)
perceived by one of the major senses, sound is used by many Particle displacement ξ
species for detecting danger, navigation, predation, and Sound intensity I
communication. In Earth's atmosphere, water, and soil virtually any
Sound intensity level (SIL)
physical phenomenon, such as fire, rain, wind, surf, or earthquake,
produces (and is characterized by) its unique sounds. Many species, Sound power Pac
such as frogs, birds, marine and terrestrial mammals, have also Sound power level (SWL)
developed special organs to produce sound. In some species these Sound energy density E
became highly evolved to produce song and (in humans) speech. Sound energy flux q
Furthermore, humans have developed culture and technology (such Acoustic impedance Z
as music, telephony and radio) that allows them to generate, record,
Speed of sound c
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
Pa) and (1 atm
Pa), that is between 101323.6 and 101326.4 Pa. Such a tiny (relative to atmospheric)
variation in air pressure at an audio frequency 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
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
immediate soft tissue damage
rocket launch equipment acoustic tests
threshold of pain
hearing damage during short-term effect
jet engine, 100 m distant
jack hammer, 1 m distant / discotheque
hearing damage from long-term exposure
traffic noise on major road, 10 m distant
moving passenger car, 10 m distant
TV set – typical home level, 1 m distant
normal talking, 1 m distant
very calm room
quiet rustling leaves, calm human breathing
auditory threshold at 2 kHz – undamaged human
ears
RMS sound
pressure
Pa
50000
100
20
6–200
2
0.6
0.2–0.6
0.02–0.2
0.02
0.002–0.02
0.0002–0.0006
0.00006
sound pressure
level
dB re 20 µPa
approx. 185
approx. 165
134
approx. 120
110–140
approx. 100
approx. 85
80–90
60–80
approx. 60
40–60
20–30
10
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.
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:
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
.
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:
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:
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
Name
Other names
Abbreviation
Size
Semitones
Interval class
Just interval
Cents
Equal temperament
Just intonation
unison
P8
12
0
2:1
1200
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
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 (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
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.
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).
Different amplitude measurements of a sine wave
There are a few ways to formalize amplitude:
In the simple wave equation
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.
2.
3.
4.
The range between tonal and noiselike character.
The spectral envelope.
The time envelope in terms of rise, duration, and decay.
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 nonharmonic 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
Tonal character, usually pitched
Objective
Periodic sound
Noisy, with or without some tonal
character, including rustle noise
Coloration
Beginning/ending
Coloration glide or formant glide
Microintonation
Vibrato
Tremolo
Attack
Final sound
Noise, including random pulses characterized by the
rustle time (the mean interval between pulses)
Spectral envelope
Physical rise and decay time
Change of spectral envelope
Small change (one up and down) in frequency
Frequency modulation
Amplitude modulation
Prefix
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.
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
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
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:
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.
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 lownoise 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
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 pressuresensitive 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):
Omnidirectional
Subcardioid
Cardioid
Supercardioid
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
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
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
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, professionalquality 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 highimpedance 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
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
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;
black RCA/TS/TRS also mono (RCA/TS), stereo (TRS only),
or undefined/other
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
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
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.
XLR-LNE 2-pin socket and plug,
originally used for mains power
connections
Male and female XLR4
panel connectors
XLR5 Socket Female XLR6
panel connector
XLR3 connectors
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:
Pin
1
2
3


Function
Chassis ground (cable shield)
Normal polarity ("hot")
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.
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
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).
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 SPDIFformatted 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 signalcarrying 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
"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.
Left/Mono
White
Right
Red
Center
Green
Left surround
Blue
Analog audio
Right surround
Gray
Left back surround Brown
Right back surround Tan
Subwoofer
Purple
Digital audio
S/PDIF
Orange
Composite analog video
Composite
Yellow
Y
Green
Component analog video (YPbPr)
Pb
Blue
Pr
Red
R
Red
G
Green
Component analog video/VGA (RGB/HV)
B
Blue
H/Horizontal sync Yellow
V/Vertical sync White
TRS connector
TRS connector
"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
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.
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 nonstandard size.
Mono and stereo compatibility
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.
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
A jack plug breaks the contact of a normally closed switch.
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
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
Tip
Signal
Ground or No
Connection
Sleeve Ground
Ring
Unbalanced Input
Signal
Ground or No
Connection
Ground
Unbalanced
Insert
Send or Return
signal
Return or Send
signal
Ground
Balanced
Stereo
Left
channel
Right
Negative/"Cold"
channel
Ground
Ground
Positive/"Hot"
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 "halfclick" 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 linelevel 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
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 selfpowered microphone: usually an expensive dynamic microphone with internal batterypowered 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
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 tipring-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
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.
Structure
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.
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
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.