Principles of Sound

Audio frequencies are those perceptible by the human ear.

All sound is produced by mechanical vibrations. When
something vibrates, it alternately compresses and expands
the air around it. These condensations and rarefactions are
transmitted to adjacent air particles. Therefore, some of
the energy of a vibrating surface is transmitted away from
it and may affect our hearing mechanism, providing the
vibrations are those we can hear.

Sound needs an elastic medium - solid, liquid, or gas - to

go through; it cannot exist in a vacuum. When a region of a
material is compressed, it exerts a tug on its neighbouring
regions, and they in turn pull on their neighbours. If the
area is expanded, it will push on its neighbours, and that
push also will go through the medium. Alternating
compressions and expansions create regions of alternating
high and low density which move throughout the material;
the result is a sound wave. As the wave goes by, each
particle in the substance oscillates back and forth.

How we graphically represent compressions and rarefractions

The pitch of the sound is determined by its frequency - the

rate at which the vibrations occur. The unit of frequency
is called a Hertz (Hz); 1 Hz is equal to 1 vibration per
second. An alternate term is "cycles per second." The human
ear is sensitive to frequencies between about 20 Hz and
20,000 Hz. Frequencies lower than 20 Hz are called
infrasonic; those higher than 20,000 Hz are called
ultrasonic.
The magnitude of the changes in a material associated with
a sound wave are remarkably small. For example, a tone of
middle C (261 Hz) played in air as loudly as the ear can
tolerate, results in density changes of less than one-tenth
of a millimetre (0.004 inches).

The wavelength of sound is the distance between two

adjacent compressions. For example, for a tone of middle C
in air, the wavelength is about 1.2 metres (4 feet). The
frequency, the wavelength, and the speed of sound have this
relationship:

speed = frequency times wavelength

The shorter the wavelength, the higher the frequency (more

cycles per second)

The most important, and technically the most impressive, of

sound receivers is the ear - it can detect sound density
changes of less than one ten-millionth of 1 percent. This
figure corresponds to a particle displacement of less
than .0000001 millimetres. Now that's small.
Because the range of acoustic amplitudes that the ear can
detect is so large, a special compressed scale has been
devised to describe acoustic intensities. In this "decibel"
scale a 10-decibel (dB) difference between two sounds is
perceived as a loudness difference of a factor of two.
Regular exposure to sounds of about 90 dB will eventually
cause hearing loss, and sounds of 130 dB can cause
immediate and permanent hearing loss.

Sources of Sound - Microphones

A microphone is a device that transforms sound into

electrical signals. The signals can then be transmitted,
amplified, recorded electrically and, finally, converted
back into sound.

All microphones contain a thin membrane - either a

diaphragm or a ribbon. The membrane vibrates in response to
the sound striking it. A transducer (the term is taken from
the Latin "lead across") connected to it generates an
electrical signal that is equivalent to the vibrations in
frequency and amplitude.

Types of Operation
Microphones may be classed by operating principle. The main
types are carbon, piezoelectric, dynamic and capacitor.

The carbon microphone used to be the basic part of a

telephone transmitter - it is not normally used for
broadcast. It consists of a cup containing loosely packed
carbon granules next to a thin metal diaphragm. When sound
waves strike the diaphragm, they cause a change in the
electrical resistance of the carbon pack that is
proportional to the change in pressure. As the carbon
microphone doesn't actually generate any electricity of its
own, an external battery is required. Also, the carbon
particles have a tendency to pack over time, which requires
an occasional tap to loosen them. Although we no longer use
carbon microphones in this business, it's a tradition that
continues to this day - tapping the microphone to see if
it's working!

Carbon microphone

In the piezoelectric, or crystal, microphone, the diaphragm

is linked to a transducer crystal (Rochelle salts or barium
titanate are commonly used). The distortion of the crystal
by the sound waves generates a small electrical signal.

Piezoelectric microphone
The dynamic microphone has the diaphragm attached to a coil
of wire that moves within the magnetic field of a permanent
magnet. The coil generates the signal voltage by
electromagnetic induction. When wire is wound in a coil,
electricity can be generated by moving a magnet within the
coil. This is because the lines of magnetic force created
by the magnet are powerful enough to jiggle electrons free
within the wire when they cut across the magnetic field at
right angles. The electricity is only generated while the
magnet is moving. It's the change that does it. Therefore,
in our microphone, the moving coil (attached to the
diaphragm), makes small electrical currents that we can use.

Dynamic microphone
To explain the capacitor (or condenser or electrostatic)
microphone, we must understand what a condenser is.

Take a parallel pair of electrodes, separated with air or

some other non-conductor - this is a condenser. Apply an
electric current to them. Electrons will rush from the more
negative one to the more positive one, in an attempt to
balance the electrical potential between them. If you take
the current away, the plates will temporarily have a charge
on them. You can now use that stored charge for other
purposes. The word condenser was thought up by the first
experimenters who thought that collecting electricity was
"condensing" it. To make it more confusing, we now call
condensers "capacitors."

Electrostatic (or condenser) microphone

Okay, then. The condenser microphone contains a thin metal

diaphragm that is stretched close and parallel to a fixed
metal electrode; this is our condenser. We then add a
charge to this condenser. The sound waves generate
vibrations in the one flexible diaphragm plate, which will
cause a small change in the otherwise fixed voltage
available to us. Since the diaphragm's vibration results in
a variation of our condenser's ability to store a charge,
we can amplify and use that change.

The electret condenser uses a permanently polarized

capacitor, so you don't need a power supply to charge the
plates. Electret microphones are still hooked up to a
battery, however, because the signals they generate are so
weak that it they have to be amplified before being sent
down the microphone's connecting wires.

Loudspeakers

A speaker is a device that converts electrical energy into

mechanical energy. This in turn, pushes air, producing a
sound.

There are different ways to make a loudspeaker, but one

particular type is the most popular because of its
efficiency and wide-range response. It is known as a
dynamic, or magnetic loudspeaker.
In the dynamic loudspeaker a permanent magnet (or
electromagnet) produces a magnetic field. A coil of wire
located within the magnetic field is energized by an
electrical signal representing the sound to be reproduced.
Since a variable magnetic field is generated in the coil,
the coil is alternately attracted to and repelled from the
magnet. The coil is glued to a cone of paper, which
vibrates and causes the surrounding air to vibrate, re-
creating the original sound. By the way, this is the
reverse principle of the electromagnetic theory used in the
dynamic microphone, discussed a few paragraphs back.

Dynamic loudspeaker (courtesy How Stuff Works)

Sound Recording

Sound recording involves the electronic detection of sounds

and their preservation in analog or digital coding in a
storage medium - usually a disc, tape, or film. In
reproduction, or playback, the encoded information is
recovered from the storage medium, amplified, and fed to
loudspeakers or headphones that re-create the original
sound.

The Storage Medium

Sound recording technology can be classified into five

storage media:

Mechanical Recording

This technology is the basis of all phonograph records. The

audio signal is represented by an undulating groove in the
surface of a cylinder or disc. For playback, the record
spins on a turntable while a lightweight stylus traces the
pattern of wiggles in the groove.

Magnetic Recording

The basis of all tape recording (both audio and video),

magnetic techniques are also used for the sound tracks of
some 70-mm motion-picture films. In all cases, a plastic
tape or film carries a thin coating of magnetic material,
usually a metal oxide, on which a varying magnetic pattern
is imposed during recording.

Optical Film

This technology is used for the sound tracks of all but a

few motion-picture films. During recording a "light-valve"
varies the amount of light passing through a lens system
reaching the sensitized film - the strength of the light
changes with the sound. In playback the developed film
transmits a varying amount of light to a photocell, which
creates a varying electrical signal.

Optical Disc

This is the basis of the compact disc (CD), digital video

disc (DVD), and the optical videodisc. The signal is
represented by a pattern of microscopic pits along a
reflective spiral track in the disc. In playback, the
pattern is "read" by a small laser and photocell.

Solid State

With the proliferation of flash memory media such as

Compact Flash, Smart Disk, Memory Stick and other devices,
we now must include this type of recording medium along
with the other, more traditional, types.

The Form of Signal

Sound recordings are also classified according to the form

of the signal that is stored:

Analog (Amplitude Modulation or AM)

All sound is characterized by a pattern of rapidly varying

air pressure. In analog recording, that pattern is imposed
directly upon the storage medium, as the undulating groove
in a phonograph record, the varying magnetic pattern in an
analog recorded tape, or the varying light pattern of a
film sound track. The principal drawback of analog
recording is that imperfections in the storage medium (for
example, the particles of dust in a record groove or on a
film sound track) can become part of the audio signal
during playback.

Frequency Modulation (FM)

Used for recording the sound and picture in videodiscs and

"HiFi" videocassettes, the sound-wave pattern is
represented by slight variations in a recorded base
frequency or carrier signal whose average frequency is
above 1 megahertz (MHz). This approach requires more
complex circuitry, but it avoids the limitations of direct
analog recording.

Digital

The sound is represented by a binary (two-state) code in

which the recorded signal alternates between "on" and
"off." Of several possible coding schemes, the most
commonly used is "pulse code modulation" (PCM). Error
correction codes are included in the recording, allowing
near-perfect re-creation of the original audio signal
during playback.

These recording methods may be used to record audio or

video. In principle, each of the forms (AM, FM, and
digital) may be used with any of the four storage
technologies, yielding up to twelve possible combinations.
For example, mechanical-type disc sound storage with stylus
playback has been used for analog recording (the familiar
phonograph record); for FM recording (the CED videodisc
system that was briefly marketed in the United States by
RCA); and for digital recording (the Teldec Mini-Disk
system, that was proposed as an alternative to the compact
disc).

Popular Audio Recording Formats

For a description of phonograph records, cassettes, audio

head configurations and tracks, CDs, and DAT (digital audio
tape), please refer to the Appendix.
Analog Recording and Playback on Magnetic Tape

In recent decades magnetic recording has become the most

popular of all recording technologies, mainly because of
the ease with which magnetic signals can be recorded,
edited, copied, erased, or rerecorded. When an electric
current flows in a coil of wire, it generates a magnetic
field. Conversely, when a magnetic field moves near a wire,
it generates an electric current in the wire. These
physical phenomena are the basis of audio tape recording
(conversion of an electrical signal to a magnetic pattern)
and playback (conversion of the magnetic pattern back to an
electrical signal).

A tape recorder consists of two systems: a tape transport

mechanism that moves the tape past the heads at a uniform
speed; and record and playback electronics that prepare the
signal for recording and then amplify it in playback.

Magnetic recording and reproducing

The actual recording or playback is done by a "head," a
small electromagnet mounted in a shielded case. For
recording, the signal current generates a magnetic field in
the head that is imposed on the magnetic particles in the
tape. For playback, the magnetic fields in the moving tape
generate tiny electric currents in the head. At each
instant the head is in magnetic contact with a very small
area of tape (in an audio cassette, that area is about
0.0001 inch wide by 0.02 inch high) containing several
thousand particles. Some recorders have separate heads for
recording and playback.

Each microscopic particle of iron oxide is an individual

bar magnet oriented lengthwise on the tape, with a north
pole and a south pole. In an unrecorded tape, about half of
the particles are magnetized with their north pole forward,
and the other half are oriented with their south pole
leading. When exposed to a magnetic field strong enough to
overcome their "coercivity" (resistance to change), the
particles adopt the imposed field direction, reversing
poles if necessary. Thus, the process of recording is
simply one of flipping each particle's magnetic orientation
one way or the other. Once set, the particles keep their
magnetic orientation until exposed to another strong field.
(For analog recording, the audio signal is combined with a
strong "AC bias" signal that alternates from north to south
and back around 100 KHz.) The result is that the audio
waveform is faithfully represented by the percentage of
north forward particles at each location along the recorded
track.

To erase a recording, an "erase head" exposes the tape to a

version of the bias signal, whose rapid polarity reversals
leave half of the particles magnetized in each direction.

Audio Consoles

Let's now turn to the nerve centre of the entire operation

- the console. It's here that signals flowing from mics,
turntables, tape recorders and players are amplified,
mixed, balanced and routed to air or tape.

At first glance, consoles may look threatening. They're

really not - and they all operate on quite basic
principles. They can be considered to have three basic
systems: input, output and monitoring. As we go through a
typical audio board, keep in mind three points: we have to
get the signals from each sound source to inputs on the
console; combine them; and monitor what we're doing.

We'll look at a single channel of the console, or "strip."

Input Selection

On most consoles, there is an ability to select from two

different inputs on a given audio "strip". Unfortunately,
you only get to choose one of these sources at a time to
work with.

Level Setting

If it's a microphone you've chosen, its low level will have

to be amplified to what we call "line level", so each strip
contains a microphone preamplifier, which can be switched
in and out of the line.

Equalizers

An equalizer is a device that alters a sound by boosting or

reducing selected frequencies. We usually have one of these
on every fader in our console (except for the submasters
and masters). More on these later on in this chapter.

More Level Setting

We have managed to get all inputs up to what is called

"line level," but we now need a way to vary the levels of
the different sources. Enter the fader...or
potentiometer...or pot...or attenuator...or gain
control...or, if you insist, volume control. In principle,
it's not much more sophisticated than a knob or sliding
fader, like on your stereo system, except that it has
calibrations on it.

Auxiliary Sends

These are rather like submasters, with two exceptions.

First, you can change the level going to the auxiliary

output from each strip's source, regardless of the position
of the fader for that strip. You also can decide whether to
send the strip's source straight away, before you've used
the fader to adjust its level, or after the fader. This
selection is called "pre-fade" or "post-fade."
Second, auxiliaries don't normally go to air - they're used
for operations like sending sources to control room or
studio floor monitors, sending special material to the
earpieces that on-camera talent are wearing, and sending
information to telephone callers for phone-in shows.
A typical audio console (featuring one strip)

Solo and Pre-Fade Listen

This type of button allows you to listen to a single

channel off-line, sometimes through a special "cue" speaker
and amplifier, and sometimes through the main monitor
speakers, overriding any existing program material on those
monitors.

On-Off or Mute

At times, we'll want to electrically separate the sound

source from the board. So, we'll usually find a channel on-
off key on each strip.

Professional audio operators do not use the strip's mute

button to select sources while they are on air! All sources
should be mixed using the main faders. This means that
words and music will not accidentally be clipped by slow-
moving operator's hands, or by unexpected cues. As well, in
the heat of the moment, it is much easier to see if an
audio strip is activated by looking at the larger faders on
the board, instead of having to search for a tiny mute
button, or a mute indicator light.

Phase Button

Normally, stereo audio signals are mixed together and laid

down on audio tracks "in phase." Electronic transmission of
sound is a series of voltage level waves. If a sound
appears on two channels, and those channels are mixed , the
sounds should "add up." If, for some reason, one channel is
"out of phase" with respect to the other, the sounds cancel.

How could one channel ever get "out of phase" with the
other in the first place? Consider that broadcast audio is
a balanced system; two wires carry the signal, and a third
is a ground connection. If somewhere in all the audio
interconnections, those two signal wires are reversed in
one of the channels, that signal will now be out of phase
relative to the other channel. The obvious way to correct
an electrical out of phase condition is to rewire the
equipment properly. If there is no time for that, most
consoles have a phase reversal switch that shifts one
channel's phase by 180 degrees, back into phase.

Pan Pots

These allow us to shift the balance of sound to any point

between the left and right channels of our stereo consoles.

Tying It All Together

We can make this "tying together" operation easy (but less

flexible) or more difficult (but more versatile). If we
bring all outputs to one final fader, that one becomes the
"master" fader, controlling all of our sources to air. We
also can group our sources together in "bunches," each
bunch going to a separate "submaster" fader, which in turn,
is bundled with the other submasters into the master. This
submastering route allows us to vary sound elements in
groups - for example, all the microphones, all of the VTR
and line sources, "the band," or any other combination
you'd like.

Keeping An Eye On Things

If our signals are too strong, they will overload

equipment. If they are too weak, they will not overpower
the background noise element in our electronics. We need an
indicator.

There are many different types of audio level meters, but

the most common one is the VU meter (VU stands for volume
unit). The VU meter is an "averaging" meter in that it
doesn't respond to sudden peaks in level - sort of like
your ears. You can get VU meters in the standard, needle
movement form, or as an LED or LCD display. They are
calibrated in a scale that ranges typically from -20 VU to
+ 3 VU.
VU meter

PPM meter

Another meter commonly used in Europe (and increasingly, in

North America) is the PPM (peak program meter). This device
reads peaks in level rather than averages. The argument
goes that although humans may not hear momentary peaks in
loudness, the equipment does. Therefore, the PPM is better
insurance against signal distortion.

LED peak reading meter

When mixing audio, "ride the gain" so the level stays

between -5 and 0 VU; ride a PPM meter at +8 dBm. It's
normal for peaks and dips to go beyond this, of course. Mix
with a light, fluid hand. Pots should not be jerked up and
down to make adjustments at the slightest fall or rise in
loudness. Changes should be smooth - you'll hear abrupt
ones. Once again, this is another good reason to use the
faders to activate channels, instead of the mute buttons.

Don't We Get To Hear It?

This part's easy. Take a tap off the master (or, using a
selector switch, any of the submasters), and amplify it
enough to drive loudspeakers. Perhaps put a level pot in
the circuit so we can turn it down (or up!) when we want
to. That should do the job. One precaution to be taken,
however: many such circuits incorporate "mute" systems so
when certain microphones' faders are actuated (or when an
external mute switch is pressed), the audio at the speakers
is automatically cut off. This prevents the dreaded
feedback.

An Additional Extra - The Tone Oscillator

This is a signal generator that produces pure sine wave

tones at selected frequencies. It is used to calibrate the
console to connected tape recorders so their VU meters
indicate the same levels, and to put reference tone levels
on tape recordings.

A Bit More About Equalizers

Perhaps the most often used signal processor is the

equalizer. Your home stereo system's bass and treble
controls are primitive equalizers. There are three types in
general use in broadcast - selectively variable (fixed
frequency), continuously variable (parametric), and
graphic. Their jobs are all the same: boost or cut the
level of a selected frequency, or set of frequencies.

High Pass, Low Pass, Band Pass Filters

A high pass filter allows you to select a frequency below

which signals will be removed, thus passing higher
frequencies.

A low pass filter, as you might expect, allows you to

select a frequency above which signals will be removed,
thus passing lower frequencies.

It is disconcerting for newcomers to the audio operating

scene to find that high pass filter controls have figures
on them corresponding to low frequencies, and low pass
filters having numbers relating to high frequencies!

A band pass filter allows you to select two frequencies -

one below and one above which signals will be removed, thus
leaving you with a band of frequencies that will be sent
onwards.

Selectively Variable Equalizer

This unit is a series of band pass filters, and provides

two or three ranges (or bands) of the frequency spectrum
that you can vary. You can change where in each range (low,
mid, high) you want to affect, and by how much boost or cut.

Selectively variable equalizer, featuring low cut, high

pass, band pass, and low pass filters

Parametric Equalizer

This unit is similar to the selectively variable equalizer

mentioned above, but you also can select how narrow (or
wide) you want the band affected to be. This is known as
affecting its "Q."

Graphic Equalizer
These are nifty looking units that allow you to "draw" on
the front panel, by using a series of sliders, the
frequency response you wish to achieve. The more sliders
you have, the more delicately you can control your output -
but the more expensive the unit becomes.

Graphic equalizer