UNIT –III

CHARACTERISTICS OF SOUND:

Sound waves travel at great distances in a very short time, but as the distance increases the
waves tend to spread out. As the sound waves spread out, their energy simultaneously spreads
through an increasingly larger area. Thus, the wave energy becomes weaker as the distance from
the source is increased. Sounds may be broadly classified into two general groups. One group is
NOISE, which includes sounds such as the pounding of a hammer or the slamming of a door. The
other group is musical sounds, or TONES. The distinction between noise and tone is based on the
regularity of the vibrations, the degree of damping, and the ability of the ear to recognize
components having a musical sequence. You can best understand the physical difference between
these kinds of sound by comparing the wave shape of a musical note, depicted in view A of figure
1-13, with the wave shape of noise, shown in view B. You can see by the comparison of the two
wave shapes, that noise makes a very irregular and haphazard curve and a musical note makes a
uniform and regular curve.

Basic Characteristics of Sound

The basic fundamentals of music are by their very nature a necessary tool to use in many of the
future papers I will be presenting over time. The fundamental characteristics of sound consist of
only pitch, duration, quality and intensity, however, the character of the sequence of sounds and
its arrangement is what makes music subjectively pleasing and individually enjoyed. Let‘s take a
closer look at these four basic characteristics that comprise the foundation for everything else we
will be discussing, as related to music.
Pitch* – In music notation, pitch can be seen visually by looking at the placement of a note on a
musical staff. By comparing the location of where two or more notes are placed graphically, we
look at their relative position to one another and we know in what direction they are related to
each other, in a position of either higher or lower than another. We make a comparison of the two
notes thereby easily identifying where each note is spatially on the staff by making a visual
distinction. This is made possible through the use of notation software or by notating music by
hand. The example below shows visually the basic concept of pitch.

Each sound or tone represented by the notes in the above diagram is produced or
transformed from a visual only presentation by the notes as shown on the staff, to an audio and
visual presentation, what we hear, when played by an instrument and what we see on the staff.
Again, the notes are relative to each other, higher or lower, and we understand their relationship
by making the visual comparison of one to the other. We can see pitch visually in this way and at
the same time hear the sound in an analog or auditory way by playing the notes on an instrument
or we can do the same thing by playing a sound clip at the same time we look at the chart below.
So, before playing the notes first look at the chart and make some distinctions such as, the first
note is lower than the second note on the chart. Then click on the link and listen to the sound,
paying attention to and identifying the differences between the two notes being played.
In essence, we have two methods of determining pitch using our senses, sight and hearing.
We will limit our understanding to these two senses at this time, unless you are so inclined to pull
out your musical instrument and play the notes now. By doing this you can experience the notes in
three senses; hearing, sight and tactile feeling. However, it is important to know that through a
multiple sensory approach such as this we can learn to associate the sound of the note on the staff
and in reverse hear the note and learn how to notate music. We can also learn to sing from this
basis too.
Duration – Duration is also a simple concept whereby we make additional distinctions based upon
the linear structure we call time. In music, the duration is determined by the moment the tone
becomes audible until the moment the sound falls outside of our ability to hear it or it simply
stops. In music notation, a half note is longer than an eighth note, a quarter note is shorter in
duration than a whole note, for example.
As shown in the following chart, visually, we see notes represented by different shapes.
These shapes determine the designated amount of time they are to be played. Silence is also
represented in the chart by the funny little shapes in between the notes. They are called rests and
this is also heard as silence. Note shapes partially determine the duration of the audible sound and
rest shapes partially determine the duration of silence in music.
By playing the sound clip you can hear the difference between the tones in terms of
duration, longer or shorter. We can also hear the difference in the length of the silence, again,
longer or shorter. Remember, we are comparing one note to the other or one rest to the other.
After your visual review, please click on the link below the chart to hear the sound clip.

Here‘s another example of duration.

The notation above shows some newly presented note lengths following the eighth note in
the second measure. These are sixteenth notes. Using the vibrato legato sound samples to
demonstrate this aurally they sound all bunched together versus the prolonged half note for
example. This is another way that composers and performers can create interesting sounds by
combining different note durations.
Quality – From a church bell tower we hear the sound of the large bell ringing in the
neighborhood. Assuming the bell is playing a C note and we compare a different instrument playing
the same C note, a tuba for example, we can make the comparison between them listening and
comparing the tonal quality or timber differences between the two instruments. This exercise will
help in understanding tonal quality. Even though the pitch is the same for both notes they sound
different, in many ways.
To further explain; below is an mp3 sample of two different instruments, one following the
other. One instrument is a violin and the other is a flute, both playing the same C note or the same
pitch. The difference we hear is not in duration or in pitch but in tonal quality or timbre. This
aspect of music is broad and encompassing of the many different possibilities available from
different instruments and from the same instrument as well. The skill and artistry of the performer
also plays a significant role and highly influences the tonal quality produced by a single instrument
as does the quality and character of the instrument itself.
I have used two different tones, the C and the G (that‘s the one in the middle), to
demonstrate the tonal characteristics by comparing the sound qualities between a flute and a
violin. The last measure provides a comparison again, but this time while both instruments are
simultaneously sounding.

All sounds that we hear are made up of many overtones in addition to a fundamental tone,
unless the tone is a pure tone produced by a tuning fork or an electronic device. So, in music when
a cellist plays a note we not only hear the note as a fundamental note but we also hear the
overtones at the same time. By making sounds from different instruments and sounding them
simultaneously we hear a collection of tonal qualities that is broad in scope however, again we still
primarily hear the loudest or the fundamental tone. The spectral analysis photo below
demonstrates this point.

Spectral Analysis
Each peak is not simply a vertical line. It has many more nuances and sounds making up the
total sound we hear. The photo shows this where in between each peak we see a lot of smaller
peaks and the width of the main peaks is broad, partly contingent upon intensity and partly on
overtones.
Note: Tonal quality and overtones can be further understood visually by taking a closer look at the
first picture in this article. It is reproduced here for convenience.
3D Sound Spectrum
The concept of and study of overtones and other sound mechanisms takes us to material
and information beyond the scope of this article. Our intention here is to provide the basic
understanding of the difference in tonal quality as compared to intensity, duration and pitch.
Intensity – Intensity is a measure of the loudness of the tone. Assuming that the pitch, duration
and tonal quality are the same, we compare two or more tones based upon loudness or intensity.
One is louder or quieter than the other. When playing a piano for instance, if we strike the keys
gently we produce a quiet sound. If we strike them hard we produce a louder sound even though
the pitch is the same. Here is an audio clip comparing intensity or loudness on the flute.

Intensity can also be seen when working with a wave form editor as the photo below
shows. The larger the wave form the louder the sound. If you‘ll notice the small ―wavy line‖ in
between each of the larger wave forms in this snapshot, even though they show up on the graph, it
is likely that you do not hear the sound in these locations.

The really super cool thing about working with wave forms is that you can edit them
extensively and make unique sounds out of the originally recorded instrument. That method of
editing sound is only one of the ways in which digital sound can be manipulated and controlled.
The Five Elements of a Great Sounding System:
Clarity
When looking at acoustic quality, Clarity is the most important element. Clarity cannot be
accomplished unless you have achieved all of the other four goals. Clarity includes the ability to:
understand dialogue in movies, understand musical lyrics, hear quiet details in a soundtrack or in
music, and have sounds be realistic. Just about every characteristic of your sound system and room
can and will affect clarity.
Having excellent clarity is the pinnacle of great sound in a system.
Focus
Sit down in the ―hot seat‖ of your home theater and play your favorite music. Now close your
eyes and imagine where each instrument is located in the sound you are hearing. Every recording
is designed to place instruments and sounds in a precise (or sometimes an intentionally
nonprecise) location. Focus is the ability of your system to accurately communicate those locations
to your ears and brain.
Proper focus includes three aspects: the position of the sound in the soundfield (left to right
and front to back), the ―size‖ of the sound (does it sound ―bigger/more pronounced‖ or
―smaller/less pronounced‖ than it should), and the stability of that image (does the sound wander
around as the instrument plays different notes, for example). Finally, focus allows you to
distinguish between different sounds in the recording, assuming the recording was done in a way
that the sounds are actually distinguishable!
Envelopment
Envelopment refers to how well your system can ―surround‖ you with the sound. You may be
surprised, but a well designed and calibrated system with only two speakers is still well capable of
surrounding you with sound. A well done 5.1 or 7.1 system will do it even better.
Proper envelopment means a 360-degree soundfield with no holes or hotspots, accurate
placement of sounds within that soundfield, and the ability to accurately reproduce the sound of
the room where the recording was made.
Dynamic Range
The difference between the softest sound and loudest sound a system can reproduce is it‘s
dynamic range. Most people focus on bumping up the loud side of things (with bigger amps, etc.).
The reality is that the dynamic range of many home theaters is limited by the quietest sounds. The
softest sounds can be buried under excessive ambient noise - whether it‘s fan noise, A/C noise,
DVR hard drives, the kitchen refrigerator in the next room, or cars driving by outside the window.
The goal for dynamic range is to easily & effortlessly reproduce loud sounds while still ensuring
that quiet sounds can be easily heard.
Response
A system‘s response is a measurement of how equally every frequency is played by the system. The
goal is a smooth response from the very low end (bass) all the way up to the highest (treble)
frequencies. Examples of uneven response include:
Boomy bass: certain bass notes knocking you out of your chair while others even a few
notes higher or lower can barely be heard Not enough bass overall
Instruments sounding “wrong”
Things sounding just generally unrealistic
A system that is tiring to listen to, causing “listener fatigue” after only a short
time.
A properly tuned system will sound smooth across all frequencies, will not cause fatigue even
at higher volumes, and will result in instruments and other acoustic elements sounding natural and
realistic.

Microphones:
I. how they work
II. specifications
III. pick up patterns
 IV. typical placements
V. microphone mystique
I. How They Work.
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.
THE 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.

THE CONDENSER MICROPHONE.

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.
When the distance changes, current flows in the wire as the battery maintains the correct charge.
The amount of current is essentially proportioinal 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 microphones use plastic diaphragms.) The major disadvantage of electrets is that
they lose their charge after a few years and cease to work.

II. 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 sometmes, 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 if you are trying to record very tiny sounds, such as a turtle snapping its jaw, but
should be considered in any situation. If you put an insensitive mic on a quiet instrument, such as
an acoustic guitar, you will have to increase the gain of the mixing console, 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
varous factors. With a dymanic, 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 than you might think, especially if you place the mic very close to
instruments. (Would you put your ear in the bell of a trumpet?) You usually get a choice between
high sensitivity and high overload points, although occasionally there is 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. The really big names throw away mic capsules that don't meet their standards.
(If you buy one Neumann mic, you are paying for five!)
 No mic is perfectly linear; the best you can do is find one with distortion that complements
the sound you are trying to record. This is one of the factors of the microphone mystique discussed
later.
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 manufacturers began adding
equalizers to each input to compensate. This effort has now paid off to the point were 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. This is another part of the microphone mystique. Problems in frequency response are
mostly encountered with sounds originating behind the mic, as discussed in the next section.
Noise.
Microphones produce a very small amount of current, which makes sense when you
consider just how light the moving parts must be to accurately 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 condensor 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 line:

Current produced by the microphone will flow down one wire of the twisted pair, and back along
the other one. Any current induced in the cable from an outside source would tend to flow the
same way in both wires, and such currents cancel each other in the transformers. This system is
expensive.
Microphone Levels
As I said, microphone outputs are of necessity very weak signals, generally around -60dBm. (The
specification is the power produced by a sound pressure of 10 uBar) The output impedance will
depend on whether the mic has a transformer balanced output . If it does not, the microphone will
be labeled "high impedance" or "hi Z" and must be connected to an appropriate input. The cable
used must be kept short, less than 10 feet or so, to avoid noise problems.

If a microphone has a transformer, it will be labeled low impedance, and will work best with
a balanced input mic preamp. The cable can be several hundred feet long with no problem.
Balanced output, low impedance microphones are expensive, and generally found in professonal
applications. Balanced outputs must have three pin connectors ("Cannon plugs"), but not all mics
with those plugs are really balanced. Microphones with standard or miniature phone plugs are
high impedance. A balanced mic can be used with a high impedance input with a suitable adapter.

You can see from the balanced connection diagram that there is a transformer at the input
of the console preamp. (Or, in lieu of a transformer, a complex circuit to do the same thing.) This is
the most significant difference between professional preamplifiers and the type usually found on
home tape decks. You can buy transformers that are designed to add this feature to a consumer
deck for about $20 each. (Make sure you are getting a transformer and not just an adapter for the
connectors.) With these accessories you can use professional quality microphones, run cables over
a hundred feet with no hum, and because the transformers boost the signal somewhat, make
recordings with less noise. This will not work with a few inexpensive cassette recorders, because
the strong signal causes distortion. Such a deck will have other problems, so there is little point
trying to make a high fidelity recording with it anyway.
III. Pick Up Patterns
Many people have the misconception that microphones only pick up sound from sources
they are pointed at, much as a camera only photographs what is in front of the lens. This would be
a nice feature if we could get it, but the truth is we can only approximate that action, and at the
expense of other desirable qualities.

MICROPHONE PATTERNS
These are polar graphs of the output produced vs. the angle of the sound source. The
output is represented by the radius of the curve at the incident angle.
Omni
The simplest mic design will pick up all sound, regardless of its point of origin, and is thus
known as an omnidirectional microphone. They are very easy to use and generally have good to
outstanding frequency response. To see how these patterns are produced, here's a sidebar on
directioal microphones.
Bi-directional
It is not very difficult to produce a pickup pattern that accepts sound striking the front or
rear of the diaphragm, but does not respond to sound from the sides. This is the way any
diaphragm will behave if sound can strike the front and back equally. The rejection of undesired
sound is the best achievable with any design, but the fact that the mic accepts sound from both
ends makes it difficult to use in many situations. Most often it is placed above an instrument.
Frequency response is just as good as an omni, at least for sounds that are not too close to the
microphone.
Cardioid
This pattern is popular for sound reinforcement or recording concerts where audience noise
is a possible problem. The concept is great, a mic that picks up sounds it is pointed at. The reality is
different. The first problem is that sounds from the back are not completely rejected, but merely
reduced about 10-30 dB. This can surprise careless users. The second problem, and a severe one, is
that the actual shape of the pickup pattern varies with frequency. For low frequencies, this is an
omnidirectional microphone. A mic that is directional in the range of bass instruments will be fairly
large and expensive. Furthermore, the frequency response for signals arriving from the back and
sides will be uneven; this adds an undesired coloration to instruments at the edge of a large
ensemble, or to the reverberation of the concert hall.

A third effect, which may be a problem or may be a desired feature, is that the microphone
will emphasize the low frequency components of any source that is very close to the diaphragm.
This is known as the "proximity effect", and many singers and radio announcers rely on it to add
"chest" to a basically light voice. Close, in this context, is related to the size of the microphone, so
the nice large mics with even back and side frequency response exhibit the strongest presence
effect. Most cardioid mics have a built in lowcut filter switch to compensate for proximity.
Missetting that switch can cause hilarious results. Bidirectional mics also exhibit this phenomenon.
Tighter Patterns
It is posible to exaggerate the directionality of cardioid type microphones, if you don't mind
exaggerating some of the problems. The Hypercardioid pattern is very popular, as it gives a better
overall rejection and flatter frequency response at the cost of a small back pickup lobe. This is
often seen as a good compromise between the cardioid and bidirectional patterns. A "shotgun"
mic carries these techniques to extremes by mounting the diaphragm in the middle of a pipe. The
shotgun is extremely sensitive along the main axis, but posseses pronounced extra lobes which
vary drastically with frequency. In fact, the frequency response of this mic is so bad it is usually
electronically restricted to the voice range, where it is used to record dialogue for film and video.
Stereo microphones
You don't need a special microphone to record in stereo, you just need two (see below). A
so called stereo microphone is really two microphones in the same case. There are two kinds:
extremely expensive professional models with precision matched capsules, adjustable capsule
angles, and remote switching of pickup patterns; and very cheap units (often with the capsules
oriented at 180 deg.) that can be sold for high prices because they have the word stereo written on
them.
IV. Typical Placement
Single microphone use
Use of a single microphone is pretty straightforward. Having chosen one with appropriate
sensitivity and pattern, (and the best distortion, frequency response, and noise characteristics you
can afford), you simply mount it where the sounds are. The practical range of distance between the
instrument and the microphone is determined by the point where the sound overloads the
microphone or console at the near end, and the point where ambient noise becomes
objectionable at the far end. Between those extremes it is largely a matter of taste and
experimentation.

If you place the microphone close to the instrument, and listen to the results, you will find
the location of the mic affects the way the instrument sounds on the recording. The timbre may be
odd, or some notes may be louder than others. That is because the various components of an
instrument's sound often come from different parts of the instrument body (the highest note of a
piano is nearly five feet from the lowest), and we are used to hearing an evenly blended tone. A
close in microphone will respond to some locations on the instrument more than others because
the difference in distance from each to the mic is proportionally large. A good rule of thumb is that
the blend zone starts at a distance of about twice the length of the instrument. If you are recording
several instruments, the distance between the players must be treated the same way.

If you place the microphone far away from the instrument, it will sound as if it is far away
from the instrument. We judge sonic distance by the ratio of the strength of the direct sound from
the instrument (which is always heard first) to the strength of the reverberation from the walls of
the room. When we are physically present at a concert, we use many cues beside the sounds to
keep our attention focused on the performance, and we are able to ignore any distractions there
may be. When we listen to a recording, we don't have those visual clues to what is happening, and
find anything extraneous that is very audible annoying. For this reason, the best seat in the house
is not a good place to record a concert. On the other hand, we do need some reverberation to
appreciate certain features of the music. (That is why some types of music sound best in a stone
church) Close microphone placement prevents this. Some engineers prefer to use close miking
techniques to keep noise down and add artificial reverberation to the recording, others solve the
problem by mounting the mic very high, away from audience noise but where adequate
reverberation can be found.
Stereo
Stereo sound is an illusion of spaciousness produced by playing a recording back through
two speakers. The success of this illusion is referred to as the image. A good image is one in which
each instrument is a natural size, has a distinct location within the sound space, and does not move
around. The main factors that establish the image are the relative strength of an instrument's
sound in each speaker, and the timing of arrival of the sounds at the listener's ear. In a studio
recording, the stereo image is produced artificially. Each instrument has its own microphone, and
the various signals are balanced in the console as the producer desires. In a concert recording,
where the point is to document reality, and where individual microphones would be awkward at
best, it is most common to use two mics, one for each speaker.
Spaced microphones
The simplest approach is to assume that the speakers will be eight to ten feet apart, and
place two microphones eight to ten feet apart to match. Either omnis or cardioids will work. When
played back, the results will be satisfactory with most speaker arrangements. (I often laugh when I
attend concerts and watch people using this setup fuss endlessly with the precise placement of the
mics. This technique is so forgiving that none of their efforts will make any practical difference.)
The big disavantage of this technique is that the mics must be rather far back from the ensemble-
at least as far as the distance from the leftmost performer to the rightmost. Otherwise, those
instruments closest to the microphones will be too prominent. There is usually not enough room
between stage and audience to achieve this with a large ensemble, unless you can suspend the
mics or have two very tall stands.
Coincident cardioids
There is another disadvantage to the spaced technique that appears if the two channels are
ever mixed together into a monophonic signal. (Or broadcast over the radio, for similar reasons.)
Because there is a large distance between the mics, it is quite possible that sound from a particular
instrument would reach each mic at slightly different times. (Sound takes 1 millisecond to travel a
foot.) This effect creates phase differences between the two channels, which results in severe
frequency response problems when the signals are combined. You seldom actually lose notes from
this interference, but the result is an uneven, almost shimmery sound. The various coincident
techniques avoid this problem by mounting both mics in almost the same spot.

This is most often done with two cardioid microphones, one pointing slightly left, one
slightly right. The microphones are often pointing toward each other, as this places the diaphragms
within a couple of inches of each other, totally eliminating phase problems. No matter how they
are mounted, the microphone that points to the left provides the left channel. The stereo effect
comes from the fact that the instruments on the right side are on-axis for the right channel
microphone and somewhat off-axis (and therefore reduced in level) for the other one. The angle
between the microphones is critical, depending on the actual pickup pattern of the microphone. If
the mics are too parallel, there will be little stereo effect. If the angle is too wide, instruments in
the middle of the stage will sound weak, producing a hole in the middle of the image. [Incidentally,
to use this technique, you must know which way the capsule actually points. There are some very
fine German cardioid microphones in which the diaphragm is mounted so that the pickup is from
the side, even though the case is shaped just like many popular end addressed models. (The front
of the mic in question is marked by the trademark medallion.) I have heard the results where an
engineer mounted a pair of these as if the axis were at the end. You could hear one cello player
and the tympani, but not much else.]

You may place the microphones fairly close to the instruments when you use this
technique. The problem of balance between near and far instruments is solved by aiming the mics
toward the back row of the ensemble; the front instruments are therefore off axis and record at a
lower level. You will notice that the height of the microphones becomes a critical adjustment. M.S.
The most elegant approach to coincident miking is the M.S. or middle-side technique.
This is usually done with a stereo microphone in which one element is omnidirectional, and the
other bidirectional. The bidirectional element is oriented with the axis running parallel to the
stage, rejecting sound from the center. The omni element, of course, picks up everything. To
understand the next part, consider what happens as instrument is moved on the stage. If the
instrument is on the left half of the stage, a sound would first move the diaphragm of the
bidirectional mic to the right, causing a positive voltage at the output. If the instrument is moved
to center stage, the microphone will not produce any signal at all. If the instrument is moved to the
right side, the sound would first move the diaphragm to the left, producing a negative volage. You
can then say that instruments on one side of the stage are 180 degrees out of phase with those on
the other side, and the closer they are to the center, the weaker the signal produced.

Now the signals from the two microphones are not merely kept in two channels and played
back over individual speakers. The signals are combined in a circuit that has two outputs; for the
left channel output, the bidirectional output is added to the omni signal. For the right channel
output, the bidirectional output is subtracted from the omni signal. This gives stereo, because an
instrument on the right produces a negative signal in the bidirectional mic, which when added to
the omni signal, tends to remove that instrument, but when subtracted, increases the strength of
the instrument. An instrument on the left suffers the opposite fate, but instruments in the center
are not affected, because their sound does not turn up in the bidirectional signal at all.

M.S. produces a very smooth and accurate image, and is entirely mono compatabile. The
only reason it is not used more extensively is the cost of the special microphone and decoding
circuit, well over $1,000.
Large ensembles
The above techniques work well for concert recordings in good halls with small ensembles.
When recording large groups in difficult places, you will often see a combination of spaced and
coincident pairs. This does produce a kind of chorusing when the signals are mixed, but it is an
attractive effect and not very different from the sound of string or choral ensembles any way.
When balance between large sections and soloists cannot be acheived with the basic setup, extra
microphones are added to highlight the weaker instruments. A very common problem with large
halls is that the reverberation from the back seems late when compared to the direct sound taken
at the edge of the stage. This can be helped by placing a mic at the rear of the audience area to get
the ambient sound into the recording sooner.
Studio techniques
A complete description of all of the procedures and tricks encountered in the recording
studio would fill several books. These are just a few things you might see if you dropped in on the
middle of a session.
Individual mics on each instrument.
This provides the engineer with the ability to adjust the balance of the instruments at the
console, or, with a multitrack recorder, after the musicians have gone home. There may be eight or
nine mics on the drum set alone.
Close mic placement.
The microphones will usually be placed rather close to the instruments. This is partially to
avoid problems that occur when an instrument is picked up in two non-coincident mics, and
partially to modify the sound of the instruments (to get a "honky-tonk" effect from a grand piano,
for instance).
Acoustic fences around instruments, or instruments in separate rooms.
The interference that occurs when when an instrument is picked up by two mics that are
mixed is a very serious problem. You will often see extreme measures, such as a bass drum stuffed
with blankets to muffle the sound, and then electronically processed to make it sound like a drum
again.
Everyone wearing headphones.
Studio musicians often play to "click tracks", which are not recorded metronomes, but
someone tapping the beat with sticks and occasionally counting through tempo changes. This is
done when the music must be synchronized to a film or video, but is often required when the
performer cannot hear the other musicians because of the isolation measures described above.
20 or 30 takes on one song.
Recordings require a level of perfection in intonation and rhythm that is much higher than
that acceptable in concert. The finished product is usually a composite of several takes.
Pop filters in front of mics.
Some microphones are very sensitive to minor gusts of wind--so sensitive in fact that they
will produce a loud pop if you breath on them. To protect these mics (some of which can actually
be damaged by blowing in them) engineers will often mount a nylon screen between the mic and
the artist. This is not the most common reason for using pop filters though:
Vocalists like to move around when they sing; in particular, they will lean into microphones. If the
singer is very close to the mic, any motion will produce drastic changes in level and sound quality.
(You have seen this with inexpert entertainers using hand held mics.) Many engineers use pop
filters to keep the artist at the proper distance. The performer may move slightly in relation to the
screen, but that is a small proportion of the distance to the microphone.
V. The Microphone Mystique
There is an aura of mystery about microphones. To the general public, a recording engineer
is something of a magician, privy to a secret arcana, and capable of supernatural feats. A few
modern day engineers encourage this attitude, but it is mostly a holdover from the days when
studio microphones were expensive and fragile, and most people never dealt with any electronics
more complex than a table radio. There are no secrets to recording; the art is mostly a
commonsense application of the principles already discussed in this paper. If there is an arcana, it
is an accumulation of trivia achieved through experience with the following problems:
Matching the microphone to the instrument.
There is no wrong microphone for any instrument. Every engineer has preferences, usually
based on mics with which he is familiar. Each mic has a unique sound, but the differences between
good examples of any one type are pretty minor. The artist has a conception of the sound of his
instrument, (which may not be accurate) and wants to hear that sound through the speakers.
Frequency response and placement of the microphone will affect that sound; sometimes you need
to exaggerate the features of the sound the client is looking for.
Listening the proper way.
It is easy to forget that the recording engineer is an illusionist- the result will never be
confused with reality by the listener. Listeners are in fact very forgiving about some things. It is
important that the engineer be able to focus his attention on the main issues and not waste time
with interesting but minor technicalities. It is important that the engineer know what the main
issues are. An example is the noise/distortion tradeoff. Most listeners are willing to ignore a small
amount of distortion on loud passages (in fact, they expect it), but would be annoyed by the extra
noise that would result if the engineer turned the recording level down to avoid it. One technique
for encouraging this attention is to listen to recordings over a varitey of sound systems, good and
bad.

Learning for yourself.

Many students come to me asking for a book or a course of study that will easily make
them a member of this elite company. There are books, and some schools have courses in
recording, but they do not supply the essential quality the professional recording engineer needs,
which is experience.
A good engineer will have made hundreds of recordings using dozens of different
microphones. Each session is an opportunity to make a new discovery. The engineer will make
careful notes of the setup, and will listen to the results many times to build an association between
the technique used and the sound achieved. Most of us do not have access to lots of professional
microphones, but we could probably afford a pair of general purpose cardioids. With about $400
worth of mics and a reliable tape deck, it is possible to learn to make excellent recordings. The trick
is to record everything that will sit still and make noise, and study the results: learn to hear when
the mic is placed badly and what to do about it. When you know all you can about your mics, buy a
different pair and learn those. Occasionally, you will get the opportunity to borrow mics. If
possible, set them up right alongside yours and make two recordings at once. It will not be long
before you will know how to make consistently excellent recordings under most conditions.

Audio amplifier:
An audio amplifier is an electronic amplifier that amplifies low-power audio signals (signals
composed primarily of frequencies between 20 - 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 like pre-
amplification, equalization, tone control, mixing/effects, or audio 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.

MIDI Messages:
The MIDI Message specification (or "MIDI Protocol") is probably the most important part of MIDI.
Though originally intended just for use over MIDI Cables to connect two keyboards, MIDI
messages are now used inside computers and cell phones to generate music, and transported over
any number of professional and consumer interfaces (USB, FireWire, etc.) to a wide variety of
MIDI-equipped devices. There are different message groups for different applications, only some of
which are we able to explain here.
MIDI is a music description language in digital (binary) form. It was designed for use with
keyboard-based musical instruments, so the message structure is oriented to performance events,
such as picking a note and then striking it, or setting typical parameters available on electronic
keyboards. For example, to sound a note in MIDI you send a "Note On" message, and then assign
that note a "velocity", which determines how loud it plays relative to other notes. You can also
adjust the overall loudness of all the notes with a Channel Volume" message. Other MIDI messages
include selecting which instrument sounds to use, stereo panning, and more.
The first specification (1983) did not define every possible "word" that can be spoken in MIDI ,
nor did it define every musical instruction that might be desired in an electronic performance. So
over the past 20 or more years, companies have enhanced the original MIDI specification by
defining additional performance control messages, and creating companion specifications which
include:
MIDI Machine Control
MIDI Show Control
MIDI Time Code
General MIDI
Downloadable Sounds
Scalable Polyphony MIDI
Alternate Applications MIDI Machine Control and MIDI Show Control are interesting extensions
because instead of addressing musical instruments they address studio recording equipment (tape
decks etc) and theatrical control (lights, smoke machines, etc.).
MIDI is also being used for control of devices where standard messages have not been defined
by MMA, such as with audio mixing console automation.
MIDI Cables & Connectors:
Many different "transports" can be used for MIDI messages. The speed of the transport
determines how much MIDI data can be carried, and how quickly it will be received.
Each transport has its own performance characteristics which might make some
difference in specific applications, but in general the transport is the least important part of
MIDI , as long as it allows you to connect all the devices you want use!
5-Pin MIDI DIN
Using a 5-pin "DIN" connector, the MIDI DIN transport was developed back in 1983, so it
is slow compared to common high-speed digital transports available today, like USB, FireWire,
and Ethernet. But MIDI-DIN is almost always still used on most MIDI-equipped devices because
it adequately handles communication speed for one device. IF you want to connect one MIDI
device to another (not a computer), MIDI cables are still the best way to go.
It used to be that connecting a MIDI device to a computer meant installing a "sound
card" or "MIDI interface" in order to have a MIDI DIN connector on the computer. Because of
space limitations, most such cards did not have actual 5-Pin DIN connectors on the card, but
provided a special cable with 5-Pin DINs (In and Out) on one end (often connected to the
"joystick port"). All such cards need "driver" software to make the MIDI connection work, but
there are a few standards that companies follow, including "MPU-401" and "SoundBlaster".
Even with those standards, however, making MIDI work could be a major task.
Over a number of years the components of the typical sound card and MIDI interface
(including the joystick port) became standard on the motherboard of most PCs, but this did not
make configuring them any easier.
Serial, Parallel, and Joystick Ports
Before USB and FireWire, personal computers were all generally equipped with serial,
parallel, and (possibly) joystick ports, all of which have been used for connecting MIDI-equipped
instruments (through special adapters). Though not always faster than MIDI-DIN, these
connectors were already available on computers and that made them an economical alternative
to add-on cards, with the added benefit that in general they already worked and did not need
special configuration.
The High Speed Serial Ports such as the "mini-DIN" ports available on early Macintosh
computers support communication speeds roughly 20 times faster than MIDI-DIN, making it
also possible for companies to develop and market "multiport" MIDI interfaces that allowed
connecting multiple MIDI-DINs to one computer. In this manner it became possible to have the
computer address many different MIDI-equipped devices at the same time. Recent multi-port
MIDI interfaces use even faster USB or FireWire ports to connect to the computer.
USB and FireWire
All recent computers are equipped with either USB and/or FireWire connectors, and
these are now the most common means of connecting MIDI devices to computers (using
appropriate format adapters). Adapters can be as simple as a short cable with USB on one end
and MIDI DIN on the other, or as complex as a 19 inch rack mountable CPU with dozens of MIDI
and Audio In and Out ports. The best part is that USB and FireWire are "plug-and-play"
interfaces which means they generally configure themselves. In most cases, all you need to do is
plug in your USB or FireWire MIDI interface and boot up some MIDI software and off you go.
Current USB technology generally supports communication between a host (PC) and a
device, so it is not possible to connect to USB devices to each other as it is with two MIDI DIN
devices. (This may change sometime in the future with new versions of USB). Since
communications all go through the PC, any two USB MIDI devices can use different schemes for
packing up MIDI messages and sending them over USB... the USB device's driver on the host
knows how that device does it, and will convert the MIDI messages from USB back to MIDI at
the host. That way all USB MIDI devices can talk to each other (through the host) without
needing to follow one specification for how they send MIDI data over USB.
Most FireWire MIDI devices also connect directly to a PC with a host device driver and
so can talk to other FireWire MIDI devices even if they use a proprietary method for formatting
their MIDI data. But FireWire supports "peer-to-peer" connections, so MMA has produced a
specification for MIDI overIEEE-1394(FireWire), which is available for download on this site (and
incorporated in IEC-61883 part
5).
Ethernet
If you are connecting a number of MIDI instruments to one or more computers, using
Ethernet seems like a great solution. In the MIDI industry there is not yet agreement on the
market desire for MIDI over Ethernet, nor on the net value of the benefits vs. challenges of
using Ethernet, and so there is currently no MMA standard for MIDI over Ethernet.
However, other Standard Setting Organizations have specifications for MIDI Over Ethernet,
and we think it appropriate that people know about those solutions. There are also proprietary
solutions for MIDI Over Ethernet, but because they are not open standards they are not
appropriate for discussion by
MMA.
IETF RTP-MIDI
The IETF RTP Payload Format for MIDIsolution has received extensive modification in response
to comments by MMA-members, and is also the foundation of Apple's own MIDI Over Ethernet
solution. Though neither solution has been officially adopted or endorsed in any way by MMA,
both technologies have stood up to MMA member scrutiny and so are likely to appear (in one
manner or another) in future MIDI hardware and/or software products.
IEEE Ethernet AVB
For the past several years, the IEEE has been developing protocols for low-latency audio and
video transport over Ethernet with high quality of service. These protocols are known as
Audio/Video Bridging, or AVB, and are part of the larger IEEE 802.1 Working Group, which
develops networking standards that enable interoperability of such ubiquitous devices as
Ethernet switches. The AVB protocols provide precision time synchronization and stream
bandwidth reservation at the network level.
The AVB protocols do not provide a standard means for interoperable communication of
content such as a live video stream. Utilizing the 802.1 AVB protocols, the IEEE P1722 AVB
Transport Protocol (AVBTP) draft standard provides the necessary content encapsulation in an
evolutionary manner by adopting the existing IEEE 1394 (Firewire) audio and video streaming
mechanisms already in use by millions of devices. However, AVBTP is not limited to bridging IEEE
1394 content, as it provides extensibility to encapsulate new and different media formats.
The MMA collaborated with the IEEE P1722 working group to enable transport of MIDI and
any future content format defined by the MMA over IEEE P1722 networks. The P1722 standard
defines MIDI 1.0 content within this protocol by referencing an MMA-authored document. The
MMA has not yet published that document, but plans to do so in the near future.
Basic MIDI Connections:
 Let's first take a look at what you need to get your MIDI (Recording) system setup:
Hardware:
Computer - either PC or laptop.
MIDI keyboard or USB keyboard with or without sounds.
Soundcard either fitted inside your computer or external soundcard.
USB or MIDI cable(s).
Software
Install drivers for soundcard (better to download latest version from manufacturer). SEARCH
TIP: Go to Google and search "Model number of card + drivers download". i.e. If your
soundcard is called "SoundcardXYZ" then type "SoundcardXYZ drivers download"
(without the quotes) into Google. There is a high probability that Google will give you
the exact page you need for the latest drivers.
Install latest drivers for keyboard (if needed) - more common for USB keyboards.
Install MIDI Sequencing package - Cubase LE
Brief Connection Concept
IMPORTANT MIDI CONNECTIONS - Always connect MIDI OUT from one device to MIDI IN
on the other or vice-versa.
If you have a computer a keyboard or any external sound modules then connect as
shown below:

If you have an additional module to add to the setup above then simply connect a MIDI
OUT from the sound module to the additional module (MIDI IN).
Having a large number of MIDI chain connections is not advisable and not really practical
when it comes to controlling your MIDI channels from within the sequencing software -
The system above only allows you 16 channels of sounds playing simultaneously. Of
course, this depends on the equipment, but let's just assume that the keyboard and
module are multi-timbral and can play 16 channels at the same time. Because of the
setup above you are limited.
MIDI Thru Box - A MIDI thru box is advisable on bigger systems to allow more than 16
channels be played back simultaneously - the MIDI output of each MIDI port on the Thru box
is controlled from within the sequencing package. For example, let's say we are using
Cubase. Track 1 is playing MIDI channel 1, Track 2 plays MIDI channel 2 etc. etc. The MIDI
output of MIDI channel 1 is routed to the MIDI Thru Box - Port 1, The MIDI output of MIDI
channel 2 is routed to the MIDI Port 2. So, for 4 MIDI ports connected to 4 different devices
you can have 64 MIDI channels!

Connect
Assuming you have installed your software and hardware correctly you are literally steps
away from completing your MIDI setup!
If you have a USB Keyboard then connect it to your USB port on your computer. Load up
your MIDI sequencing software and see if you can see the MIDI input from your
sequencing software. Cubase LE is great for this and will show if you connection has
been establised by displaying a light trigger when ever you play your keyboard.
 If you have a MIDI keyboard then connect the MIDI cable from your MIDI Out on the
keyboard to MIDI In on your soundcard. As above, if you have Cubase installed then it
will display a connection if you depress a key on the keyboard.
Want to play the sounds on your keyboard?
If you want to playback the sounds of your keyboard then you have to connect the MIDI
Out from your soundcard to the MIDI In of your keyboard.
So, when recording, you play out the notes from your keyboard into your computer
(sequencer) then after you've finished recording the computer will playback MIDI
recorded information back from the MIDI Out port of the computer to the MIDI In of
your keyboard. It's quite simple! Multitrack MIDI recording - Simple! Same as above,
keep recording and pre-recorded tracks will playback when you are recording additional
tracks.
This is a generic description of your MIDI setup and you may have to customise it slightly for
your own setup since very few MIDI setups are the same it's almost impossible to give a direct
answer to this popular topic.

Sound card:
A sound card (also known as an audio card) is a computer expansion card that facilitates the
input and output of audio signals to and from a computer under control of computer programs.
Typical uses of sound cards include providing the audio component for multimedia applications
such as music composition, editing video or audio, presentation, education, and entertainment
(games). Many computers have sound capabilities built in, while others require additional
expansion cards to provide for audio capability.

Sound cards usually feature a digital-to-analog converter (DAC),

which converts recorded or generated digital data into an analog format. The output signal is
connected to an amplifier, headphones, or external device using standard interconnects, such as
a TRS connector or an RCA connector. If the number and size of connectors is too large for the
space on the backplate the connectors will be off-board, typically using a breakout box, or an
auxiliary backplate. More advanced cards usually include more than one sound chip to provide
for higher data rates and multiple simultaneous functionality, eg between digital sound
production and synthesized sounds (usually for real-time generation of music and sound effects
using minimal data and CPU time). Digital sound reproduction is usually done with multichannel
DACs, which are capable of multiple digital samples simultaneously at different pitches and
volumes, or optionally applying real-time effects like filtering or distortion. Multi-channel digital
sound playback can also be used for music synthesis when used with a compliance, and even
multiple-channel emulation. This approach has become common as manufacturers seek to
simplify the design and the cost of sound cards.
Most sound cards have a line in connector for signal from a cassette tape recorder or similar
sound source. The sound card digitizes this signal and stores it (under control of appropriate
matching computer software) on the computer's hard disk for storage, editing, or further
processing. Another common external connector is the microphone connector, for use by a
microphone or other low level input device. Input through a microphone jack can then be used
by speech recognition software or for Voice over IP applications.
Audio file format:
An audio file format is a file format for storing audio data on a computer system. It can be a raw
bitstream, but it is usually a container format or an audio data format with defined storage layer.
The general approach towards storing digital audio is to sample the audio voltage which, on
playback, would correspond to a certain level of signal in an individual channel with a certain
resolution—the number of bits per sample—in regular intervals (forming the sample rate). This
data can then be stored uncompressed, or compressed to reduce the file size
Types of formats:
It is important to distinguish between a file format and a CODEC. A codec performs the encoding
and decoding of the raw audio data while the data itself is stored in a file with a specific audio
file format. Most of the publicly documented audio file formats can be created with one of two
or more encoders or codecs. Although most audio file formats support only one type of audio
data (created with an audio coder), a multimedia container format (as MKV or AVI) may support
multiple types of audio and video data.
There are three major groups of audio file formats:
Uncompressed audio formats, such as WAV, AIFF, AU or raw header-less PCM; formats
with lossless compression, such as FLAC, Monkey's Audio (filename extension APE),
WavPack (filename extension WV), Shorten, TTA, ATRAC Advanced Lossless, Apple
Lossless, MPEG-4 SLS, MPEG-4 ALS, MPEG-4 DST, Windows Media Audio Lossless (WMA
Lossless). formats with lossy compression, such as MP3, Vorbis, Musepack,
AAC, ATRAC and lossy Windows Media Audio (WMA).
Uncompressed audio formats
There is one major uncompressed audio format, PCM which is usually stored as a .wav on
Windows or as .aiff on Mac OS. WAV and AIFF are flexible file formats designed to store more or
less any combination of sampling rates or bitrates. This makes them suitable file formats for
storing and archiving an original recording. There is another uncompressed audio format which
is .cda (Audio CD Track) .cda is from a music CD and is 0% compressed.
The AIFF format is based on the IFF format. The WAV format is based on the RIFF file format, which is
similar to the IFF format.
BWF (Broadcast Wave Format) is a standard audio format created by the European Broadcasting
Union as a successor to WAV. BWF allows metadata to be stored in the file. See European
Broadcasting Union: Specification of the Broadcast Wave Format (EBU Technical document 3285,
July 1997). This is the primary recording format used in many professional audio workstations in
the television and film industry. BWF files include a standardized Timestamp reference which
allows for easy synchronization with a separate picture element. Stand-alone, file based, multi-
track recorders from Sound Devices, Zaxcom, HHB USA, Fostex, and Aaton all use BWF as their
preferred format.
Lossless compressed audio formats
A lossless compressed format requires much more processing time than an uncompressed format
but is more efficient in space usage.
Uncompressed audio formats encode both sound and silence with the same number of bits per
unit of time. Encoding an uncompressed minute of absolute silence produces a file of the same
size as encoding an uncompressed minute of symphonic orchestra music. In a lossless
compressed format, however, the music would occupy a marginally smaller file and the silence
take up almost no space at all.
Lossless compression formats (such as the most widespread FLAC, WavPack. Monkey's Audio,
ALAC/Apple Lossless) provide a compression ratio of about 2:1. Development in lossless
compression formats aims to reduce processing time while maintaining a good compression
ratio.
Free and open file formats
wav – standard audio file container format used mainly in Windows PCs. Commonly
used for storing uncompressed (PCM) , CD-quality sound files, which means that they
can be large in size—around 10 MB per minute. Wave files can also contain data
encoded with a variety of (lossy) codecs to reduce the file size
Open file formats
gsm– designed for telephony use in Europe, gsm is a very practical format for telephone
quality voice. It makes a good compromise between file size and quality. Note that wav
files can also be encoded with the gsm codec.
dct– A variable codec format designed for dictation. It has dictation header information
and can be encrypted (often required by medical confidentiality laws).
vox– the vox format most commonly uses the Dialogic ADPCM (Adaptive Differential
Pulse Code Modulation) codec.
Proprietary formats
mp3– MPEG Layer-3 format is the most popular format for downloading and storing
music. By eliminating portions of the audio file that are less audible, mp3 files are
compressed to roughly one-tenth the size of an equivalent PCM file sacrificing quality.
aac– the Advanced Audio Coding format is based on the MPEG2 and MPEG4 standards.
aac files are usually ADTS or ADIF containers.
mp4/m4a– MPEG-4 audio most often AAC but sometimes MP2/MP3, MPEG-4
SLS, CELP, HVXC and other audio object types defined in MPEG-4 Audio
The .ram file contains no audio data itself. dss– Digital Speech Standard files
are an Olympus proprietary format. It is a fairly old and poor codec. Gsm or mp3 are
generally preferred where the recorder allows. It allows additional data to be held in the
file header. msv– a Sony proprietary format for Memory Stick compressed voice files. dvf–
a Sony proprietary format for compressed voice files; commonly used by Sony dictation
recorders.
IVS – A proprietary version with Digital Rights Management developed by 3D Solar UK
Ltd for use in music downloaded from their Tronme Music Store and interactive music
and video player. m4p– A proprietary version of AAC in MP4 with Digital Rights
Management developed by Apple for use in music downloaded from their iTunes Music
Store.
iklax– An iKlax Media proprietary format, the iKlax format is a multi-track digital audio
format allowing various actions on musical data, for instance on mixing and volumes
 arrangements. mxp4– a Musinaut proprietary format allowing play of different
versions (or skins) of the same song. It allows various interactivity scenarios between the
artist and the end user.
3gp- multimedia container format can contain proprietary formats as AMR, AMR-WBor
AMRWB+, but also some open formats amr- AMR-NB audio, used primarily for speech
awb- AMR-WB audio, used primarily for speech
Codec:
A codec is a device or computer program capable of encoding and/or decoding a
digitaldata stream or signal. The word codec is a portmanteau of 'compressor-decompressor' or,
more commonly, 'coder-decoder'. A codec (the program) should not be confused with a coding or
compression format or standard – a format is a document (the standard), a way of storing data,
while a codec is a program (an implementation) which can read or write such files. In practice
"codec" is sometimes used loosely to refer to formats, however.
A codec encodes a data stream or signal for transmission, storage or encryption, or
decodes it for playback or editing. Codecs are used in videoconferencing, streaming media and
video editing applications. A video camera's analog-to-digital converter (ADC) converts its analog
signals into digital signals, which are then passed through a video compressor for digital
transmission or storage. A receiving device then runs the signal through a video decompressor,
then a digital-to-analog converter (DAC) for analog display. The term codec is also used as a
generic name for a video conferencing unit.
Media codecs:
Codecs are often designed to emphasize certain aspects of the media, or their use, to be
encoded. For example, a digital video (using a DV codec) of a sports event needs to encode
motion well but not necessarily exact colors, while a video of an art exhibit needs to perform
well encoding color and surface texture.
Audio codecs for cell phones need to have very low latency between source encoding and
playback; while audio codecs for recording or broadcast can use high-latency audio compression
techniques to achieve higher fidelity at a lower bit-rate.
There are thousands of audio and video codecs ranging in cost from free to hundreds of
dollars or more. This variety of codecs can create compatibility and obsolescence issues. By
contrast, raw uncompressed PCM audio (44.1 kHz, 16 bit stereo, as represented on an audio CD
or in a .wav or .aiff file) is a standard across multiple platforms.
Many multimedia data streams contain both audio and video, and often some metadata
that permit synchronization of audio and video. Each of these three streams may be handled by
different programs, processes, or hardware; but for the multimedia data streams to be useful in
stored or transmitted form, they must be encapsulated together in a container format.
Lower bit rate codecs allow more users, but they also have more distortion. Beyond the
initial increase in distortion, lower bit rate codecs also achieve their lower bit rates by using more
complex algorithms that make certain assumptions, such as those about the media and the packet
loss rate. Other codecs may not make those same assumptions. When a user with a low bit-rate
codec talks to a user with another codec, additional distortion is introduced by each transcoding.
The notion of AVI being a codec is incorrect as AVI is a container format, which many
codecs might use (although not to ISO standard). There are also other well-known containers
such as Ogg, ASF, QuickTime, RealMedia, Matroska, DivX Media Format and containers defined
as ISO standards, such as MPEG transport stream, MPEG program stream, MP4 and ISO base
media file format.
Audio player (software):
An audio player is a kind of media player for playing back digital audio, including optical
discs such as CDs, SACDs, DVD-Audio, HDCD, audio files and streaming audio.
In addition to VCR-like functions like playing, pausing, stopping, rewinding, and forwarding,
some common functions include playlisting, tagging format support, and equalizer.
Many of the audio players also support simple playback of digital videos in which we can also
run movies.
Digital audio player:
Digital audio player, shortened to DAP, MP3 player or, rarely, as an OGG player, is a
consumer electronic device that stores, organizes and plays digital audio files. In contrast, analog
audio players play music from cassette tapes, or records. Portable devices that also play video
and text are referred to as portable media players.
Sound recording and reproduction:

Sound recording and reproduction is an electrical or mechanical inscription and

recreation of sound waves, such as spoken voice, singing, instrumental music, or sound effects.
The two main classes of sound recording technology are analog recording and digital recording.
Acoustic analog recording is achieved by a small microphone diaphragm that can detect changes
in atmospheric pressure (acoustic sound waves) and record them as a graphic representation of
the sound waves on a medium such as a phonograph (in which a stylus senses grooves on a
record). In magnetic tape recording, the sound waves vibrate the microphone diaphragm and
are converted into a varying electric current, which is then converted to a varying magnetic field
by an electromagnet, which makes a representation of the sound as magnetized areas on a
plastic tape with a magnetic coating on it. Analog sound reproduction is the reverse process,
with a bigger loudspeaker diaphragm causing changes to atmospheric pressure to form acoustic
sound waves. Electronically generated sound waves may also be recorded directly from devices
such as an electric guitar pickup or a synthesizer, without the use of acoustics in the recording
process other than the need for musicians to hear how well they are playing during recording
sessions.
Digital recording and reproduction converts the analog sound signal picked up by the
microphone to a digital form by a process of digitization, allowing it to be stored and transmitted
by a wider variety of media. Digital recording stores audio as a series of binary numbers
representing samples of the amplitude of the audio signal at equal time intervals, at a sample
rate so fast that the human ear perceives the result as continuous sound. Digital recordings are
considered higher quality than analog recordings not necessarily because they have higher
fidelity (wider frequency response or dynamic range), but because the digital format can prevent
much loss of quality found in analog recording due to noise and electromagnetic interference in
playback, and mechanical deterioration or damage to the storage medium. A digital audio signal
must be reconverted to analog form during playback before it is applied to a loudspeaker or
earphones.
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 Armstrong published studies about the fundamental operation of the triode
vacuum tube before World War I, inventors like Orlando R. Marsh and his Marsh Laboratories,
as well as 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. 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.
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 Armstrong'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
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, the Victor Talking Machine Company, and
the Columbia Phonograph Co. (Edison was left out due to their decreasing market share and a
stubborn Thomas Edison). 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 Pathé 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. Although they were contemplating a deal,
Victor Talking Machine was apprised of the new Columbia deal, so they too quickly signed.
Columbia made its first released 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
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.The American Federation of Musicians took out ads in newspapers,
protesting the replacement of real musicians with mechanical playing devices, especially in
theatres.
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.
Audio and Multimedia
Multimedia content on the Web, by its definition - including or involving the use of several
media - would seem to be inherently accessible or easily made accessible.
However, if the information is audio, such as a RealAudio feed from a news conference or
the proceedings in a courtroom, a person who is deaf or hard of hearing cannot access that
content unless provision is made for a visual presentation of audio content. Similarly, if the
content is pure video, a blind person or a person with severe vision loss will miss the message
without the important information in the video being described.
Remember from Section 2 that to be compliant with Section 508, you must include text
equivalents for all non-text content. Besides including alternative text for images and image map
areas, you need to provide textual equivalents for audio and more generally for multimedia
content.

Some Definitions
A transcript of audio content is a word-for-word textual representation of the audio,
including descriptions of non-text sounds like "laughter" or "thunder." Transcripts of audio
content are valuable not only for persons with disabilities but in addition, they permit searching
and indexing of that content which is not possible with just the audio. "Not possible" is, of
course too strong. Search engines could, if they wanted, employ voice recognition to audio files,
and index that information - but they don't.
When a transcript of the audio part of an audio-visual (multimedia) presentation is
displayed synchronously with the audio-visual presentation, it is called captioning. When
speaking of TV captioning, open captions are those in which the text is always present on the
screen and closed captions are those viewers can choose to display or not.
Descriptive video or described video intersperses explanations of important video with
the normal audio of a multimedia presentation. These descriptions are also called audio
descriptions.
Wave Pad Audio Editing Software:
Professional sound editing software for PC & Mac
This audio editing software is a full featured professional audio and music editor for Windows
and Mac OS X. It lets you record and edit music, voice and other audio recordings. When editing
audio files you can cut, copy and paste parts of recordings then add effects like echo,
amplification and noise reduction. WavePad works as a wav or mp3 editor but it also supports a
number of other file formats including vox, gsm, wma, real audio, au, aif, flac, ogg and more.
Typical Audio Editing Applications
Software audio editing for studios and professional journalists.
Edit sound files to broadcast over the internet with the BroadWave Streaming Audio Server
Normalizing the level of audio files during mastering before burning to CD.
Editing mp3 files for your iPod, PSP or other portable device.
As a music editor (includes ringtones creator formats).
Music editing and recording to produce mp3 files.
Voice editing for multimedia productions (use with our Video Editor).
Restoration of audio files including removing excess noise such as hiss and hums.
System Requirements
Works on Windows XP 2000/2003/Vista/2008 and Windows 7
For earlier Windows versions (98, ME)
Mac OS X 10.2 or later;
Pocket PC 2003, Smartphone 2003 (Windows CE 4), Windows Mobile 5 Pocket PC /
Smartphone,
Windows Mobile 6
To run under Linux use WINE.