audio visual entertainment system chapter 2
TRANSCRIPT
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CHAPTER IIREVIEW OF RELATED LITERATURE
This chapter presented the conceptual and research literature which helped in
providing direction in the completion of this study.
Conceptual Literature
Equalizers
Equalizers are an essential part of any sound system. They have many
applications for various users. An equalizer is a filter that allows a person to control the
tone (frequency response), of a sound system.
Equalizers are built to control the loss and gain of frequencies within a sound
system. This allows a sound system to sound natural and full. It also gives it the ability
to maximize volume while eliminating feedback. Many stereos today are built with a
graphic equalizer right in the system. But for a high-end stereo system this unit is
generally separate and allows fine tuning. Since there are not usually any microphone
inputs in a home stereo application, the adjustments usually do not have to take into
account the ambient sound in the room, but do help compensate for the "acoustics" in
the room.
Originally, analogue equalizers worked by passing an AC (alternating current)
signal through capacitors and inductors. The phase of the signal was shifted as it
passed through. This shifted signal was then recombined with the original signal for a
cancelling or partially cancelling effect (frequencies can also be enhanced). This could
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be done for specific frequencies so that different frequencies could be adjusted to
certain levels simultaneously.
Today's modern digital equalizers mimic the behavior of analogue equalizers.
They do this using taps on a digital delay line. This is really a series of memory
locations that the signal (or at least a number representing the signal) is passed
through. It goes first to location 0, then to location 1 and then 2 and so on until it
reaches the output phase. This setup is called a shift register and the effect is the same
as if the signal was passed through a capacitor and an inductor.
You can vary the signal by changing how many cells are in your shift register or
by choosing different registers as the output. The signal is recombined with the original
signal with the expected result. (Indepthinfo, 2009)
Graphic Equalizers
The great advantage of a graphic equalizer over other equalizers is that it is easy
to visualize and adjust the controls. Most people have seen the typical sliding controls.
The drawbacks of the graphic equalizer ate that it has fixed frequencies and Q, which
limits the user's ability to be precise.
Graphic equalizers often come with home stereo systems, but can be seen in
places of worship, banquet halls and in the equipment of live music providers. Graphic
equalizers are common in high-end sound systems. When you go to a concert, the
"sound man" sitting in the back of the hall pushing slides up and down is probably using
a graphic equalizer.
Graphic equalizers have an unchanging Q, which is usually designated as a 1, 2
or 3. These numbers basically describe how much of an octave each frequency control
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covers. The designation 1 covers one octave, a two means that each control covers 1/2
octave, and a 3 covers 1/3 of an octave. This means the higher the Q, the smaller the
range covered, but the more precise each control can be.
The little slide buttons are called potentiometers. They are placed side-by-side on
the graphic display. Normally the slides will form a smooth wave pattern. This is
because the noise being cancelled or enhanced generally spans more than one
frequency in different strengths.
Computers can function as a graphic equalizer when processing music and
sound filesfor its speakers. The interface on the computer normally looks very much like
a graphic equalizer's controls on a separate unit.
Parametric Equalizers
The great advantage of a parametric equalizer is that it allows the user to change
the frequency and Q. However, this is offset by the fact that this very ability complicates
the inexperienced user's efficacious use the system. Adjustments can be so complex
that the needed change might be difficult to determine.
A parametric equalizer uses knobs for its control functions, which makes it more
difficult to visualize the set-up of the equalizer. Even so, it admirably performs the main
functions of an equalizer which is to control the loss and gain in a frequency within a
sound system.
Parametric equalizers usually have 3 to 6 bands. Some have overlapping
frequency ranges. Others have broadband control which allows it to be used over the
complete frequency range. Most parametric equalizers have a switchable range switch
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that allows operation in a x1 or x10 mode, allowing the frequency to be equalized on an
even finer scale.
Found in almost every venue, the parametric equalizer certainly has its
advocates, but it has by no means replaced the graphic equalizer as the preferred
device for sound technicians
Equalizer Operation
An equalizer is used to change the frequency response of an audio system. This
enables the sound technician or the user to make adjustments that enhance the quality
of the audio produced. However, it is well to remember that a sound system is like a
chain, it is only as good as its weakest link. Poor microphones or speakers, et cetera
can only be partially compensated for by a good equalizer.
A speaker that is not producing an even output, where some frequencies are
louder than others can be smoothed out by adjusting the slide controls on a graphic
equalizer. This takes a sharp ear and a degree of experimental adjustment in the
inexperienced user. The sound coming from the speaker can seem to be too harsh and
lacking in base.
Where the speaker is mounted in a room can make a significant difference as to
the sound quality. A speaker mounted high up and in the corner of a room will actually
increase the bass, sometimes so much that it must be adjusted for with the graphic
equalizer.
The shape of a room, the construction of the walls, ceiling and floors and even
the presence of people (who actually absorb sound and prevent echoes) can be a
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factor. Although some echoes cannot be removed, some clarity of sound can be gained
by reducing some of the base frequencies.
Feedback can be reduced with 1/3 band equalizers. This can be done by turning
up the volume on a microphone until a squeal is heard then by turning down a specific
frequency associated with the squeal. Thus the feedback can be eliminated. This
means you can then increase volume a little more and adjust again for feedback. There
is, of course, a limit on how much volume can be gained in this manner. Some new
equalizers come with an automatic feedback controller. It can sense the feedback and
puts a narrow filter on the frequency in question.
Generally, in an in-place sound system, the original settings of the equalizer are
determined by a sound technician using a special (and expensive) instrument called an
RTA (real time spectrum analyzer). Be sure to note where all the settings are on the
equalizer after the job has been done, (in case some unauthorized person makes some
radical adjustments). (Indepthinfo, 2009)
Speakers
A loudspeaker (or "speaker") is an electroacoustic transducer that converts
an electrical signal into sound. The speaker moves in accordance with the variations of
an electrical signal and causes sound waves to propagate through a medium such as
air or water.
After the acoustics of the listening space, loudspeakers (and other
electroacoustic transducers) are the most variable elements in a modern audio system
and are usually responsible for most distortion and audible differences when comparing
sound systems.
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Driver Design
The most common type of driver uses a lightweight diaphragm, orcone,
connected to a rigid basket, or frame, via a 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 voice
coil, making it a variable electromagnet. The coil and the driver's magnetic system
interact, generating a mechanical force that causes the coil (and thus, the attached
cone) to move back and forth, thereby reproducing 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 energy storage issues; 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; thus, driver design
involves trade-offs. For example, paper is light and typically well damped, but is not stiff;
metal may be stiff and light, but it usually has poor damping; plastic can be light, but
typically, the stiffer it is made, the poorer the damping. As a result, many cones are
made of some sort of composite material. For example, a cone might be made of
cellulose paper, into which some carbon fiber, Kevlar, fiberglass,
or hemp or bamboo fibers have been added; or it might use a honeycomb sandwich
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construction; or a coating might be applied to it so as to provide additional stiffening or
damping.
The chassis, frame, or basket, is designed to be rigid, avoiding deformation
which would change critical alignments with the magnet gap, perhaps causing the voice
coil to rub against the sides of the gap. Chassis are typically cast from aluminum alloy,
or stamped from thin steel sheet, although molded plastic baskets are becoming
common, especially for inexpensive, low-mass drivers. Metallic chassis can play an
important role in conducting heat away from the voice coil; heating during operation
changes resistance, causing physical dimensional changes, and if extreme, may even
demagnetize permanent magnets.
The suspension system keeps the coil centered in the gap and provides a
restoring (centering) force that returns the cone 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, impregnated with a stiffening resin. The name comes from the shape of early
suspensions, which were two concentric rings of Bakelite material, joined by six or eight
curved "legs". Variations of this topology included the addition of a felt disc to provide a
barrier to particles that might otherwise cause the voice coil to rub. The German firm
Rulik still offers drivers with uncommon spiders made of wood.
The cone surround can be rubber or polyester foam, or a ring of corrugated, resin
coated fabric; it is attached to both the outer diaphragm circumference and to the frame.
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These different surround materials, their shape and treatment can dramatically affect
the acoustic output of a driver; each class and implementation having advantages and
disadvantages. Polyester foam, for example, is lightweight and economical, but is
degraded by exposure to ozone, UV light, humidity and elevated temperatures, limiting
its useful life to about 15 years.
The wire in a voice coil is usually made of copper, though aluminumand,
rarely, silvermay 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 co-axially inside the gap; it moves back and forth within a
small circular volume (a hole, slot, or groove) in the magnetic structure. 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 (called the pole
piece) being the other. The pole piece and backplate are often a single piece, called the
poleplate or yoke.
Modern driver magnets are almost always permanent and made
of ceramic, ferrite, Alnico, or, more recently, rare earth magnets. A trend in designdue
to increases in transportation costs and a desire for smaller, lighter devices (as in many
home theater multi-speaker installations)is the use of the last instead of heavier ferrite
types. Very few manufacturers still use electrically powered field coils, as was common
in the earliest designs (one such is French). When high field-strength permanent
magnets became available, Alnico, an alloy of aluminum, nickel, and cobalt became
popular, since it dispensed with the power supply issues of field-coil drivers. Alnico was
used for almost exclusively until about 1980. Alnico magnets can be partially degaussed
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(i.e., demagnetized) by accidental 'pops' or 'clicks' caused by loose connections,
especially if used with a high power amplifier. This damage can be reversed by
"recharging" the magnet.
Full-range drivers
A full-range driver is designed to have the widest frequency response possible.
These drivers are small, typically 3 to 8 inches (7.6 to 20 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 (or
more accurately, wide-range) drivers are most commonly heard in public address
systems, in televisions (although some models are suitable for hi-fi listening), small
radios, intercoms, some computer speakers, etc. In hi-fi speaker systems, the use of
wide-range drive units can avoid undesirable interactions between multiple drivers
caused by non-coincident driver location or crossover network issues. Fans of wide-
range driver hi-fi speaker systems claim a coherence of sound, said to be due to the
single source and a resulting lack of interference, and likely also to the lack of crossover
components. Detractors typically cite wide-range drivers' limited frequency response
and modest output abilities (most especially at low frequencies), together with their
requirement for large, elaborate, expensive enclosuressuch as transmission lines, or
hornsto approach optimum performance.
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
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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.
Limited-range drivers, also used alone, are typically found in computers, toys,
and clock radios. These drivers are less elaborate and less expensive than wide-range
drivers, and they may be severely compromised to fit into very small mounting locations.
In these applications, sound quality is a low priority. The human ear is remarkably
tolerant of poor sound quality, and the distortion inherent in limited-range drivers may
enhance their output at high frequencies, increasing clarity when listening to spoken
word material.
Subwoofer
A subwoofer is a woofer driver used only for the lowest part of the audio
spectrum: typically below 200 Hz for consumer systems, below 100 Hz for professional
live sound, and below 80 Hz in THX-approved systems.Because the intended range of
frequencies is limited, subwoofer system design is usually simpler in many respects
than for conventional loudspeakers, often consisting of a single driver enclosed in a
suitable box or enclosure.
To accurately reproduce very low bass notes without unwanted resonances
(typically from cabinet panels), subwoofer systems must be solidly constructed and
properly braced; good speakers are typically quite heavy. Many subwoofer systems
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include power amplifiers and electronic sub-filters, with additional controls relevant to
low-frequency reproduction. These variants are known as "active subwoofers". In
contrast, "passive" subwoofers require external amplification.
Woofer
A woofer is a driver that reproduces low frequencies. The driver combines with
the enclosure design to produce suitable low frequencies. Some loudspeaker systems
use a woofer for the lowest frequencies, sometimes well enough that a subwoofer is not
needed. 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 can work low enough that, combined with a woofer that responds high
enough, the two drivers add coherently in the middle frequencies.
Crossover
Used in multi-driver speaker systems, the crossover is a subsystem that
separates the input signal into different frequency ranges suited to each driver. The
drivers receive only the power in their usable frequency range (the range they were
designed for), thereby reducing distortion in the drivers and interference between them.
Crossovers can bepassive oractive. A passive crossover is an electronic circuit
that uses a combination of one or more resistors, inductors, or non-polar capacitors.
These parts are formed into carefully designed networks and are most often placed
between the power 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, but do
cause overall signal loss and a significant reduction in damping factor between the
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voice coil and the crossover. An active crossover is an electronic filter circuit that divides
the signal into individual frequency bands before power amplification, thus requiring at
least one power amplifier for each bandpass. Passive filtering may also be used in this
way before power amplification, but it is an uncommon solution, due to inflexibility
compared to active filtering. Any technique that uses crossover filtering followed by
amplification is commonly known as bi-amping, tri-amping, quad-amping, and so on,
depending on the minimum number of amplifier channels. Some loudspeaker designs
use a combination of passive and active crossover filtering, such as a passive crossover
between the mid- and high-frequency drivers and an active crossover between the low-
frequency driver and the combined mid- and high frequencies.
Passive crossovers are commonly installed inside speaker boxes and are by far
the most usual type of crossover for home and low-power use. In car audio systems,
passive crossovers may be in a separate box, necessary to accommodate the size of
the components used. Passive crossovers may be simple for low-order filtering, or
complex to allow steep slopes such as 18 or 24 dB per octave. Passive crossovers can
also be designed to compensate for undesired characteristics of driver, horn, or
enclosure resonances, and can be tricky to implement, due to component interaction.
Passive crossovers, like the driver units that they feed, have power handling limits, have
insertion losses (10% is often claimed), and change the load seen by the amplifier. The
changes are matters of concern for many in the hi-fi world. When high output levels are
required, active crossovers may be preferable. Active crossovers may be simple circuits
that emulate the response of a passive network, or may be more complex, allowing
extensive audio adjustments. Some active crossovers, usually digital loudspeaker
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management systems, may include facilities for precise alignment of phase and time
between frequency bands, equalization, and dynamics (compression and limiting)
control.
Some hi-fi and professional loudspeaker systems now include an active
crossover circuit as part of an onboard amplifier system. These speaker designs are
identifiable by their need for AC power in addition to a signal cable from a pre-amplifier.
This active topology may include driver protection circuits and other features of a digital
loudspeaker management system. Powered speaker systems are common in computer
sound (for a single listener) and, at the other end of the size spectrum, in modern
concert sound systems, where their presence is significant and steadily increasing.
Enclosures
Most loudspeaker systems consist of drivers mounted in an enclosure, or
cabinet. The role of the enclosure is to provide a place to physically mount the drivers,
and to prevent sound waves emanating from the back of a driver from interfering
destructively with those from the front; these typically cause cancellations (e.g., comb
filtering) and significantly alter the level and quality of sound at low frequencies.
The simplest driver mount is a flat panel (i.e., baffle) with the drivers mounted in
holes in it. However, in this approach, sound 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.
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Since panels of infinite dimensions are impractical, most enclosures function by
containing the rear radiation from the moving diaphragm. 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, glossy wall material,
internal bracing, curved cabinet wallsor more rarely, visco-elastic materials (e.g.,
mineral-loaded bitumen) or thin lead sheeting applied to the interior enclosure walls.
However, a rigid enclosure reflects sound internally, which can then be
transmitted back through the loudspeaker diaphragmagain 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 sound radiation so it can add constructively
to the output from the front of the cone. Designs that do this (including bass
reflex,passive radiator, 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 driver mounting locations 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,
providing a separate enclosure mounting for each driver, or (less commonly) using
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electronic techniques to achieve the same effect. These attempts have resulted in some
unusual cabinet designs.
The speaker mounting scheme (including cabinets) can also cause diffraction,
resulting in peaks and dips in the frequency response. The problem is usually greatest
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, using absorptive material around a driver, or some combination of
these and other schemes. (Wikipedia, 2008)
Television
Television (TV) is a widely used telecommunication medium for transmitting and
receiving moving images, either monochromatic ("black and white") or color, usually
accompanied by sound. "Television" may also refer specifically to a television
set, television programming or television transmission. The word is derived from
mixed Latin and Greek roots, meaning "far sight": Greek tele (), far, and Latin visio,
sight (from video,vis- to see, or to view in the first person). (Webster, 2007)
Commercially available since the late 1930s, the television set has become
common in homes, businesses and institutions, particularly as a source
of entertainment and news. Since the 1970s the availability of video
cassettes, laserdiscs, DVDs and now Blu-ray Discs, have resulted in the television set
frequently being used for viewing recorded as well as broadcast material.
Although other forms such as closed-circuit television (CCTV) are in use, the
most common usage of the medium is for broadcast television, which was modeled on
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the existing radio broadcasting systems developed in the 1920s, and uses high-
powered radio-frequency transmitters to broadcast the television signal to individual TV
receivers.
Broadcast TV is typically disseminated via radio transmissions on designated
channels in the 54890 megahertz frequency band. Signals are now often transmitted
with stereo and/or surround sound in many countries. Until the 2000s broadcast TV
programs were generally recorded and transmitted as an analog signal, but in recent
years public and commercial broadcasters have been progressively introducing digital
television broadcasting technology.
A standard television set comprises multiple internal electronic circuits, including
those for receiving and decoding broadcast signals. A visual display device which lacks
a tuner is properly called a monitor, rather than a television. A television system may
use different technical standards such as digital television (DTV) and high-definition
television (HDTV). Television systems are also used for surveillance, industrial process
control, and guiding of weapons, in places where direct observation is difficult or
dangerous. (Wikipedia, 2009)
Audio Amplifier
An audio amplifier is an electronic amplifier that amplifies low-power
audio signals (signals composed primarily of frequencies between20 - 20 000 Hz, the
human range of hearing) to a level suitable for driving loudspeakers and is the final
stage in a typical audio playback chain.
The preceding stages in such a chain are low power audio amplifiers which
perform tasks likepre-amplification, equalization, tone control,mixing/effects, or audio
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sources like record players, CD players, and cassette players. Most audio amplifiers
require these low-level inputs to adhere to line levels.
While the input signal to an audio amplifier may measure only a few
hundred microwatts, its output may be tens, hundreds, or thousands of watts.
Design parameters
Key design parameters for audio amplifiers are frequency response, gain, noise,
and distortion. These are interdependent; increasing gain often leads to undesirable
increases in noise and distortion. While negative feedback actually reduces the gain, it
also reduces distortion. Most audio amplifiers are linear amplifiers operating in class AB.
Filters and Preamplifiers
Historically, the majority of commercial audio preamplifiers made had complex
filter circuits for equalization and tone adjustment, due to the far from ideal quality of
recordings, playback technology, and speakers of the day.
Using today's high quality (often digital) source material, speakers, etc., such
filter circuits are usually not needed. Audiophiles generally agree that filter circuits are to
be avoided wherever possible. Today's audiophile amplifiers do not have tone controls
or filters.
Since modern digital devices, including CD and DVD players, radio receivers and
tape decks already provide a "flat" signal at line level, the preamp. is not needed other
than as volume control. One alternative to a separate preamp. is to simply use passive
volume and switching controls, sometimes integrated into a power amp. to form an
"integrated" amplifier.
Applications
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Important applications include public address systems, theatrical and concert
sound reinforcement, and domestic sound systems. The sound card in a personal
computer contains several audio amplifiers (depending on number of channels), as
does every stereo or home-theatre system. (Wikipedia, 2007)
Microphone
A microphone is an example of a transducer, a device that changes information
from one form to another. Sound information exists as patterns of air pressure; the
microphone changes this information into patterns of electric current. The recording
engineer is interested in the accuracy of this transformation, a concept he thinks of as
fidelity.
A variety of mechanical techniques can be used in building microphones. The
two most commonly encountered in recording studios are the magneto-dynamic and the
variable condenser designs. (artsites.ucsc.edu)
Dynamic Microphone
In the magneto-dynamic, commonly called dynamic, microphone, sound waves
cause movement of a thin metallic diaphragm and an attached coil of wire. A magnet
produces a magnetic field which surrounds the coil, and motion of the coil within this
field causes current to flow. The principles are the same as those that produce
electricity at the utility company, realized in a pocket-sized scale. It is important to
remember that current is produced by the motion of the diaphragm, and that the amount
of current is determined by the speed of that motion. This kind of microphone is known
as velocity sensitive. (artsites.ucsc.edu)
RibbonMicrophone
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The ribbon microphone, a high quality device widely used in commercial broadcasting
and recording, has thin corrugated metallic ribbon suspended between the poles of a
permanent magnet. When vibrated by sound, the ribbon cuts the magnetic lines of force
between the poles in alternating directions, generating an alternating voltage across the
ribbon. This voltage is then translated to travel along the line of the microphone and
converted into sound waves. (Encyclopedia Americana, 2000)
CondenserMicrophone
In a condenser microphone, the diaphragm is mounted close to, but not touching,
a rigid backplate. (The plate may or may not have holes in it.) A battery is connected to
both pieces of metal, which produces an electrical potential, or charge, between them.
The amount of charge is determined by the voltage of the battery, the area of the
diaphragm and backplate, and the distance between the two. This distance changes as
the diaphragm moves in response to sound and when the distance changes, current
flows in the wire as the battery maintains the correct charge. The amount of current is
essentially proportional to the displacement of the diaphragm, and is so small that it
must be electrically amplified before it leaves the microphone.
A common varient of this design uses a material with a permanently imprinted
charge for the diaphragm. Such a material is called an electret and is usually a kind of
plastic. (You often get a piece of plastic with a permanent charge on it when you unwrap
a record. Most plastics conduct electricity when they are hot but are insulators when
they cool.) Plastic is a pretty good material for making diaphragms since it can be
dependably produced to fairly exact specifications. (Some popular dynamic
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microphones use plastic diaphragms.) The major disadvantage of electrets is that they
lose their charge after a few years and cease to work. (artsites.ucsc.edu)
Crystal Microphone
The crystal microphone, which is widely used for public address systems and
home recording work, depends for its action on the piezoelectric effect of certain
crystals, most commonly Rochelle crystal. When pressure, in this case, sound waves, is
applied to the crystal in the proper direction, a proportionately varying voltage is
produced between opposite faces of the crystal. The advantages of the crystal
microphone are its relatively high output voltage, good sound quality, and low cost.
Crystal microphones are quite sensitive to extremes of heat and humidity, and they do
not tolerate rough handling or mechanical shock. (Encyclopedia Americana, 2000)
Specifications
There is no inherent advantage in fidelity of one type of microphone over
another. Condenser types require batteries or power from the mixing console to
operate, which is occasionally a hassle, and dynamics require shielding from stray
magnetic fields, which makes them a bit heavy sometimes, but very fine microphones
are available of both styles. The most important factor in choosing a microphone is how
it sounds in the required application. The following issues must be considered:
Sensitivity
This is a measure of how much electrical output is produced by a given sound.
This is a vital specification when recording very tiny sounds, such as a turtle snapping
its jaw, but should be considered in any situation. Placing an insensitive mic on a quiet
instrument, such as an acoustic guitar, the gain of the mixing console will have to be
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increased, adding noise to the mix. On the other hand, a very sensitive mic on vocals
might overload the input electronics of the mixer or tape deck, producing distortion.
Overload characteristics
Any microphone will produce distortion when it is overdriven by loud sounds. This
is caused by various factors. With a dynamic, the coil may be pulled out of the magnetic
field; in a condenser, the internal amplifier might clip. Sustained overdriving or extremely
loud sounds can permanently distort the diaphragm, degrading performance at ordinary
sound levels. Loud sounds are encountered more often, especially if the mic is placed
very close to instruments. There is usually a choice between high sensitivity and high
overload points, although occasionally there is also a switch on the microphone for
different situations.
Linearity or Distortion
This is the feature that runs up the price of microphones. The distortion
characteristics of a mic are determined mostly by the care with which the diaphragm is
made and mounted. High volume production methods can turn out an adequate
microphone, but the distortion performance will be a matter of luck. Many manufacturers
have several model numbers for what is essentially the same device. They build a
batch, and then test the mics and charge a premium price for the good ones.
No mic is perfectly linear; the best that can be done is to find one with distortion
that complements the sound.
Frequency response
A flat frequency response has been the main goal of microphone companies for
the last three or four decades. In the fifties, mics were so bad that console
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manufacturers began adding equalizers to each input to compensate. This effort has
now paid off to the point where most professional microphones are respectably flat, at
least for sounds originating in front. The major exceptions are mics with deliberate
emphasis at certain frequencies that are useful for some applications. Problems in
frequency response are mostly encountered with sounds originating behind the mic.
Noise
Microphones produce a very small amount of current when the live moving parts
follow sound waves. To be useful for recording or other electronic processes, the signal
must be amplified by a factor of over a thousand. Any electrical noise produced by the
microphone will also be amplified, so even slight amounts are intolerable. Dynamic
microphones are essentially noise free, but the electronic circuit built into condenser
types is a potential source of trouble, and must be carefully designed and constructed of
premium parts.
Noise also includes unwanted pickup of mechanical vibration through the body of
the microphone. Very sensitive designs require elastic shock mountings, and mics
intended to be held in the hand need to have such mountings built inside the shell.
The most common source of noise associated with microphones is the wire
connecting the mic to the console or tape deck. A mic preamp is very similar to a radio
reciever, so the cable must be prevented from becoming an antenna. The basic
technique is to surround the wires that carry the current to and from the mic with a
flexible metallic shield, which deflects most radio energy. A second technique, which is
more effective for the low frequency hum induced by the power company into our
environment, is to balance the microphone line. (Wikipedia, 2009)
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DVD Player
A DVD player is a device that plays discs produced under both the DVD-Video
and DVD-Audiotechnical standards, two different and incompatible standards. It was
invented in 1994.
Output
Due to multiple audio (and video) output devices, a consumer has many outputs
on a DVD player, and may become confused with connecting a player to a TV or
amplifier. Most systems include an optional digital audio connector for this task, which is
then paired with a similar input on the amplifier. The physical connection is typically
RCA connectors or TOSLINK, which transmits a S/PDIF stream carrying either
uncompressed digital audio (PCM) or the original compressed audio data (Dolby Digital,
DTS, MPEG audio) to be decoded by the audio equipment.
Video
Video is another issue which continues to present most problems. Current
players typically output analog video only, both composite video on an RCA jack as well
as S-Video in the standard connector. However, neither of these connectors was
intended to be used for progressive video, so yet another set of connectors has started
to appear, to carry a form of component video, which keeps the three components of
the video, one luminance signal and two color difference signal, as stored on the DVD
itself, on fully separate wires (whereas S-Video uses two wires, uniting and degrading
the two color signals, and composite uses only one, uniting and degrading all three
signals). The connectors are further confused by using a number of different physical
connectors on different player models, RCA or BNC, as well as using VGA cables in a
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non-standard way (VGA is normally analog RGBa different, incompatible form of
component video). Even worse, there are often two sets of component outputs, one
carrying interlaced video, and the other progressive, or an interlaced/progressive switch
(either a physical switch or a menu setting).
In Europe (but not most other PAL areas), SCART connectors are typically used,
which can carry composite and analog RGB interlaced video signals (RGB can be
progressive, but not all DVD players and displays support this mode) or Y/C (S-Video),
as well as analog two-channel sound and automatic 4:3 or 16:9 (widescreen) switching
on a single convenient multi-wire cable. The analog RGB component signal offers video
quality which is superior to S-Video and identical to YPbPr component video. However,
analog RGB and S-Video signals can not be carried simultaneously, due to each using
the same pins for different uses, and displays often must be manually configured as to
the input signal, since no switching mode exists for S-Video. (A switching mode does
exist to indicate whether composite or RGB is being used.) Some DVD players and set-
top boxes offer YPbPr component video signals over the wires in the SCART connector
intended for RGB, though this violates the official specification and manual configuration
is again necessary. (Hypothetically, unlike RGB component, YPbPr component signals
and S-Video Y/C signals could both be sent over the wire simultaneously, since they
share the luminance (Y) component.)
HDMI is a new digital connection for carrying high-definition video, similar to DVI.
Along with video, HDMI also supports up to eight-channel digital audio. DVD players
with connectors for high-definition video can upconvert the source to formats used for
higher definition video (e.g.,720p, 1080i, 1080p, etc.), before outputting the signal. By
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no means, however, will the resulting signal be high-definition video; that is, aside from
optional deinterlacing, upconverting generally consists of merely scaling the video's
dimensions to match that of higher resolution formats, foregoing the scaling that would
normally occur in the output device. (Wikipedia, 2008)
Related Literature
Past studies have also been conducted on the effectivity of audio and visual
systems as a medium of communication whether in the academe or outside its walls.
A recent study shows that pupils can spend as much as 40% to 50% of their
school day involved in listening to their teacher. However, as other studies have
indicated (Journal of the American Medical Association) 14. 9% of children aged from 6
to19 years may suffer some form of hearing loss. Similar studies have also showed that
around 80% of pupils may have some form of occasional hearing loss during their time
at school.
In a recommendation published by the UK Department of Education & Skills
(DfES) "Building Bulletin '93" it states that 'All children benefit from improved speech
clarity, not only those with permanent or temporary hearing loss. Academic performance
has been shown to improve for all class members with improvements noted in task
behaviour, attentiveness, understanding of instructions, less repetition required, better
attendance and improved levels of verbal recognition. Furthermore, due to the clarity of
speech from the teacher, similar improvements in learning performance are also noted
in students for whom English is a second language. '
More recently Government legislation in the USA, UK and Europe requires that
new schools in particular need to comply with a minimum standards of acoustic
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performance in classrooms. Effective levels of speech recognition for pupils is
considered paramount and the use of sound reinforcement systems is also viewed as
an ideal solution.
A typical sound reinforcement system provides the teacher with a wireless or
infrared microphone (more usually a lapel or pendant type) which links to an amplifier
and loudspeaker system.
Sound reinforcement systems raise the level of the teacher's voice but are
intended to be non-intrusive. These systems have been in use quite extensively in the
USA over the past 15 years or so and are seen as a significant opportunity to improve
academic performance in the UK and in Europe. (David Edis-Bates, 2008)
On the other hand, some studies have looked upon software improvements on
the audio visual system. This software allows the user to experience works of art such
as paintings and concerts on a different perspective.
In this study some software artists have approached the challenge of visualizing
music, not to produce entertaining or entrancing aesthetic experiences, but instead to
provide analytic insight into the structure of a musical signal. These works exchange the
expressive visual languages of painting and abstract cinema for the conventions of
legibility found in diagrams and music notation systems. An early example of this is
Stephen Malinowskis MusicAnimationMachine (1982-2001), a software artwork which
generated scrolling piano roll representations of MIDI sound files as a real-time
graphic accompaniment to the musics playback. The earliest versions of the Music
Animation Machine represented notes with colored bars whose vertical position
corresponded to their pitch. Later variations of Malinowskis project incorporated
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additional visual schema for representing the harmonic or dissonant qualities of musical
chords, the spans of melodic intervals, and the timbres of different instrument tracks.
(Golan Levin, 2009)
Synthesis
Audio Visual Systems have a multitude of purposes in todays modern world.
This study aims to develop an audio visual entertainment system that has high end
specs while having low cost. The public sound system we have in the country today, the
local videoke, has the essential parts of a first class audio visual system. The main
control panel of the videoke is what separates it from the rest of the rest of the audio
visual systems. This control panel, along with new and improved components, compose
the enhancements that the researchers propose with this Audio Visual Entertainment
System.