Now Hear That Home
We sell Loudspeakers, speaker and audio components. Our home theater products
include speakers, A/V processors, and amplifiers. Our professional processors
and speakers are used in recording and production studios.
High-end audio is a term used to describe equipment that is purported by the
manufacturers to be the best, regardless of the price.
Definition of 'high end'
High-end audio can refer to the build quality of the components, but more
specifically, refers to the ability to reproduce a recording with the highest
fidelity to the original performance that has been committed to the recording.
Typical qualitative attributes that are scaled by audiophile publications and
experts are:
accuracy vs. warmth
tonal color vs. speed
timbre
size of sound stage vs. depth (spatial origins)
clarity
pace
timing
A theoretically perfect high-end audio system would create the illusion of the
listener being present in the performance venue and with the musical performers
performing on stage. There would be no sonic signature that imparts any clue as
to the fact that the performance is a playback of a recording instead of
witnessing a live performance given by the actual musicians in the particular
performance venue. This is obviously more important with performances involving
acoustic instruments and without studio manipulations of vocals.
It is important to note that the term high-end is not always synonymous with
audiophile equipment
Professional recording studios
Professional recording studios seldom use high-end audio gear for mixing and
monitoring recording sessions. Instead, studios use players, amplifiers, signal
processors, and speakers that are built to very high standards. These speakers
are referred to as studio monitors and are specially crafted to produce very
accurate sound, reflecting exactly what is on the recording. Most high-end
speakers will tend to add color or tone shaping the music so that it sounds
"better". For this reason studio monitors must be used to ensure that changes
being made to the audio are accurately represented to the engineer.
Publications that interested parties can peruse include Stereophile (US), The
Absolute Sound (US), Hifi News (UK) and Hifi + (UK). Also the Web site Hi Fi
Wigwam (UK) has the best advice on audio equpiment from the forum users.
Costs
High-end audio equipment can be extremely expensive. It is sometimes referred to
as cost-no-object equipment. Owners of high-end audio tend to be either
audiophiles or conspicuous consumers. Audiophiles run the gamut from budget to
high-end in terms of equipment price range and are primarily concerned with the
quality of music reproduction (accuracy with personal preferences). However,
even though the retail price of the product may be high, regular components,
circuit boards and wires are often used inside. This gives the manufacturer very
high premiums, which is essential as these devices are not sold in large
quantities.
Snake oil business
The high-end market has became full of equipment that supposedly improves the
sound quality, even though such an effect is not physically possible and no
controlled blind listening tests have found any differences in sound. Yet, there
are numerous testimonials from people who have bought or otherwise tested these
items about how the products have improved the perceived audio quality. Some
audio reviewers have began talking of the listening feeling. Listening to the
equipment can be more pleasing and the perceived audio quality may in fact be
better when the listener knows that he is listening a high-end system down to
the last component.
There are not only such obvious hoaxes like premium power cables, magnetic
stones and speaker cable stands (not for organizing the cables, but just lifting
them off the floor), all marketed with pseudo-scientific claims about how they
improve the audio quality, but also products which might have some effect on the
audio, even if not always for the better.
Even serious products are primarily designed to look good, rather than trying to
maximize the audio quality. A typical high-end tube amplifier, for example,
always has its vacuum tubes exposed outside the case of the device, as the
glowing tubes add to the listening experience. This is a trade-off with audio
quality, as hiding the tubes inside the case, protected from electromagnetic
interference, would reduce the noise levels of the equipment (not necessarily
audibly, though).
Technology
Graphical representations (sonogram) of a sound wave: analog (red), 4-bit
digital (black).
Graphical representations (sonogram) of a sound wave: analog (red), 4-bit
digital (black).
The two main classes of sound recording technology are analog recording and
digital recording.
Analog recording is achieved by a small microphone diaphragm that can detect
changes in atmospheric pressure (acoustic sound waves) and record them as
graphic sound waves on a medium. The first of these recordings were called
sonograms and had no playback mechanism available. With an exclusively
mechanical phonograph, the analog conversion is in the form of sonogram grooves
carved by a stylus. Newer phonographs use electronics in the process. With
magnetic tape, the analog conversion is first in the form of electrical current
waves from the microphones conversion of diaphragm movement to electromagnetic
fluctuation (flux) that modulate an electric signal, and second of magnetic
particles drawn into sonogram-shaped clusters by flux from a tape head sensing
the electrical current changes. Analog sound reproduction is the reverse process
with a bigger loudspeaker diaphragm causing changes to atmospheric pressure to
form acoustic sound waves.
Digital recording and reproduction uses the same analog technologies, with
digitization of the sonographic data and signal, allowing it to be stored and
transmitted on a wider variety of media. The digital binary numeric data is a
representation of the periodic vector points in the raw analog acoustic data at
a sample rate most often too frequent for the human ear to distinguish
differences in quality. Digital recordings are not necessarily at a higher
sample rate, but are often considered higher quality because of less
interference from dust or electromagnetic interference in playback and less
mechanical deterioration from corrosion or mishandling the storage medium.
A loudspeaker, speaker, or speaker system is an electromechanical transducer
that converts an electrical signal to sound. The term loudspeaker can refer to
individual transducer devices (otherwise known as drivers), or to complete
systems consisting of an enclosure incorporating one or more drivers and
electrical filter components. Loudspeakers, just as with other electro-acoustic
transducers, are the most variable elements in an audio system and are
responsible for the greatest degree of audible differences between sound
systems.
To adequately reproduce a wide range of frequencies, most loudspeaker systems
require more than one driver, particularly for high sound pressure level or high
accuracy applications. Individual drivers are used to cover different frequency
ranges. The drivers are named subwoofers (very low frequencies), woofers (low
frequencies), mid-range speakers (middle frequencies), tweeters (high
frequencies) and sometimes supertweeters which are drivers optimized for higher
frequencies than a normal tweeter.
The terms for different speaker drivers differ depending on the application. In
2-way loudspeakers, there is usually no driver called "mid-range". Home stereos
use the designation "tweeter" for high frequencies whereas professional audio
systems for concerts typically designate all types of high frequency drivers as
"HF" or "highs" or "horns".
When multiple drivers are used in a system, a "filter network", called a
crossover, is used to separate the incoming signal into different frequency
bands appropriate for each driver. A loudspeaker system with n separate
frequency bands is described as "n-way speakers": a 2-way system will have
woofer and tweeter speakers; a 3-way system is either a combination of woofer,
mid-range and tweeter or subwoofer, woofer and tweeter.
History
Alexander Graham Bell patented the first electrical loudspeaker as part of his
telephone in 1876, which was followed in 1878 by an improved version from Ernst
Siemens. Nikola Tesla reportedly created a similar device in 1881, but was not
issued a patent.[1] During this time, Thomas Edison was issued a British patent
for a system using compressed air as an amplifying mechanism for his early
cylinder phonographs, but he ultimately settled for the familiar metal horn
driven by a membrane attached to the stylus. In 1898, Horace Short patented a
design for a loudspeaker driven by compressed air, then sold the rights to
Charles Parsons, who was issued several additional British patents before 1910.
A few companies, including Victor Talking Machine Company and Pathe, produced
record players using compressed-air loudspeakers. However, these designs were
significantly limited by their poor sound quality and their inability to
reproduce sound at low volume. Variants of the system were used for public
address applications, and more recently other variations have been used to test
space equipment resistance to the very loud sound and vibration levels that
launching rockets produce.
The modern design of moving-coil drivers was established by Oliver Lodge in
(1898)[2]. The moving coil principle was patented in 1924 by Chester W. Rice and
Edward W. Kellogg.
These first loudspeakers used electromagnets because large, powerful permanent
magnets were generally not available at a reasonable price. The coil of an
electromagnet, called a field coil, was energized by current through a second
pair of connections to the driver. This winding usually served a dual role,
acting also as a choke coil filtering the power supply of the amplifier to which
the loudspeaker was connected. AC ripple in the current was attenuated by the
action of passing through the choke coil; however, AC line frequencies tended to
modulate the audio signal being sent to the voice coil and added to the audible
hum of a powered-up sound reproduction device.
The quality of loudspeaker systems until the 1950s was poor. Continuous
developments in enclosure design and materials have led to significant audible
improvements. The most notable improvements in modern speakers are improvements
in cone materials, the introduction of higher temperature adhesives, improved
permanent magnet materials, improved measurement techniques, computer aided
design and finite element analysis.
Driver design
Cut-away view of a dynamic loudspeaker
Cut-away view of a dynamic loudspeaker
The most common type of driver uses a lightweight diaphragm connected to a rigid
basket, or frame, via flexible suspension that constrains a coil of fine wire to
move axially through a cylindrical magnetic gap. When an electrical signal is
applied to the voice coil, a magnetic field is created by the electric current
in the coil which thus becomes an electromagnet. The coil and the driver's
magnetic system interact, generating a mechanical force which causes the coil,
and so the attached cone, to move back and forth and so reproduce sound under
the control of the applied electrical signal coming from the amplifier. The
following is a description of the individual components of this type of
loudspeaker.
The diaphragm is usually manufactured with a cone or dome shaped profile. A
variety of different materials may be used, but the most common are paper,
plastic and metal. The ideal material would be stiff (to prevent uncontrolled
cone motions), light (to minimize starting force requirements) and well damped
(to reduce vibrations continuing after the signal has stopped). In practice, all
three of these criteria cannot be met simultaneously using existing materials,
and thus driver design involves tradeoffs. For example, paper is light and
typically well damped, but not stiff; metal can be made stiff and light, but it
is not usually well damped; plastic can be light, but typically the stiffer it
is made, the less well-damped it is. As a result, many cones are made of some
sort of composite material. This can be a matrix of fibers including Kevlar or
fiberglass, a layered or bonded sandwich construction, or simply a coating
applied to stiffen or damp a cone.
The basket or frame must be designed for rigidity to avoid deformation, which
will change the magnetic conditions in the magnet gap, and could even cause the
voice coil to rub against the walls of the magnetic gap. Baskets are typically
cast or stamped metal, although molded plastic baskets are becoming common,
especially for inexpensive drivers. The frame also plays a considerable role in
conducting heat away from the coil.
The suspension system keeps the coil centered in the gap and provides a
restoring force to make the speaker cone return to a neutral position after
moving. A typical suspension system consists of two parts: the "spider", which
connects the diaphragm or voice coil to the frame and provides the majority of
the restoring force; and the "surround", which helps center the coil/cone
assembly and allows free pistonic motion aligned with the magnetic gap. The
spider is usually made of a corrugated fabric disk, generally with a coating of
a material intended to improve mechanical properties. Unusually, a German
manufacturer, Klangfilm, used bakelite for spiders in some of its early drivers,
and another German company currently offers a spider made of wood. The surround
can be a roll of rubber or foam, or a ring of corrugated fabric (often coated),
attached to the outer circumference of the cone and to the frame. The choice of
suspension materials affects driver lifetime, especially in the case of foam
surrounds which are susceptible to aging and environmental damage.
The wire in a voice coil is usually made of copper, though aluminum, and rarely
silver, may be used. Voice coil wire cross sections can be circular,
rectangular, or hexagonal, giving varying amounts of wire volume coverage in the
magnetic gap space. The coil is oriented coaxially inside the gap, a small
circular volume (a hole, slot, or groove) in the magnetic structure within which
it can move back and forth. The gap establishes a concentrated magnetic field
between the two poles of a permanent magnet; the outside of the gap being one
pole and the center post (a.k.a., the pole-piece) being the other. The center
post and back-plate are sometimes a single piece called the yoke.
Modern driver magnets are almost always permanent and made of ceramic, ferrite,
Alnico, or, more recently, neodymium magnet. A current trend in design, due to
increases in transportation costs and a desire for smaller, lighter devices (as
in many home theater multi-speaker installations), is the use of neodymium
magnet instead of ferrite types. Very few manufacturers use electrically powered
field coils as was common in the earliest designs. The size and type of magnet
and details of the magnetic circuit differ, depending on design goals. For
instance, the shape of the pole piece affects the magnetic interaction between
the voice coil and the magnetic field, and is sometimes used to modify a
driver's behavior. As well, a 'shorting ring' or cap is sometimes used near the
magnetic gap to reduce adverse distortion effects of high current in the voice
coil.
Driver design, and the combination of one or more drivers into an enclosure to
make a speaker system, is both an art and science. Adjusting a design to improve
performance is done using magnetic, acoustic, mechanical, electrical, and
material science theory, high precision measurements, and the observations of
experienced listeners. Designers can use an anechoic chamber to ensure the
speaker can be measured independently of room effects, or any of several
electronic techniques which can, to some extent, replace such chambers. Some
developers eschew anechoic chambers in favor of specific standardized room
setups intended to simulate real-life listening conditions. A few of the issues
speaker and driver designers must confront are distortion, lobing, phase
effects, off axis response and crossover complications.
The fabrication of finished loudspeaker systems has become segmented, depending
largely on price, shipping costs, and weight limitations. High-end speaker
systems, which are heavier (and often larger) than economic shipping allows
outside local regions, are usually made in their target market area and can cost
$140,000 or more per pair.[3] The lowest-priced speaker systems and most drivers
are manufactured in China or other low-cost manufacturing locations. Although
the manufacture of drivers has become largely commoditized, the fabrication and
subsequent sale of finished speaker systems still carries high profits. Partly
for this reason, manufacturers are increasingly combining power amplifier
electronics (a typically lower profit item) with finished speaker systems to
create powered speakers with an overall higher market value.[citation needed]
Driver types
Exploded view of a dome tweeter
Exploded view of a dome tweeter
An audio engineering rule of thumb is that individual electrodynamic drivers
provide quality performance over at most about 3 octaves. Multiple drivers
(i.e., subwoofers, woofers, mid-range drivers, tweeters) are generally used in a
complete loudspeaker system to provide performance beyond 3 octaves.
Full range drivers
Full-range
A full-range driver is designed to have the widest frequency response possible,
despite the rule of thumb cited above. These drivers are small, typically 2 to 6
inches (5 to 16 cm) in diameter to permit reasonable high frequency response,
and carefully designed to give low distortion output at low frequencies, though
with reduced maximum output level. Full range drivers are a possible approach to
avoid degrading effects of multiple driver systems, caused by non-coincident
driver location and by crossover design and implementation issues. Those
favoring the full range driver approach claim a coherence of sound (said to be
due to the single source and a resulting lack of phase interference, and likely
to the lack of obscuring electrical crossover components) and feel the
disadvantages of restricted frequency bandwidth and reduced output power more
than compensated for. Another disadvantage is often a requirement for elaborate
cabinetry (i.e., transmission lines, horns, etc) to increase efficiency at low
frequencies to barely adequate levels by better matching the driver to the air
at those frequencies, thus increasing the output level at low frequencies.
Full range drivers often employ an additional cone called a whizzer: a small,
light cone attached to the joint between the voice coil and the primary cone.
The whizzer cone extends the high frequency response of the driver and broadens
its high frequency directivity, which would otherwise be greatly narrowed due to
the outer diameter cone material failing to keep up with the central voice coil
at higher frequencies. The main cone in a whizzer design is manufactured so as
to flex more in the outer diameter than in the center. The result is that the
main cone delivers low frequencies and the whizzer cone contributes most of the
higher frequencies. Since the whizzer cone is smaller than the main diaphragm,
output dispersion at high frequencies is improved relative to an equivalent
single larger diaphragm.
Another common use of single drivers is in devices not primarily intended for
high quality sound reproduction, such as computers, toys, clock radios, and
pocket sized music players. A single driver is less expensive than several, and
there is no need for a crossover network, further reducing cost. In this use,
high fidelity is at most a secondary consideration. Human hearing is able to
tolerate listening to a reduced bandwidth, and upper harmonic synthesis can be
used to 'fill in' missing bass tones that the driver is too small to usefully
reproduce.
Subwoofer
Subwoofer
A subwoofer is a woofer driver used only for the lowest part of the audio
spectrum: typically below 100-120 Hz. Because the intended range of frequencies
in these is limited, subwoofer system design is usually simpler in many respects
than for conventional loudspeakers, often consisting of a single subwoofer
driver enclosed in a suitable cabinet or enclosure.
To accurately reproduce very low bass notes without unwanted resonances (i.e.,
from cabinet panels), subwoofer systems must be solidly constructed and properly
braced; good ones are typically heavy. Many subwoofer systems include power
amplifiers and electronic filters, with additional controls relevant to low
frequency reproduction. These variants are known as "active subwoofers". Passive
subwoofers require external amplification.
Woofer
Woofer
A woofer is a driver that reproduces low frequencies. Some loudspeaker systems
use a woofer for the lowest frequencies, making it possible to avoid using a
subwoofer. Additionally, some loudspeakers use the woofer to handle middle
frequencies, eliminating the mid-range driver. This can be accomplished with the
selection of a tweeter that responds low enough combined with a woofer that
responds high enough that the two drivers add coherently in the middle
frequencies.
Mid-range driver
Mid-range speaker
A mid-range speaker is a loudspeaker driver which reproduces middle frequencies.
Mid-range drivers can be made of paper or composite materials, or be compression
drivers. If the mid-range driver is cone-shaped, it can be mounted on the front
baffle of a loudspeaker enclosure, or it can be mounted at the throat of a horn
for added output level and control of radiation pattern. If it is a compression
driver, it is invariably mated to a horn.
Tweeter
Tweeter
A tweeter is a high-frequency driver that typically reproduces the highest
frequency band of a loudspeaker. Many varieties of tweeter design exist, each
with differing abilities with regard to frequency response, output fidelity,
power handling, maximum output level, etc. Soft dome tweeters are widely found
in home stereo systems, and horn-loaded compression drivers are common in
professional sound reinforcement. Ribbon tweeters have gained popularity in
recent years, as their output power has been increased to levels useful for
professional sound reinforcement, and their pattern control is conveniently
shaped for concert sound.
Loudspeaker system design
Crossover
Audio crossover
A passive crossover
A passive crossover
An active crossover
An active crossover
Used in multi-driver speaker systems, the crossover is a device that separates
the input signal into different frequency ranges suited to each driver. Each
driver, therefore, receives the frequency range it was designed for, so the
distortion in each driver, and interference between the drivers, is reduced.
Crossovers can be passive or active. A passive crossover is an electronic
circuit using a combination of one or more resistors, inductors and non-polar
capacitors. These parts are formed into carefully designed networks, and placed
between the amplifier and the loudspeaker drivers to divide the amplifier's
signal into the necessary frequency bands before being delivered to the
individual drivers. Passive crossover circuits need no external power beyond the
audio signal itself. An active crossover is an electronic filter circuit which
divides the complete signal into individual frequency bands before
amplification, thus requiring one amplifier for each bandpass. The active
crossover requires an external power supply.
Passive crossovers are generally installed inside speaker boxes and are by far
the most common type of crossover for home and low power use. In car audio
systems, passive crossovers are often in a separate box due to the size of some
of the passive components used. Passive crossovers convert a non-trivial part of
the amplifier power they handle into heat, so when high power output is needed,
active crossovers are often used. Active crossovers allow more precise alignment
of phase and time between frequency bands; equivalently tight adjustment using
only passive components is a difficult engineering problem, in part because of
wide component tolerances and because of complex interactions between the
drivers themselves and the passive crossover components.
Many new loudspeaker designs have begun incorporating active crossover circuitry
and onboard amplification. Such designs typically require AC power and take low
level signal inputs instead of high level amplifier output connections. Ideally,
this approach offers the advantages of close alignment of phase between
frequency bands, active protection circuits to protect drivers, and virtually no
loss of amplifier power in long cable runs or passive crossover components.
Self-powered loudspeakers are being used in many applications such as
small-scale computer sound (for one listener) and large-scale concert sound
systems (for large halls full of listeners). Self-powered concert loudspeakers
provide the additional benefit of improved predictability in sound quality; the
touring concert sound engineer need not worry about customized crossover
settings in each venue changing the characteristics of a loudspeaker.
Enclosures
Loudspeaker enclosure
An unusual 3-way speaker system. The cabinet is narrow to reduce a diffraction
effect called the 'baffle step'.
An unusual 3-way speaker system. The cabinet is narrow to reduce a diffraction
effect called the 'baffle step'.
Most loudspeaker systems consist of drivers mounted in an enclosure, or cabinet.
The role of the enclosure is to provide a place to mount the drivers and to
prevent sound waves from the back of a driver from interfering destructively
with those from the front -- doing so typically causes cancellations (eg, comb
filtering) and significantly alters the level and quality sound at low
frequencies.
The simplest driver mount is a flat panel (ie, baffle) with the drivers mounted
in a hole in it. However, in this approach, frequencies with a wavelength longer
than the baffle dimensions are canceled out because the antiphase radiation from
the rear of the cone interferes with the radiation from the front. With an
infinitely large panel, this interference could be entirely prevented. A
sufficiently large sealed box can approach this behavior.[4][5].
Since panels of infinite dimensions are impractical, most enclosures function by
containing the rear radiation from the cone. A sealed enclosure prevents
transmission of the sound emitted from the rear of the loudspeaker by confining
the sound in a rigid and airtight box. Techniques used to reduce transmission of
sound through the walls of the cabinet include thicker cabinet walls, lossy wall
material, internal bracing, curved cabinet walls or more rarely visco-elastic
materials (eg, mineral loaded bitumen), or thin lead sheeting applied to
interior enclosure walls.
However, a rigid enclosure internally reflects sound which can then be
transmitted back through the loudspeaker cone, again resulting in degradation of
sound quality. This can be reduced by internal absorption using absorptive
materials (often called "damping") such as fiberglass, wool or synthetic fiber
batting within the enclosure. The internal shape of the enclosure can also be
designed to reduce this by reflecting sounds away from the loudspeaker diaphragm
where they may then be absorbed.
Other enclosure types alter the rear radiation so it can add constructively to
the output from the front of the cone. Designs that do this (including bass
reflex, passive radiators, transmission line, etc) are often used to extend the
effective low frequency response, and increase low frequency output of the
driver.
To make the transition between drivers as seamless as possible, system designers
have attempted to time-align (or phase adjust) the drivers by moving one or more
drivers forward or back, so that the acoustic center of each driver is in the
same vertical plane. This may also involve tilting the face speaker back, or
providing separate enclosure mounting for each driver, or, less commonly, using
electronic techniques to achieve the same effect. These attempts account for
some unusual cabinet designs.
Any speaker mounting scheme (including cabinets) will also cause diffraction,
causing peaks and dips in the frequency response. This is usually a problem at
higher frequencies where wavelengths are similar to, or smaller than, cabinet
dimensions. The effect can be minimized by rounding the front edges of the
cabinet, curving the cabinet itself, using a smaller or narrower enclosure,
choosing a strategic driver arrangement, or using absorptive material around a
driver.
Wiring connections
Five-way binding posts on a loudspeaker connected using banana plugs.
Five-way binding posts on a loudspeaker connected using banana plugs.
A 4 Ohm loudspeaker with two pairs of binding posts capable of accepting
bi-wiring after the removal of two metal straps
A 4 Ohm loudspeaker with two pairs of binding posts capable of accepting
bi-wiring after the removal of two metal straps
Most loudspeakers use two wiring points to connect to the source of the signal
(for example, to the audio amplifier or receiver). This is usually done using
binding posts, or spring clips on the back of the enclosure. If the wires for
left and right speakers (in a stereo setup) are not connected 'in phase' with
each other (the + and - connections on the speaker and amplifier should be
connected + to + and - to -) the loudspeakers will be out of polarity. Given
identical signals, motion in one cone will be in the opposite direction of the
other. This will typically cause monophonic material within a stereo recording
to be canceled out, reduced in level and made more difficult to localize, all
due to destructive interference of the sound waves. The cancellation effect is
most noticeable at frequencies where the speakers are separated by a quarter
wavelength or less; low frequencies are affected the most. This type of wiring
error doesn't damage speakers but isn't optimal.
Specifications
Specifications label on a loudspeaker
Specifications label on a loudspeaker
Speaker specifications generally include:
* Speaker or driver type (individual units only) – Full-range, woofer, tweeter
or mid-range.
* Size of individual drivers. For cone drivers, this number may be the outside
diameter of the frame, the diameter of the surround, or the diameter of the
cone. It may also be the distance from the center of one mounting hole to its
opposite. Voice coil diameter may also be specified. If the loudspeaker has a
compression horn driver, the diameter of the horn throat may be given.
* Rated Power – Nominal (or even continuous) power, and peak (or maximum
short-term) power a loudspeaker can handle (i.e., maximum input power before
thermally destroying the loudspeaker. It is never the sound output the
loudspeaker produces). A driver may be damaged at much less than its rated power
if driven past its mechanical limits at lower frequencies (e.g., by bass heavy
electronica or theatre organ music). Tweeters can also be damaged by amplifier
clipping (lots of high frequency energy in such cases) or by music, or sine wave
input at high frequencies. Each of these situations pass more energy to a
tweeter than it can survive without damage.
* Impedance – typically 4 Ω (ohms), 8 Ω, etc.
* Baffle or enclosure type (enclosed systems only) – Sealed, bass reflex, etc.
* Number of drivers (complete speaker systems only) – 2-way, 3-way, etc.
and optionally:
* Crossover frequency(ies) (multi-driver systems only) – The nominal frequency
boundaries of the signal division between drivers.
* Frequency response – The measured, or specified, output over a specified range
of frequencies for a constant input level varied across those frequencies. It
often includes a variance limit such as within "+/- 2.5 dB".
* Thiele/Small parameters (individual drivers only) – these include the driver's
Fs (resonance frequency), Qts (a driver's Q (more or less, its damping factor)
at resonant frequency), Vas (the equivalent air compliance volume of the
driver), etc.
* Sensitivity – The sound pressure level produced by a loudspeaker in a
non-reverberant environment, usually specified in dB, and measured at 1 meter
with an input of 1 watt or 2.83 volts, typically at one or more specified
frequencies. This rating is often inflated by manufacturers.
* Maximum SPL – The highest output the loudspeaker can manage, short of damage
or not exceeding a particular distortion level. This rating is often inflated by
manufacturers and is commonly given without reference to frequency range or
distortion level.
Electrical characteristics of a dynamic loudspeaker
Electrical characteristics of a dynamic loudspeaker
The load a driver presents to an amplifier consists of a complex electrical
impedance -- a combination of resistance, and both capacitive and inductive
reactance, which combines properties of the driver, its mechanical motion,
effects of crossover components (if any are in the signal path between amplifier
and driver), and effects of air loading on the driver as modified by the
enclosure and its environment. Most amplifiers output specifications are given
at a specific power into an ideal resistive load. However, a loudspeaker does
not really have a constant resistance across its frequency range. Instead, the
voice coil is inductive, the driver has mechanical resonances, the enclosure
changes the driver's electrical and mechanical characteristics, and a passive
crossover between the drivers and the amplifier contributes its own variations.
The result is a load resistance which varies fairly widely with frequency, and
usually a varying phase relationship between voltage and current as well, also
changing with frequency.
Electromechanical measurements
Fully characterizing the sound output quality of a loudspeaker driver or system
in words is essentially impossible. Objective measurements provide information
about several aspects of performance, so informed comparisons and improvements
can be made. Examples of typical measurements are: amplitude and phase
characteristics vs. frequency; impulse response under one or more conditions
(eg, square waves, sine wave bursts, ...); directivity vs. frequency (eg,
horizontally, vertically, spherically, ...); harmonic and intermodulation
distortion vs. SPL output using any of several test signals; stored energy (ie,
'ringing') at various frequencies; impedance vs. frequency and small signal vs.
large signal performance. Most of these measurements require relatively
expensive equipment to perform and good judgement, but the raw sound pressure
level output is rather easier to report and so is often the only specified
value, sometimes in misleadingly exact terms. The sound pressure level (SPL) a
loudspeaker produces is measured in decibels (dBspl).
Efficiency vs. sensitivity
Loudspeaker efficiency is defined as the sound power output divided by the
electrical power input. Most loudspeakers are actually very inefficient
transducers; about 1% of the electrical energy sent by an amplifier to a typical
home loudspeaker is converted to the acoustic energy we can hear. The remainder
is converted to heat, mostly in the voice coil and magnet assembly. The main
reason for this is the difficulty of achieving proper impedance matching between
the acoustic impedance of the drive unit and that of the air into which it is
radiating. The efficiency of loudspeaker drivers varies with frequency as well.
For instance, the output of a woofer driver decreases as the input frequency
decreases.
Driver ratings based on the SPL for a given input are called sensitivity ratings
and are notionally similar to efficiency. Sensitivity is usually defined as so
many decibels at 1 W electrical input, measured at 1 meter, often at a single
frequency. The voltage used is often 2.83 VRMS, which is 1 watt into an 8 Ω
(nominal) speaker impedance (approximately true for many speaker systems).
Measurements taken with this reference are quoted as dB with 2.83 V @ 1 m.
The sound pressure output is measured at (or mathematically scaled to be
equivalent to a measurement taken at) one meter from the loudspeaker and on-axis
or directly in front of it under the condition that the loudspeaker is radiating
into an infinitely large space and mounted on an infinite baffle. Clearly then,
sensitivity does not correlate precisely with efficiency, as it also depends on
the directivity of the driver being tested and the acoustic environment in front
of the actual loudspeaker. For example, a cheerleader's horn produces more sound
output in the direction it is pointed, by concentrating sound waves from the
cheerleader in one direction, and thus "focusing" them. The horn also improves
the impedance matching between voice and the air, which produces more acoustic
power for a given speaker power. In some cases, impedance matching (via careful
enclosure design) will allow the speaker to produce more power.
* Typical home loudspeakers have sensitivities of about 85 to 95 dB for 1 W @ 1
m - an efficiency of 0.5-4%.
* Sound reinforcement and public address loudspeakers have sensitivities of
perhaps 95 to 102 dB for 1 W @ 1 m - an efficiency of 4-10%.
* Rock concert, stadium PA, marine hailing, etc speakers generally have higher
sensitivities of 103 to 110 dB for 1 W @ 1 m - an efficiency of 10-20%.
A driver with a higher maximum power rating cannot necessarily be driven to
louder levels than a lower rated one, since sensitivity and power handling are
largely independent properties. In the examples that follow, assume for
simplicity that the drivers being compared have the same electrical impedance,
are operated at the same frequency which is within both driver's respective pass
bands, and that power compression and distortion are low. For the first example,
a speaker 3 dB more sensitive than another will produce double the sound
pressure level (or be 3 dB louder) for the same power input. Thus a 100 W driver
("A") rated at 92 dB for 1 W @ 1 m sensitivity will output twice as much
acoustic power as a 200 W driver ("B") rated at 89 dB for 1 W @ 1 m when both
are driven with 100 W of input power. For this particular example, when driven
at 100 W, speaker A will produce the same SPL, or loudness, speaker B would
produce with 200 W input. Thus a 3 dB increase in sensitivity of the speaker
means that it will need half the amplifier power to achieve a given SPL. This
translates into a smaller, less complex power amplifier and often to reduced
overall cost.
It is not possible to combine high efficiency, especially at low frequencies,
with compact enclosure size, and adequate low frequency response. One can, more
or less, only choose two of the three parameters when designing a speaker
system. So, for example, if extended low frequency performance and a small box
size are important, one must accept low efficiency.[6] This rule of thumb is
sometimes called Hoffman's Iron Law (after J. A. Hoffman, the H in KLH).[7]
Listening environment
Room acoustics
The interaction of a loudspeaker system with its environment is complex and is
largely out of the loudspeaker designer's control. Most listening rooms present
a more or less reflective environment, depending on size, shape, volume, and
furnishings. This means the sound reaching a listener's ears consists not only
of sound directly from the speaker system, but also the same sound delayed by
traveling to and from (and being modified by) one or more surfaces. These
reflected sound waves, when added to the direct sound, cause cancellation and
addition at assorted frequencies (eg, from resonant room modes), thus changing
the timbre and character of the sound at the listener's ears. The human brain is
very sensitive to small variations, including some of these, and this is part of
the reason why a loudspeaker system sounds different at different listening
positions or in different rooms.
A significant factor in the sound of a loudspeaker system is the amount of
absorption and diffusion present in the environment. Clapping one's hands in a
typical empty room, without draperies or carpet, will produce a zippy, fluttery
echo which is due both to a lack of absorption and to reverberation (that is,
repeated echoes) from flat reflective walls, floor, and ceiling. The addition of
hard surfaced furniture, wall hangings, shelving and even baroque plaster
ceiling decoration, will change the echoes, due primarily to the diffusion
caused by somewhat reflective objects with shapes and surfaces having sizes on
the order of the sound wavelengths being diffused. This somewhat breaks up the
simple reflections otherwise caused by bare flat surfaces, and spreads the
reflected energy of an incident wave over a larger angle on reflection.
Placement
In a typical rectangular listening room, the hard, parallel surfaces of the
walls, floor and ceiling cause primary acoustic resonance nodes in each of the
three dimensions: left-right, up-down and forward-backward.[8] Furthermore,
there are more complex resonance modes involving three, four, five and even all
six boundary surfaces combining to create standing waves. Low frequencies excite
these modes the most, since long wavelengths are not much affected by furniture
compositions or placement. The mode spacing is critical, especially in small and
medium size rooms like recording studios, home theaters and broadcast studios.
The proximity of the loudspeakers to room boundaries affects how strongly the
resonances are excited as well as affecting the relative strength at each
frequency. The location of the listener is critical, too, as a position near a
boundary can have a great effect on the perceived balance of frequencies. This
is because standing wave patterns are most easily heard in these locations and
at lower frequencies, below the Schroeder frequency - typically around 200-300
Hz, depending on room size.
Directivity
Acousticians, in studying the radiation of sound sources have developed some
concepts important to understanding how loudspeakers are perceived. The simplest
possible radiating source is a point source, sometimes called a simple source.
An ideal point source is an infinitesimally small point radiating sound. It may
be easier to imagine a tiny pulsating sphere, uniformly increasing and
decreasing in diameter, sending out sound waves in all directions equally,
independent of frequency.
Any object radiating sound, including a loudspeaker system, can be thought of as
being composed of combinations of such simple point sources. The radiation
pattern of a combination of point sources will not be the same as for a single
source, but rather will depend on the distance and orientation between the
sources, the position relative to them from which the listener hears the
combination, and the frequency of the sound involved. Using geometry and
calculus, some simple combinations of sources are easily solved; others are not.
One simple combination is two simple sources separated by a distance and
vibrating out of phase, one miniature sphere expanding while the other is
contracting. The pair is known as a doublet, or dipole, and the radiation of
this combination is similar to that of a very small dynamic loudspeaker
operating without a baffle. The directivity of a dipole is a figure 8 shape with
maximum output along a vector which connects the two sources and minimums to the
sides when the observing point is equidistant from the two sources, where the
sum of the positive and negative waves cancel each other. While most drivers are
dipoles, depending on the enclosure to which they are attached, they may radiate
as monopoles, dipoles (or bipoles). If mounted on a finite baffle, and these out
of phase waves allowed to interact, dipole peaks and nulls in the frequency
response result. When the rear radiation is absorbed or trapped in a box, the
diaphragm becomes a monopole radiator. Bipolar speakers, made by mounting
in-phase monopoles (both moving out of or into the box in unison) on opposite
sides of a box, are a method of approaching omnidirectional radiation patterns.
Polar plots of a four-driver industrial columnar public address loudspeaker
taken at six frequencies. Note how the pattern is nearly omnidirectional at low
frequencies, converging to a wide fan-shaped pattern at 1 kHz, then separating
into lobes and getting weaker at higher frequencies
Polar plots of a four-driver industrial columnar public address loudspeaker
taken at six frequencies. Note how the pattern is nearly omnidirectional at low
frequencies, converging to a wide fan-shaped pattern at 1 kHz, then separating
into lobes and getting weaker at higher frequencies[9]
In real life, individual drivers are actually complex 3D shapes such as cones
and domes, and they are placed on a baffle for various reasons. A mathematical
expression for the directivity of a complex shape, based on modeling
combinations of point sources, is usually not possible, but in the farfield, the
directivity of a loudspeaker with a circular diaphragm will be close to that of
a flat circular piston, so it can be used as an illustrative simplification for
discussion. As a simple example of the mathematical physics involved, consider
the following: the formula for farfield directivity of a flat circular piston in
an infinite baffle is p(\theta) = \frac{p_0 J_1(k_a \sin \theta)}{k_a \sin
\theta} where k_a=\frac{2\pi a}{\lambda}, p0 is the pressure on axis, a is the
piston radius, λ is the wavelength (i.e. \lambda = \frac{c}{f} =
\frac{\text{speed of sound}}{\text{frequency}}) θ is the angle off axis and J1
is the Bessel function of the first kind.
A planar source will radiate sound uniformly for low frequencies whose
wavelength is shorter than the dimensions of the planar source, and as frequency
increases, the sound from such a source will be focused into an increasingly
narrower angle. The smaller the driver, the higher the frequency where this
narrowing of directivity occurs. Even if the diaphragm is not perfectly
circular, this effect occurs such that larger sources are more directive.
Several loudspeaker designs have been built which have approximately this
behavior. Most are electrostatic or planar magnetic designs.
Various manufacturers use different driver mounting arrangements to create a
specific type of sound field in the space for which they are designed. The
resulting radiation patterns may be intended to more closely simulate the way
sound is produced by real instruments, or simply create a controlled energy
distribution from the input signal (some using this approach are called
monitors, as they are useful in checking the signal just recorded in a studio).
An example of the first is a room corner system with many small drivers on the
surface of a 1/8 sphere. A system design of this type was patented by, and
actually produced commercially, by Professor Amar Bose -- the 2201. Later Bose
models have deliberately emphasized production of both direct and reflected
sound by the loudspeaker itself, regardless of its environment. The designs are
controversial in high fidelity circles, but have proven commercially successful.
Several other manufacturers' designs follow similar principles.
Directivity is an important issue because it affects the frequency balance of
sound a listener hears, and also the interaction of the speaker system with the
room and its contents. A speaker which is very directive (ie, on an axis
perpendicular to the speaker face) may result in a reverberant field lacking in
high frequencies, giving the impression the speaker is deficient in treble even
though it measures well on axis (eg, "flat" across the entire frequency range).
Speakers with very wide, or rapidly increasing directivity at high frequencies,
can give the impression that there is too much treble (if the listener is on
axis) or too little (if the listener is off axis). This is part of the reason
why on-axis frequency response measurement is not a complete characterization of
the sound of a given loudspeaker.
Other driver designs
Other types of drivers which depart from the most commonly used direct radiating
electro-dynamic driver mounted in an enclosure include:
Horn loudspeakers
A three-way loudspeaker that uses horns in front of each of the three drivers: a
shallow horn for the tweeter, a long, straight horn for mid frequencies and a
folded horn for the woofer
A three-way loudspeaker that uses horns in front of each of the three drivers: a
shallow horn for the tweeter, a long, straight horn for mid frequencies and a
folded horn for the woofer
Horn speaker
Horn speakers are the oldest form of loudspeaker system, having been used from
very early on for cylinder recording players. They use a shaped waveguide in
front of or behind the driver to increase the directivity of the loudspeaker and
to transform a small diameter, high pressure condition at the driver cone
surface to a large diameter, low pressure condition at the mouth of the horn.
This increases the sensitivity of the loudspeaker and focuses the sound over a
narrower area. The size of the throat, mouth, the length of the horn, as well as
the area expansion rate along it must be carefully chosen to match the drive to
properly provide this transforming function over a range of frequencies (every
horn performs poorly outside its acoustic limits, at both high and low
frequencies). The length and cross-sectional mouth area required to create a
bass or sub-bass horn require a horn many feet long. 'Folded' horns can reduce
the total size, but compel designers to make compromises and accept increased
complication such as cost and construction. Some horn designs not only fold the
low frequency horn, but use the walls in a room corner as an extension of the
horn mouth. In the late 1940s, horns whose mouths took up much of a room wall
were not unknown amongst hi-fi fans. Room sized installations became much less
acceptable when two or more were required.
A horn loaded speaker can have a sensitivity as high as 110 dB @ 2.83 volts (1
watt @ 8 ohms) @ 1 meter. This is a hundredfold increase in output compared to a
speaker rated at 90 dB sensitivity, and is invaluable in applications where high
sound levels are required or amplifier power is limited.
Piezoelectric speakers
Piezoelectric speakers are frequently used as beepers in watches and other
electronic devices, and are sometimes used as tweeters in less-expensive speaker
systems, such as computer speakers and portable radios. Piezoelectric speakers
have several advantages over conventional loudspeakers: they are resistant to
overloads which would normally destroy most high frequency drivers, and they can
be used without a crossover due to their electrical properties. There are also
disadvantages: some amplifiers can oscillate when driving capacitive loads like
most piezoelectrics, which results in distortion or damage to the amplifier.
Additionally, their frequency response, in most cases, is inferior to that of
other technologies. This is why they are generally used in single frequency
(beeper) or non-critical applications.
Piezoelectric speakers can have extended high frequency output, and this is
useful in some specialized circumstances; for instance, sonar applications in
which piezoelectric variants are used as both output devices (generating
underwater sound) and as input devices (acting as the sensing components of
underwater microphones). They have advantages in these applications, not the
least of which is simple and solid state construction which resists the effects
of seawater better than, say, a ribbon based device would.
Electrostatic loudspeakers
Electrostatic loudspeaker
Electrostatic loudspeakers use a high voltage electric field (rather than a
magnetic field) to drive a thin membrane between two perforated conductive
plates called stators. Because they are driven over the entire membrane surface
rather than from a small voice coil, they can provide a more linear and lower
distortion response than dynamic drivers. They have the disadvantage that the
diaphragm excursion is severely limited because of practical construction
limitations. The further apart the stators are positioned, the higher the
voltage must be to achieve acceptable efficiency, which increases the tendency
for attracting dust and producing electrical arcs. For many years electrostatic
loudspeakers had a reputation as a generally unreliable and occasionally
dangerous product. Arcing remains a potential problem with current technologies,
especially when the panels are allowed to collect dust or dirt, or when driven
with high signal levels.
Electrostatics are inherently dipole radiators and due to the thin flexible
membrane cannot be used in enclosures to reduce low frequency cancellation as
with common cone drivers. Due to this and the low excursion capability, full
range electrostatic loudspeakers are large by nature, and even so are not
outstanding performers at the lowest frequencies. To reduce the size of
commercial products, they are often used as a high frequency driver in
combination with a conventional dynamic driver which handles the bass
frequencies.
Ribbon and planar magnetic loudspeakers
A ribbon speaker consists of a thin metal-film ribbon suspended in a magnetic
field. The electrical signal is applied to the ribbon which moves with it, thus
creating the sound. The advantage of a ribbon driver is that the ribbon has very
little mass; thus, it can accelerate very quickly, yielding very good
high-frequency response. Ribbon loudspeakers are often very fragile -- some can
be torn by a strong gust of air. Most ribbon tweeters emit sound in a dipole
pattern; a very few have backings which limit the dipole radiation pattern.
Above and below the ends of the more or less rectangular ribbon, there is less
audible output due to phase cancellation, but the precise amount of directivity
depends on ribbon length. Ribbon designs generally require exceptionally
powerful magnets which make them costly to manufacture. Ribbons have a very low
resistance that most amplifiers cannot drive directly. As a result, a step down
transformer is typically used to increase the current through the ribbon. The
amplifier "sees" a load that is the ribbon's resistance times the transformer
turns ratio squared. The transformer must be carefully designed so that its
frequency response and parasitic losses do not degrade the sound, further
increasing cost and complication relative to conventional designs.
Planar magnetic speakers (having printed or embedded conductors on a flat
diaphragm) are sometimes described as "ribbons", but are not truly ribbon
speakers. The term planar is generally reserved for speakers which have roughly
rectangular shaped flat surfaces that radiate in a bipolar (i.e., front and
back) manner. Planar magnetic speakers consist of a flexible membrane with a
voice coil printed or mounted on it. The current flowing through the coil
interacts with the magnetic field of carefully placed magnets on either side of
the diaphragm, causing the membrane to vibrate more or less uniformly and
without much bending or wrinkling. The driving force covers a large percentage
of the membrane surface and reduces resonance problems inherent in coil-driven
flat diaphragms.
Bending wave loudspeakers
Bending wave transducers use a diaphragm that is intentionally flexible. The
rigidity of the material increases from the center to the outside. Short
wavelengths radiate primarily from the inner area, while longer waves reach the
edge of the speaker. To prevent reflections from the outside back into the
center, long waves are absorbed by a surrounding damper. Such transducers can
cover a wide frequency range (80 Hz to 35,000 Hz) and have been promoted as
being close to an ideal point sound source.[10][11] This uncommon approach is
currently being taken by only two manufacturers, in very different arrangements.
Flat panel loudspeakers
There have been many attempts to reduce the size of speaker systems, or
alternatively to make them less obvious. One such attempt was the development of
voice coils mounted to flat panels to act as sound sources. These can then be
made in a neutral color and hung on walls where they will be less noticeable
than many speakers, or can be deliberately painted with patterns in which case
they can function decoratively. There are two related problems with flat panel
techniques: first, a flat panel is necessarily more flexible than a cone shape
in the same material, and therefore will move as a single unit even less, and
second, resonances in the panel are difficult to control, leading to
considerable distortions. Some progress has been made using such lightweight,
rigid, materials as Styrofoam, and there have been several flat panel systems
commercially produced in recent years.
Distributed mode loudspeakers
A newer implementation of the flat panel speaker system involves an
intentionally flexible panel and an "exciter", mounted off-center in a location
such that it excites the panel to vibrate, but with minimal resonances. Speakers
using such techniques can reproduce sound with a wide directivity pattern
(paradoxically somewhat like a point source) and have been used in some computer
speaker designs and bookshelf loudspeakers.[12]
Heil air motion transducers
Dr. Oskar Heil invented the air motion transducer in the 1960s. In this
approach, a pleated diaphragm is mounted in a magnetic field and forced to close
and open under control of a music signal. Air is forced from between the pleats
in accordance with the imposed signal, generating sound. The drivers are less
fragile than ribbons and considerably more efficient (and able to produce higher
absolute output levels) than ribbon, electrostatic, or planar magnetic tweeter
designs.
ESS, a California manufacturer, licensed the design, employed Dr. Heil, and
produced a range of speaker systems using his tweeters during the 1970s and
1980s. Radio Shack, a large US retail store chain, also sold speaker systems
using such tweeters for a time. At present, there are two manufacturers of these
drivers, both in Germany, one of which produces a range of high end professional
speakers using tweeters and midrange drivers based on the technology.
Plasma arc speakers
Plasma arc loudspeaker
Plasma arc loudspeakers use electrical plasma as a radiating element. Since
plasma has minimal mass, but is charged and therefore can be manipulated by an
electric field, the result is a very linear output at frequencies far higher
than the audible range. Problems of maintenance and reliability for this
approach tend to make it unsuitable for mass market use. In 1978 Dr. Alan Hill
of the Los Alamos National Laboratory designed the Hill Plasmatronics, an $8000
tweeter whose plasma was generated from helium gas.[13] This avoided the ozone
and nitrous oxide produced by RF decomposition of air in an earlier generation
of plasma tweeters made by the pioneering DuKane Corporation, who produced the
Ionovac (marketed as the Ionofane in the UK) during the 1950s. Currently, there
remain a few manufacturers in Germany, and a do it yourself design has been
published.
A less expensive variation on this theme is the use of a flame for the driver,
as flames contain ionized (electrically charged) gases.[14]
Digital speakers
Digital speakers
Digital speakers have been the subject of experiments by Bell Labs as far back
as the 1920s. The design is simple; each bit drives an independent speaker
driver. Increasingly significant bits drive speakers of twice the area of the
previous (often in a ring around the previous driver). A value of "1" causes
that driver to be driven to full amplitude; a value of "0" causes it to be
completely shut off.
There are two problems with this design which have led to it being abandoned as
impractical for the present. First, for a reasonable number of bits (required
for adequate sound reproduction quality), the size of the system becomes very
large. Secondly, due to analog digital conversion, the effect of aliasing is
unavoidable, so that the audio output is "reflected" at equal amplitude in the
frequency domain, on the other side of the sampling frequency, causing an
unacceptably high level of ultrasonics to accompany the desired output.
The term "digital" or "digital-ready" is often used for marketing purposes on
speakers or headphones, but these systems are not digital in the sense described
above. Rather, this is a somewhat deceptive marketing tactic, in which the
manufacturer is trying to capitalize on the popularity of digital sound
recordings and equipment.
Early developments
Music sample:
*
Au Clair de la Lune
Play sound
This 1860 phonautogram by Edouard-Leon Scott is the earliest known recorded
human voice.
* Problems playing the files? See media help.
The automatic reproduction of music can be traced back as far as the 14th
century, when Flanders introduced a mechanical bell-ringer controlled by a
rotating cylinder. Similar designs appeared in barrel organs (15th century),
musical clocks (1598), barrel pianos (1805), and musical boxes (1815). All of
these machines could play stored music, but they could not play arbitrary
sounds, could not record a live performance, and were limited by the physical
size of the medium. The first device that could record sound mechanically (but
could not play it back) was the phonautograph, developed in 1857 by Edouard-Leon
Scott. One of his paper recordings of Au Clair de la Lune, a French folk song,
was digitally converted to sound in 2008. It is believed to be the oldest
existing recording of a recognisable human voice.[1]. Since the above recording
was recovered the same team have since recovered a recording of a 435 Hz tuning
fork (at that time the French standard concert pitch for A' - now 440 Hz). The
tuning fork is barely audible. This second recording has thus become the oldest
known recording of a recognisable sound.
The player piano, first demonstrated in 1876, used a punched paper scroll that
could store an arbitrarily long piece of music. This piano roll moved over a
device known as the 'tracker bar', which first had 58 holes, was expanded to 65
and then was upgraded to 88 holes (generally, one for each piano key). When a
perforation passed over the hole, the note sounded. Piano rolls were the first
stored music medium that could be mass-produced, although the hardware to play
them was much too expensive for personal use. Technology to record a live
performance onto a piano roll was not developed until 1904. Piano rolls have
been in continuous mass production since around 1898.[citation needed] A 1908
U.S. Supreme Court copyright case noted that, in 1902 alone, there were between
70,000 and 75,000 player pianos manufactured, and between 1,000,000 and
1,500,000 piano rolls produced.[2] The use of piano rolls began to decline in
the 1920s although one type is still being made today. The fairground organ,
developed in 1892, used a similar system of accordion-folded punched cardboard
books.
Phonograph cylinder
Music sample:
*
"Kham Hom" ("Sweet Words")
Play sound
Phonograph cylinder recording of Siamese (Thai) musicians visiting Berlin,
Germany in 1900.
* Problems playing the files? See media help.
The first practical sound recording and reproduction device was the mechanical
phonograph cylinder, invented by Thomas Edison in 1877 and patented in 1878.[3]
The invention soon spread across the globe and over the next two decades the
commercial recording, distribution and sale of sound recordings became a growing
new international industry, with the most popular titles selling millions of
units by the early 1900s. The development of mass-production techniques enabled
cylinder recordings to become a major new consumer item in industrial countries
and the cylinder was the main consumer format from the late 1880s until around
1910.
Disc phonograph
The next major technical development was the invention of the gramophone disc,
generally credited to Emile Berliner and commercially introduced in the United
States in 1889.
Discs were easier to manufacture, transport and store, and they had the
additional benefit of being louder (marginally) than cylinders, which by
necessity, were single-sided. Sales of the Gramophone record overtook the
cylinder ca. 1910, and by the end of World War I the disc had become the
dominant commercial recording format. In various permutations, the audio disc
format became the primary medium for consumer sound recordings until the end of
the 20th century, and the double-sided 78 rpm shellac disc was the standard
consumer music format from the early 1910s to the late 1950s.
Although there was no universally accepted speed, and various companies offered
discs that played at several different speeds, the major recording companies
eventually settled on a de facto industry standard of nominally 78 revolutions
per minute, though the actual speed differed between America and the rest of the
world. The specified speed was 78.26 rpm in America and 77.92 rpm throughout the
rest of the world (this was related to the speed of a mains-driven synchronous
motor). The nominal speed of the disc format gave rise to its common nickname,
the "seventy-eight" (though not until other speeds had become available).
Discs were made of shellac or similar brittle plastic like materials, played
with needles made from a variety of materials including mild steel, thorn and
even sapphire. Discs had a distinctly limited playing life which was heavily
dependent on how they were reproduced.
The earlier, purely acoustic methods of recording had limited sensitivity and
frequency range. Mid-frequency range notes could be recorded but very low and
very high frequencies could not. Instruments such as the violin transferred
poorly to disc; however this was partially solved by retrofitting a conical horn
to the sound box of the violin. The horn was no longer required once electrical
recording was developed.
The vinyl microgroove record was introduced in the late 1940s, and the two main
vinyl formats -- the 7-inch single turning at 45 rpm and the 12-inch LP
(long-playing) record turning at 33 1/3 rpm -- had totally replaced the 78 rpm
shellac (sometimes vinyl) disc by the end of the 1950s. Vinyl offered improved
performance, both in stamping and in playback, and came to be generally played
with polished diamond styli, and when played properly (precise tracking weight,
etc.) offered longer life. Vinyl records were, over-optimistically, advertised
as "unbreakable". They were not, but were much less brittle and breakable than
shellac. Nearly all were tinted black, but some were colored, as red, swirled,
translucent, etc.
Electrical recording
Sound recording began as a mechanical process and remained so until the early
1920s (with the exception of the 1899 Telegraphone) when a string of
groundbreaking inventions in the field of electronics revolutionised sound
recording and the young recording industry. These included sound transducers
such as microphones and loudspeakers, and various electronic devices such as the
mixing desk, designed for the amplification and modification of electrical sound
signals.
After the Edison phonograph itself, arguably the most significant advances in
sound recording were the electronic systems invented by two American scientists
between 1900 and 1924.
In 1906 Lee De Forest invented the "Audion" triode vacuum-tube, electronic
valve, which could greatly amplify weak electrical signals, (one early use was
to amplify long distance telephone in 1915) which became the basis of all
subsequent electrical sound systems until the invention of the transistor. The
valve was quickly followed by the invention of the Regenerative circuit,
Super-Regenerative circuit and the Superheterodyne receiver circuit, all of
which were invented and patented by the young electronics genius Edwin Armstrong
between 1914 and 1922. Armstrong's inventions made higher fidelity electrical
sound recording and reproduction a practical reality, facilitating the
development of the electronic amplifier and many other devices; after 1925 these
systems had become standard in the recording and radio industry.
While E. H. Armstrong published studies about the fundamental operation of the
triode vacuum tube before World War I, scientists at Bell Telephone Laboratories
achieved their own understanding about the triode and were utilizing the audion
as a repeater in weak telephone circuits. By 1925 it was possible to place a
long distance telephone call with these repeaters between New York and San
Francisco in 20 minutes, both parties being clearly heard.
With this technical prowess, Joseph P. Maxfield and Henry C. Harrison from Bell
Telephone Laboratories were skilled in using mechanical analogs of electrical
circuits and applied these principles to sound recording and reproduction.[4]
They were ready to demonstrate their results by 1924 using the Wente condenser
microphone and the vacuum tube amplifier to drive the "rubber line" wax recorder
to cut a master audio disc. [5]
Meanwhile, radio continued to develop. Armstrong's groundbreaking inventions
(including FM radio) also made possible the broadcasting of long-range,
high-quality radio transmissions of voice and music. The importance of
Armstong's Superheterodyne circuit cannot be over-estimated -- it is the central
component of almost all analog amplification and both analog and digital
radio-frequency transmitter and receiver devices to this day.
American singer Jan Peerce recording in the 1940s.
American singer Jan Peerce recording in the 1940s.
Beginning during World War One, experiments were undertaken in the United States
and Great Britain to reproduce among other things, the sound of a Submarine
(u-boat) for training purposes. The acoustical recordings of that time proved
entirely unable to reproduce the sounds, and other methods were actively sought.
Radio had developed independently to this point, and now Bell Laboritories
sought a marriage of the two disparate technologies, greater than the two
separately. The first experiments were not very promising, but by 1920 greater
sound fidelity was achieved using the electrical system than had ever been
realized acoustically. One early recording made without fanfare or announcement
was the dedication of the Tomb of the Unknown Soldier at Arlington Cemetery.
By early 1924 such dramatic progress had been made, that Bell Labs arranged a
demonstration for the leading recording companies, Victor Talking Machine, and
Columbia Phonograph Co's.
Columbia, always in financial straits, could not afford it, and Victor,
essentially leaderless since the mental collapse of founder Eldridge Johnson,
left the demonstration without comment. English Columbia, by then a separate
company, got hold of a test pressing made by Pathe' from these sessions, and
realized the immediate and urgent need to have the new system. Bell was only
offering its method to United States Companies, and to circumvent this, Managing
Director Louis Sterling of English Columbia, bought his once parent company, and
signed up for electrical recording. When Victor Talking Machine was apprised of
the Columbia deal, they too quickly signed. Columbia made its first electrical
recordings on February 25, 1925 with Victor following a few weeks later. The two
then agreed privately to "be quiet" until November 1925, by which time enough
electrical repertory would be available.
Other recording formats
This period also saw several other historic developments including the
introduction of the first practical magnetic sound recording system, the
magnetic wire recorder, which was based on the work of Danish inventor Valdemar
Poulsen. Magnetic wire recorders were effective, but the sound quality was poor,
so between the wars they were primarily used for voice recording and marketed as
business dictating machines.
In the 1930s radio pioneer Guglielmo Marconi developed a system of magnetic
sound recording using steel tape. This was the same material used to make razor
blades, and not surprisingly the fearsome Marconi-Stille recorders were
considered so dangerous that technicians had to operate them from another room
for safety. Because of the high recording speeds required, they used enormous
reels about one metre in diameter, and the thin tape frequently broke, sending
jagged lengths of razor steel flying around the studio.
The K1 Magnetophon was the first practical tape recorder, developed by AEG in
Germany in 1935.
The other major invention in sound recording in this period was the optical
sound-on-film system, also generally credited to Lee De Forest. Although famous
early "Talkies" like The Jazz Singer used a sound-on-disc system, the film
industry eventually adopted the optical sound-on-film system and it
revolutionised the movie industry in the 1930s, ushering in the era of 'talking
pictures'. Optical sound-on-film, based on the photoelectric cell, became the
standard film audio system throughout the world until it was superseded in the
1960s.
Magnetic tape
magnetic tape sound recording
Other important inventions of this period were magnetic tape and the tape
recorder (Telegraphone). Paper-based tape was first used but was soon superseded
by polyester and acetate backing due to dust drop and hiss. Acetate was more
brittle than polyester and snapped easily. This technology, the basis for almost
all commercial recording from the 1950s to the 1980s, was invented by German
audio engineers in the 1930s, who also discovered the technique of AC biasing,
which dramatically improved the frequency response of tape recordings. Tape
recording was perfected just after the war by American audio engineer John T.
Mullin with the help of Crosby Enterprises (Bing Crosby), whose pioneering
recorders were based on captured German recorders, and the Ampex company
produced the first commercially available tape recorders in the late 1940s.
Magnetic tape brought about sweeping changes in both radio and the recording
industry. Sound could be recorded, erased and re-recorded on the same tape many
times, sounds could be duplicated from tape to tape with only minor loss of
quality, and recordings could now be very precisely edited by physically cutting
the tape and rejoining it.
Within a few years of the introduction of the first commercial tape recorder,
the Ampex 200 model, launched in 1948, American musician-inventor Les Paul had
invented the first multitrack tape recorder, bringing about another technical
revolution in the recording industry. Tape made possible the first sound
recordings totally created by electronic means, opening the way for the bold
sonic experiments of the Musique Concrète school and avant garde composers like
Karlheinz Stockhausen, which in turn led to the innovative pop music recordings
of artists such as Frank Zappa, The Beatles and The Beach Boys.
Tape enabled the radio industry for the first time to pre-record many sections
of program content such as advertising, which formerly had to be presented live,
and it also enabled the creation and duplication of complex, high-fidelity,
long-duration recordings of entire programs. It also, for the first time,
allowed broadcasters, regulators and other interested parties to undertake
comprehensive logging of radio broadcasts. Innovations like multitracking and
tape echo enabled radio programs and advertisements to be pre-produced to a
level of complexity and sophistication that was previously unattainable and tape
also led to significant changes to the pacing of program content, thanks to the
introduction of the endless-loop tape cartridge.
Stereo and Hi-fi
Magnetic tape also enabled the development of the first practical commercial
sound systems that could record and reproduce high-fidelity stereophonic sound.
Experiments with stereo dated back to the 1880s and during the 1930s and 1940s
there were many attempts to record in stereo using discs, but these were
hampered by problems with synchronization.
The first major breakthrough in practical stereo sound was made by Bell
Laboratories, who in 1937 demonstrated a practical system of two-channel stereo,
using dual optical sound tracks on film. Major movie studios quickly developed
three-track and four-track sound systems, and the first stereo sound recording
in a commercial film was made by Judy Garland for the MGM movie Listen, Darling
in 1938. The first commercially-released movie with a full surround soundtrack
was Walt Disney's Fantasia, released in 1940. The sound for this production was
originally recorded on a completely separate magnetic film, but because of the
complex equipment required to present it, it was shown as a road show, but only
in the United States. Regular releases of the film were on standard mono optical
35 mm stock until the film was transferred to multichannel 70mm stock in the
1970s.
German audio engineers working on magnetic tape are reported to have developed
stereo recording by 1943, but it was not until the introduction of the first
commercial two-track tape recorders by Ampex in the late 1940s that stereo tape
recording became commercially feasible. However, despite the availability of
multitrack tape, stereo did not become the standard system for commercial music
recording for some years and it remained a specialist market during the 1950s.
This changed after the late 1957 introduction of the "Westrex stereo phonograph
disc".
Decca Records in England came out with FFRR (Full Frequency Range Recording) in
the 1940s which became internationally accepted and a worldwide standard for
higher quality recordings on vinyl records. The Ernest Ansermet recording of
Igor Stravinsky's Petrushka was key in the development of full frequency range
records and alterting the listening public to high fidelity in 1946.[6]
Most pop singles were mixed into monophonic sound until the mid 1960s, it was
common for major pop releases to be issued in both mono and stereo until the
early 1970s. Many Sixties pop albums now available only in stereo were
originally intended to be released only in mono, and the so-called "stereo"
version of these albums were created by simply separating the two tracks of the
master tape. In the mid Sixties, as stereo became more popular, many mono
recordings (such as The Beach Boys' Pet Sounds) were remastered using the
so-called "fake stereo" method, which spread the sound across the stereo field
by directing higher-frequency sound into one channel and lower-frequency sounds
into the other.
1950s and beyond
Magnetic tape transformed the recording industry, and by the late-1950s the vast
majority of commercial recordings were being mastered on tape. The electronics
revolution that followed the invention of the transistor brought other radical
changes, the most important of which was the introduction of the world's first
"personal music device", the miniaturized transistor radio, which became a major
consumer luxury item in the 1960s, transforming radio broadcasting from a static
group experience into a mobile, personal listening activity.
The first multitrack recording made using magnetic tape was "How High the Moon"
by Les Paul, on which Paul played eight overdubbed guitar tracks. In the 1960s
Brian Wilson of The Beach Boys, Frank Zappa and The Beatles (with producer
George Martin) were among the first popular artists to explore the possibilities
of multitrack techniques and effects on their landmark albums Pet Sounds, Freak
Out! and Sgt. Pepper's Lonely Hearts Club Band.
The next important innovation was small cartridge based tape systems of which
the compact cassette, introduced by the Philips electronics company in 1964 is
the best known. It eventually entirely replaced the competing formats, the
larger 8-track tape (used primarily in cars) and the fairly similar 'Deutsche
Cassette' developed by the German company Grundig. This latter system was not
particularly common in Europe and practically unheard of in America. The compact
cassette became a major consumer audio format and advances in microelectronics
eventually allowed the development of the Sony Walkman, introduced in the 1970s,
which was the first personal music player and gave a major boost to the mass
distribution of music recordings. Cassettes became the first successful consumer
recording/re-recording medium. The gramophone record was a pre-recorded playback
only medium, and reel-to-reel tape was too difficult for most consumers and far
less portable.
A key advance in audio fidelity came with the Dolby A noise reduction system,
invented by Ray Dolby and introduced in 1966. A competing system dbx, invented
by David Blackmer, found most success in professional audio. A simpler variant
of Dolby's noise reduction system, known as Dolby B greatly improved the sound
of cassette tape recordings by reducing the practical effect of the recorded
hiss inherent in the narrow tape used. It, and variants, also eventually found
wide application in the recording and film industries. Dolby B was crucial to
the popularisation and commercial success of the compact cassette as a domestic
recording and playback medium, and became a part of the booming "hi-fi" market
of the 1970s and beyond. The compact cassette also benefited enormously from
developments in the tape material itself as materials with wider frequency
responses and lower inherent noise were developed, often based on cobalt and/or
chrome oxides as the magnetic material instead of the more usual iron oxide.
The multitrack audio cartridge had been in wide use in the radio industry, from
the late 1950s to the 1980s, but in the 1960s the pre-recorded 8-track cartridge
was launched as a consumer audio format by Bill Lear of the Lear Jet aircraft
company (and although its correct name was the 'Lear Jet Cartridge', it was
seldom referred to as such). Aimed particularly at the automotive market, they
were the first practical, affordable car hi-fi systems, and could produce
superior sound quality to the compact cassette. However the smaller size and
greater durability -- augmented by the ability to create home-recorded music
"compilations" since 8-track recorders were rare -- saw the cassette become the
dominant consumer format for portable audio devices in the 1970s and 1980s.
There had been experiments with multi-channel sound for many years -- usually
for special musical or cultural events -- but the first commercial application
of the concept came in the early 1970s with the introduction of Quadraphonic
sound. This spin-off development from multitrack recording used four tracks
(instead of the two used in stereo) and four speakers to create a 360-degree
audio field around the listener. Following the release of the first consumer
4-channel hi-fi systems, a number of popular albums were released in one of the
competing four-channel formats; among the best known are Mike Oldfield's Tubular
Bells and Pink Floyd's The Dark Side of the Moon. Quadraphonic sound was not a
commercial success, partly because of competing and somewhat incompatible
four-channel sound systems (eg, CBS, JVC, Dynaco and others all had systems) and
generally poor quality, even when played as intended on the correct equipment,
of the released music. It eventually faded out in the late 1970s, although this
early venture paved the way for the eventual introduction of domestic Surround
Sound systems in home theatre use, which have gained enormous popularity since
the introduction of the DVD. This widespread adoption has occurred despite the
confusion introduced by the multitude of available surround sound standards.
The replacement of the thermionic valve (vacuum tube) by the smaller, cooler and
less power-hungry transistor also accelerated the sale of consumer high-fidelity
"hi-fi" sound systems from the 1960s onward. In the 1950s most record players
were monophonic and had relatively low sound quality; few consumers could afford
high-quality stereophonic sound systems. In the 1960s, American manufacturers
introduced a new generation of "modular" hi-fi components -- separate
turntables, pre-amplifiers, amplifiers, both combined as integrated amplifiers,
tape recorders, and other ancillary equipment (like the graphic equaliser),
which could be connected together to create a complete home sound system. These
developments were rapidly taken up by Japanese electronics companies, which soon
flooded the world market with relatively cheap, high-quality components. By the
1980s, corporations like Sony had become world leaders in the music recording
and playback industry.
Digital recording
A digital sound recorder
A digital sound recorder
The invention of digital sound recording and the compact disc in 1982 brought
significant improvements in the durability of consumer recordings. The CD
initiated another massive wave of change in the consumer music industry, with
vinyl records effectively relegated to a small niche market by the mid-1990s.
However, the introduction of digital systems was initially fiercely resisted by
the record industry which feared wholesale piracy on a medium which was able to
produce perfect copies of original released recordings. However, various
protection system (principally SCMS) persuaded the industry to bow to the
inevitable.
The most recent and revolutionary developments have been in digital recording,
with the invention of purely electronic consumer recording formats such as the
WAV digital music file and the compressed file type, the MP3. This generated a
new type of portable solid-state computerised digital audio player, the MP3
player. Another invention, by Sony, was the minidisc player, using ATRAC
compression on small, cheap, re-writeable discs. This was in vogue in the 1990s,
and is still popular, especially in a newer, longer playing and higher fidelity
version. New technologies such as Super Audio CD, DVD-A, Blu-ray Disc and HD DVD
continue to set a very high rate of change in digital audio storage.
This technology spreads across various associated fields, from hi-fi to
professional audio, internet radio and podcasting.
Technological developments in recording and editing have transformed the record,
movie and television industries in recent decades. Audio editing became
practicable with the invention of magnetic tape recording, but the use of
computers has made editing operations faster and easier to execute with
software, and the use of hard-drives for storage has made recording cheaper.
Today, the process of making a recording is separated into tracking, mixing and
mastering. Multitrack recording makes it possible to capture signals from
several microphones, or from different 'takes' to tape or disc, with maximized
headroom and quality, allowing previously unavailable flexibility in the mixing
and mastering stages for editing, level balancing, compressing and limiting,
adding effects such as reverberation, equalisation, flanging, and much more.
In the 1920s, the early talkies featured the new sound-on-film technology which
used photoelectric cells to record and reproduce sound signals that were
optically recorded directly onto the movie film. The introduction of talking
movies, spearheaded by The Jazz Singer in 1927 (though it used a sound on disk
technique, not a photoelectric one), saw the rapid demise of live cinema
musicians and orchestras. They were replaced with pre-recorded soundtracks,
causing the loss of many jobs.[7] The American Federation of Musicians took out
ads in newspapers, protesting the replacement of real musicians with mechanical
playing devices, especially in theatres.[8]
Voice to note
Voice-to-note refers to the capability of personal computers to be able to
recognize notes that are sung, hummed, or whistled into a microphone. The pitch
and duration of the notes are then calculated and converted into MIDI music
files.[citation needed]
Legal status
UK
Since 1934, sound recordings are treated differently from musical works under
copyright law.[9] Copyright, Designs and Patents Act 1988 defines a sound
recording to mean (a) a recording of sounds, from which the sounds may be
reproduced, or (b) a recording of the whole or any part of a literary, dramatic
or musical work, from which sounds reproducing the work or part may be produced,
regardless of the medium on which the recording is made or the method by which
the sounds are reproduced or produced. It thus covers vinyl records, tapes,
compact discs, digital audiotapes, and MP3s which embody recordings.
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. 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.
A Neumann U87 condenser microphone
A Neumann U87 condenser microphone
The most common design today uses a thin membrane which vibrates in response to
sound pressure. This movement is subsequently translated into an 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.
Varieties
Condenser, capacitor or electrostatic microphones
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: DC-biased and RF (or HF) condenser microphones.
With a DC-biased microphone, the plates are biased with a fixed charge (Q). The
voltage maintained across the capacitor plates changes with the vibrations in
the air, according to the capacitance equation (Q = C \ V), where Q = charge in
coulombs, C = capacitance in farads and V = potential difference in volts. The
capacitance of the plates is inversely proportional to the distance between them
for a parallel-plate capacitor. (See capacitance for details.)
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 changes
instantaneously to reflect the change in capacitance. The voltage across the
capacitor varies above and below the bias voltage. The voltage difference
between the bias and the capacitor is seen across the series resistor. The
voltage across the resistor is amplified for performance or recording.
RF condenser microphones use a comparatively low RF voltage, generated by a
low-noise oscillator. The oscillator may either be frequency modulated by the
capacitance changes produced by the sound waves moving the capsule diaphragm, or
the capsule may be part of a resonant circuit that modulates the amplitude of
the fixed-frequency oscillator signal. Demodulation yields a low-noise audio
frequency signal with a very low source impedance. This technique permits the
use of a diaphragm with looser tension, which may be used to achieve better
low-frequency response. 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
biasing technique.
Patti Smith singing into a Shure SM58 microphone
Patti Smith singing into a Shure SM58 microphone
Condenser microphones span the range from inexpensive Karoake mics to
high-fidelity recording mics. 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 or CAD M179.
Electret condenser microphones
Electret microphone
First patent on foil electret microphone by G. M. Sessler et al. (pages 1 to 3)
First patent on foil electret microphone by G. M. Sessler et al. (pages 1 to 3)
An electret microphone is a relatively new type of capacitor microphone invented
at Bell laboratories in 1962 by Gerhard Sessler and Jim West[1]. The
externally-applied charge described above under condenser microphones is
replaced by a permanent charge in an electret material. An electret is a
ferroelectric material that has been permanently electrically charged or
polarized. The name comes from electrostatic and magnet; a static charge is
embedded in an electret by alignment of the static charges in the material, much
the way a magnet is made by aligning the magnetic domains in a piece of iron.
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. Moving coil microphones use the same dynamic
principle as in a loudspeaker, only reversed. A small movable induction coil,
positioned in the magnetic field of a permanent magnet, is attached to the
diaphragm. When sound enters through the windscreen of the microphone, the sound
wave moves the diaphragm. When the diaphragm vibrates, the coil moves in the
magnetic field, producing a varying current in the coil through electromagnetic
induction. A single dynamic membrane 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.
The AKG D 112, for example, is designed for bass response rather than treble[2].
In audio engineering several kinds of microphones are often used at the same
time to get the best result.
Ribbon microphones use a thin, usually corrugated metal ribbon suspended in a
magnetic field. The ribbon is electrically connected to the microphone's output,
and its vibration within the magnetic field generates the electrical signal.
Ribbon microphones are similar to moving coil microphones in the sense that both
produce sound by means of magnetic induction. Basic ribbon microphones detect
sound in a 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, were
once valued for this reason, but a good low-frequency response could only be
obtained if the ribbon is suspended very loosely, and this made them fragile.
Modern ribbon materials, including new nanomaterials[3] have now been introduced
that eliminate those concerns, and even improve the effective dynamic range of
ribbon microphones at low frequencies. Protective wind screens can reduce the
danger of damaging a vintage ribbon, and also reduce plosive artifacts in the
recording. Properly designed wind screens produce negligible treble attenuation.
In common with other classes of dynamic microphone, ribbon microphones don't
require phantom power; in fact, this voltage can damage some older ribbon
microphones. (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.)
Edmund Lowe using a ribbon microphone
Edmund Lowe using a ribbon microphone
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
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. 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
Laser microphones are portrayed in movies as spying devices. Consist on a laser
beam bouncing off the surface of a window or plane that is affected by sound in
a room, this movement changes the refraction angle of the beam, the receiver
will sense this change in light intensity and can be changed into sound once
again.
Liquid microphones
Water microphone
Early microphones did not produce intelligible speech, until Alexander Graham
Bell made improvements including a variable resistance microphone/transmitter.
Bell’s liquid transmitter consisted of a metal cup filled with water with a
small amount of sulfuric acid added. A sound wave caused the diaphragm to move,
forcing a needle to move up and down in the water. The electrical resistance
between the wire and the cup was then inversely proportional to the size of the
water meniscus around the submerged needle. Elisha Gray filed a caveat for a
version using a brass rod instead of the needle. Other minor variations and
improvements were made to the liquid microphone by Majoranna, Chambers, Vanni,
Sykes, and Elisha Gray, and one version was even patented by Reginald Fessenden
in 1903. These were the first working microphones, but they were not practical
for commercial application. The famous first phone conversation between Bell and
Watson took place using a liquid microphone.
MEMS microphones
The MEMS (MicroElectrical-Mechanical System) microphone is also called a
microphone chip or silicon microphone. The pressure-sensitive diaphragm is
etched directly into a silicon chip by MEMS techniques, and is usually
accompanied with integrated preamplifier. Most MEMS microphones are variants of
the condenser microphone design. Often MEMS mics have built in analog-to-digital
converter (ADC) circuits on the same CMOS chip making the chip a digital
microphone and so more readily integrated with modern digital products. Major
manufacturers producing MEMS silicon microphones are Analog Devices, Akustica
(AKU200x), Infineon (SMM310 product), Knowles Electronics, Memstech (MSMx)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. Less
commonly, microphones themselves can be used as speakers, almost always as
tweeters. This is less common since microphones are not designed to handle the
power that speaker components are routinely required to cope with. One instance
of such an application was the STC microphone-derived 4001 super-tweeter, which
was successfully used in a number of high quality loudspeaker systems from the
late 1960s to the mid-70s.
Capsule design and directivity
The shape of the microphone defines its directivity. Inner elements are of major
importance, such as the structural shape of the capsule. Outer elements may
include 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 sources arriving edge-on at the diaphragm produce no pressure
differential, giving pressure-gradient microphones their characteristic
figure-eight, or bi-directional patterns.
The capsule of a pressure transducer microphone is closed on one side, which
results in an omnidirectional pattern, responding to a change in pressure
regardless of the direction to the source.
Other polar patterns are derived by creating a capsule shape that combines these
two effects in different ways. The cardioid, for instance, features a partially
closed backside.[4]
Microphone polar patterns
(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 (or nondirectional) microphone's response is generally
considered to be a perfect sphere in three dimensions. In the real world, this
is not the case. As with directional microphones, the polar pattern for an
"omnidirectional" microphone is a function of frequency. The body of the
microphone is not infinitely small and, as a consequence, it tends to get in its
own way with respect to sounds arriving from the rear, causing a slight
flattening of the polar response. This flattening increases as the diameter of
the microphone (assuming it's cylindrical) reaches the wavelength of the
frequency in question. Therefore, the smallest diameter microphone 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.
An example of a nondirectional microphone is the round black eight ball.[5]
Unidirectional
A unidirectional microphone is sensitive to sounds from only one direction. The
diagram above illustrates a number of these patterns. The microphone faces
upwards in each diagram. The sound intensity for a particular frequency is
plotted for angles radially from 0 to 360°. (Professional diagrams show these
scales and include multiple plots at different frequencies. The diagrams given
here provide only an overview of typical pattern shapes, and their names.)
Cardioids
US664A University Sound Dynamic Supercardioid Microphone
US664A University Sound Dynamic 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
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,
putting them very close to the sound source (at distances of a few centimeters)
results in a bass boost. This is known as the proximity effect[6]
Application-specific 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.
A stereo microphone integrates two microphones in one unit to produce a
stereophonic signal. A stereo microphone is often used for broadcast
applications or field recording where it would be impractical to configure two
separate condenser microphones in a classic X-Y configuration (see microphone
practice) for stereophonic recording. Some such microphones have an adjustable
angle of coverage between the two channels.
A noise-canceling microphone is intended for noisy environments such as aircraft
cockpits. They are normally installed as boom mics on headsets. They pick up
environmental noise, ideally without also picking up the intended signal, with
one diaphragm and electrically combine the output with the intended signal
picked up with another diaphragm. In older designs, there is no active
electronics involved in the cancellation technique, unlike active noise
cancellation microphones. So, in the common configuration, the intended signal
is voice and one diaphragm is mounted close to the mouth. The other is, often,
placed behind the first, farther away from the intended signal source and
electrically out of phase with the first. After combination, signals other than
the voice are greatly reduced, substantially increasing intelligibility. Some
noise-canceling microphones are also throat microphones.
Connectors
Electronic symbol for a microphone.
Electronic symbol for a microphone.
The most common connectors used by microphones are:
* Male XLR connector on professional microphones
* ¼ inch mono phone plug on less expensive consumer microphones
* 3.5 mm (Commonly referred to as 1/8 inch mini) stereo (wired as mono) mini
phone plug on very inexpensive and computer microphones
Some microphones use other connectors, such as 1/4 inch TRS (tip ring sleeve),
5-pin XLR, or stereo mini phone plug (1/8 inch TRS) on some stereo microphones.
Some lavalier microphones use a proprietary connector for connection to a
wireless transmitter. Since 2005, professional-quality microphones with USB
connections have begun to appear, designed for direct recording into
computer-based software.
Impedance-matching
Microphones have an electrical characteristic called impedance, measured in ohms
(Ω), that depends on the design. Typically, the rated impedance is stated.[7]
Low impedance is considered under 600 Ω. Medium impedance is considered between
600 Ω and 10 kΩ. High impedance is above 10 kΩ.
Most professional microphones are low impedance, about 200 Ω or lower.
Low-impedance microphones are preferred over high impedance for two reasons: one
is that using a high-impedance microphone with a long cable will result in loss
of high frequency signal due to the capacitance of the cable; the other is that
long high-impedance cables tend to pick up more hum (and possibly
radio-frequency interference (RFI) as well). However, some devices, such as
vacuum tube guitar amplifiers, have 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, 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 to 60 dB lower (a factor of 100 to 1000) than a line
input. Microphone inputs include the necessary amplification to handle these
very low level signals. Certain ribbon and dynamic microphones, which are most
linear when operated into a load of known impedance, are exceptions[clarify].[8]
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
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,[7] 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. Very quiet microphones
have existed for years for special applications, such the Brüel & Kjaer 4179,
with a noise level around 0 dB SPL. Recently some microphones with low noise
specifications have been introduced in the studio/entertainment market, such as
models from Neumann and Røde that advertise noise levels between 5 and 7 dBA.
Typically this is achieved by altering the frequency response of the capsule and
electronics to result in lower noise within the A-weighting curve while
broadband noise may be increased.
The maximum SPL (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
This section does not cite any references or sources.
Please help improve this article by adding citations to reliable sources.
Unverifiable material may be challenged and removed. (March 2008)
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
Measurement microphones are used in sound analysis meters, noise measurement (in
public nuisance abatement contexts), acoustic laboratories, loudspeaker design
and quality control work, etc. They are made with greater care than most
microphones and generally come with a calibration certificate. However, like
most manufactured products there can be variations, which may change over the
lifetime of the device. Accordingly, it is regularly necessary to test the test
microphones. This service is offered by some microphone manufacturers and by
independent certified testing labs. Some test enough microphones to justify an
in-house calibration lab. Depending on the application, measurement microphones
must be tested periodically (every year or several months, typically) and after
any potentially damaging event, such as being dropped (most such mikes come in
foam-padded cases to reduce this risk) or exposed to sounds beyond the
acceptable level.
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
Microphone array
A microphone array is any number of microphones operating in tandem. There are
many applications:
* Systems for extracting voice input from ambient noise (notably telephones,
speech recognition systems, hearing aids)
* Surround sound and related technologies
* Locating objects by sound: acoustic source localization, e.g. military use to
locate the source(s) of artillery fire. Aircraft location and tracking.
* High fidelity original recordings
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.). Most
microphones have an integral windscreen built around the microphone diaphragm. A
screen of plastic, wire mesh or a metal cage is held at a distance from the
microphone diaphragm, to shield it. This cage provides a first line of defense
against the mechanical impact of objects or wind. Some microphones, such as the
Shure SM58, may have an additional layer of foam inside the cage to further
enhance the protective properties of the shield. Beyond integral microphone
windscreens, there are three broad classes of additional wind protection.
Microphone covers
Microphone covers are often made of soft open-cell polyester or polyurethane
foam because of the inexpensive, disposable nature of the foam. Optional
windscreens are often available from the manufacturer and third parties. A 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.[9]. One disadvantage of polyurethane foam microphone
covers is that they can deteriorate over time. Windscreens also tend to collect
dirt and moisture in their open cells and must be cleaned to prevent high
frequency loss, bad odor and unhealthy conditions for the person using the
microphone. On the other hand, a major advantage of concert vocalist windscreens
is that one can quickly change to a clean windscreen between users, reducing the
chance of transferring germs. Windscreens of various colors can be used to
distinguish one microphone from another on a busy, active stage.
Pop filters
Pop filters or pop screens are used in controlled studio environments to keep
plosives down when recording. A typical pop filter is composed of one or more
layers of acoustically semi-transparent material such as woven nylon stretched
over a circular frame and a clamp and a flexible mounting bracket to attach to
the microphone stand. The pop shield is placed between the vocalist and the
microphone. The need for 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.
Blimps
Blimps (also known as zeppelins) are large hollow windscreens used to surround
microphones for outdoor location audio, such as nature recording, electronic
news gathering, and for film and video shoots. They can cut wind noise by as
much as 25 dB, especially low-frequency noise. The blimp is essentially a hollow
cage or basket with acoustically transparent material stretched over the outer
frame. The blimp works by creating a volume of still air around the microphone.
The microphone is often further isolated from the blimp by an elastic suspension
inside the basket. This reduces wind vibrations and handling noise transmitted
from the cage. To extend the range of wind speed conditions in which the blimp
will remain effective, many have the option of fitting a secondary cover over
the outer shell. This is usually a furry material with long soft hairs and a
weave that is as acoustically transparent as possible. The hair acts as a filter
to any wind turbulence hitting the blimp. A synthetic furry cover can reduce
wind noise by a further 12 dB.[10]. One disadvantage of all windscreen types is
that the microphone's high frequency response is attenuated by a small amount
depending on the density of the protective layer.
Acronym Definition
NHTH: NHT Home. Naphtha Hydrotreater Home.
NHTH: NHT Home. National Heritage Trust Home.
NHTH: NHT Home. National Housing Trust Home.
NHTH: NHT Home. Nava-Hopi Tours Home.
NHTH: NHT Home. New Horizons Technology Home.
NHTH: NHT Home. North Harris Trust Home.
NHTH: NHT Home. Now Hear This Home.
NHTH: NHT Home. Now Here This Home.
Acronym Definition
NHTH: Naturally HTH. Naturally Hope That Helps
NHTH: Naturally HTH. Naturally Hope This Helps
NHTH: Not HTH. Not Haliotis Tuberculata Hemocyanin
NHTH: Not HTH. Not Hand To Hand combat
NHTH: Not HTH. Not Hand To Heart (I'm being honest)
NHTH: Not HTH. Not Hang Them High
NHTH: Not HTH. Not Happy To Help
NHTH: Not HTH. Not Hard to Handle
NHTH: Not HTH. Not Harry the Horse
NHTH: Not HTH. Not Have to Hurry (internet chat)
NHTH: National HTH. National Hawthorne, Nevada USA (Airport Code)
NHTH: Not HTH. Not Head to Head
NHTH: Not HTH. Not Heart To Heart
NHTH: Not HTH. Not Helix-Turn-Helix (Protein Structure)
NHTH: Not HTH. Not Hell This Hurts
NHTH: New HTH. New High Test Hypochlorite
NHTH: Not HTH. Not Highway to Hell
NHTH: New HTH. New Highway Traffic Headquarters
NHTH: Not HTH. Not Hill Top Hose
NHTH: Not HTH. Not Hilltop Hoods
NHTH: Now HTH. Now Hit the Highway
NHTH: Not HTH. Not Hjälper Torr Hud
NHTH: New HTH. New Hollywood Tower Hotel
NHTH: New HTH. New Home Town Hero
NHTH: Not HTH. Not Hoping This Helps
NHTH: Not HTH. Not Hotter Than Hell
NHTH: Now HTH. Now How the Hell?
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