Sound Manual

CONTENT.
Chapter 1. Physical Acoustics.

Introduction. Sound. Frequency. Amplitude. Speed of sound. Wavelength.
Period. Phase. Harmonics. Sound Pressure Levels. The Decibel. The
Decibel related to electrical units. Relationship between dB and acoustic
signal. RMS. Volume. Level. Revenue. The ear.
Episode 2. Psychoacoustics.
Introduction. Psychoacoustic Sensations. Height. Sonority. Doorbell.
Sound Directionality. Masking.
Chapter 3. Architectural Acoustics.

Introduction. Echoes. Early Reflections. atmosphere. Sound Absorption.
Reverberation Time. Acoustic Absorbing Materials. Acoustic Isolation.
Chapter 4. Signals and Systems.

Introduction. Signs. Systems. Block Diagrams. Noise. Dynamic range.
Distortion. Frequency Response. Signal Processing.
Chapter 5. Electricity.
Electrical circuits. Electric current. Strain. Ideal Voltage Source.
Endurance. Electrical Power. Impedance.
Chapter 6. Microphones.
Introduction. Directionality. Dynamic Microphones. Capacitive
Microphones. Impedance. Noise. Distortion. Phantom Fountain.
Characteristics of some Microphones. Tips and some Techniques for
Microphones. Stereo Microphone Techniques. Shure Product Selection.
Chapter 7. Amplifiers.
Introduction. Revenue. Signal Levels. Classification of Amplifiers.
Maximum Output Power. Sensitivity. Signal/noise ratio. Frequency
Response. Distortion. Input Impedance. Damping Factor. Channel
Separation. Other features.
Chapter 8. Speakers and Acoustic Boxes.

Introduction. Classification by Frequency Ranges. Moving Coil Speakers.
Compression Exciters. Acoustic Boxes. Sensitivity. Frequency Response.
Directionality.
Chapter 9. Frequency Processors.
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Introduction. Filters. Quality Factor. Stepped Equalizers. Graphic
Equalizers. Parametric Equalizers. Semiparametric Equalizers. Frequency
Spectrum.
Chapter 10. Dynamic Processors.

Introduction. Compressor-Limiter. Noise Gates. Expanders. Threshold.
Ratio. Attack Time. Recovery time. Revenue.
Chapter 11. Mixing Consoles.

Introduction. Specific Functions of a Console. Structure of a Mixing
Console. Equalizer. Insertion Connections. Auxiliary Connections.
Auxiliary Shipments. Auxiliary Returns. Groups or Submasters. Phantom
Fountain. Monitoring. Mute selectors. Solo Selectors. VU meters.
Presentation of the Entry Channels. Departure Section. Connections for
Tape Recorder. Consoles Specifications. Distortion. Noise. Overload
Margin. Frequency Response. Indicators. Other Specifications. Conclusion.
Chapter 12. Monitors.

Brief relate.
Chapter 13. Cables and Connectors.

Introduction. Electrostatic and Electromagnetic Isolation. Principles of
Balanced Circuits. Cables with Maya of 1 and 2 conductors. Cables without
Maya and Speaker Cables. Multipair Cables (Snakes). Connectors.
Telephone Plug. Phono connector (RCA). XLR connector. Banana
connector. Speakon EP4 and EP8 connector. Direct Boxes.
Glossary.
Bibliography.
2
INTRODUCTION.
The purpose of this manual is to better prepare one of the links in the
chain: The Engineer ; that for different reasons in the Spanish-speaking
world tends to not have the necessary theoretical and practical preparation
to carry out its function properly, which is why we have a countless
number of empirical Pseudo-Engineers who do their best to carry out a job
in those who are involved by true vocation.
In the sound process, an attempt is made to magnify or amplify the
performance of the Praise group or the Preacher before the public, so we
engineers and technicians must implement all our capacity and experience
in the use of the equipment necessary to achieve this goal. so that the public
can receive it as it is and as it is generated on stage, without distorting it,
from any point of view.
The transducer is the instrument that converts one type of energy into
another; For example: a guitar is a transducer, which converts the vibration
generated by a musician on its strings, amplifying it through a wooden
body and converting them into sound waves so that others can perceive
them.
The microphone is another transducer, which perceives these sound
waves and converts them into voltage, which is transmitted to other
equipment to then process it and finally record it on magnetic tape or
amplifier through a speaker.
The magnetic tape recorder is also a transducer, because it converts
the electrical energy generated by the microphone into magnetic energy, so
that it can be stored on a tape.
The speaker is another transducer, which converts electrical energy
into sound waves.
The ear is even a transducer that converts sound waves into brain
waves.
As you can see, in the sound process there are a large number of
transducers, and in one way or another, engineers and technicians are at the
mercy of their quality to be able to do our work properly.
I personally believe that that is the magic of the whole thing: “Make
them work according to your own criteria and not according to the device's
own criteria.”
3
NATURE OF SOUND.
When you speak or play an instrument, a

compression-rarefaction of air particles.
This phenomenon is known as sound pressure waves.
Sound reaches our ears, through the air, in the form of these waves
that are transformed into electrical impulses and interpreted by the brain as
speech, music or noise.
The compression-rarefaction that occurs in air is similar to what
occurs in water when we throw a stone; the waves that are generated have a
high part or compression and a low part or rarefaction.
The characteristics of a sound pressure wave are the following:
Frequency, Amplitude, Wavelength, Period, Speed, Timbre, Acoustic
Envelope.
FREQUENCY.
Sound pressure waves are cyclical and we can represent them

graphically using a circle or cycle (0  to 360  )
If we follow the trajectory of the circle at a constant speed, setting the

interval to 1 second, we have 1 cycle per second. This unit is called HERT
and measures the frequency of a wave.
If we double the speed we have 2 cycles in one second or the
frequency of 2 hertz and so on.
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AMPLITUDE.
Apart from the frequency with which a wave repeats, it has a

compression-rarefaction force and is called amplitude. Its unit is the decibel
of sound pressure.
If we start from a zero equilibrium point, the upward force from zero
is compression and has a positive value, the downward force is rarefaction
and has a negative value.
Pressure waves propagate longitudinally, if we return to the example
of the circle and unfold it longitudinally we obtain the graphic
representation of a wave called Sinusoidal.
We then have the positive amplitude, the negative amplitude and the peak-
to-peak amplitude.
Frequency and amplitude are two characteristics that are related in the
following way:
1 and 2 same frequency different amplitude.

3 and 4 same frequency different amplitude.
1 and 3 same amplitude different frequency.
2 and 4 same amplitude different frequency.
SPEED OF SOUND.
5
The speed of sound in air is:
331.4 m/s 0  C + 0.6 m/s * 1  C
This means that every time the temperature increases by 1  C we

must add 0.6 m/s and when it decreases we will subtract 0.6 m/s.
For example, if we want to know the speed of sound at 21  C:
21  C * 0.6 m/s = 12.6 m/s

12.6 m/s + 331.4 m/s = 344 m/s
The speed of sound in feet/sec is 1130 peis/sec.
WAVELENGTH.
It is the distance that a cycle of any frequency needs to develop

completely.
We can know the wavelength of any frequency using the following
formula:
 = V/F
 = Wavelength.
V = Speed of Sound.
F = Frequency.
For example, if we want to know the wavelength of 300 hertz at 21 

C:
 = 344 m/s / 300 Hz = 1.14 m.
This means that when a 300 Hz sine wave is generated. 1.14 meters
are required to develop a complete cycle.
PERIOD.
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It is the time it takes to travel the distance of one cycle (  ) of any
frequency and we can determine it using the following formula:
T = Period in Seconds.
T = 1 / FF = Frequency.
1 = Constant.
For example, if we want to know the period (T) of 300 Hz:
T = 1 / 300 Hz = 0.00333 seconds.
This means that at 21  C 1.14 m. They are traveled in 0.00333

seconds for a 300 Hz wave.
PHASE.
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If 2 sine waves of the same frequency are generated in the same
direction regardless of their amplitude, we have two possibilities:
1. Let them start at exactly the same time. Then all their degrees
coincide and you have two waves in PHASE. Its amplitude is added
(reinforced).
2. Let them not start the same. Then their degrees do not match and we
have two waves OUT OF PHASE. Its amplitude is cancelled.
For it to exist out of phase, only 1  phase shift is required; in this

case there will be a cancellation proportional to the phase shift.
In the event that there is a 180  phase shift and the amplitude is the
same, there will be a total cancellation.
This phase shift can be measured in degrees, time or distance.
HARMONICS.
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A harmonic is a frequency multiple of a fundamental. The
fundamental is the base frequency of a note emitted by any instrument.
Harmonics have a different amplitude and frequency relationship with
respect to their fundamental, depending on the instrument that generates
them.
When a piano and a saxophone generate the same note, what you are
hearing is the same fundamental but with different harmonics. What makes
them sound different is the relationship with respect to the fundamental and
is known as timbre.
An instrument is represented by a complex waveform that is the result
of the sum of the fundamental and harmonics.
Using filters, a complex wave can be decomposed until obtaining a
series of sine waves that would be its fundamental and harmonics.
We can reverse the process and add a series of sine waves to a Sine
Wave until we obtain a tone similar to “x” instrument, this if we know the
number of harmonics and the amplitude relationship they have with respect
to the fundamental.
SOUND PRESSURE LEVELS.
The ear operates in a 10 (13): 1 energy range which is an extremely

wide range, and so a logarithmic scale is implemented to compress the
measurements into more workable figures.
The system used to measure sound pressure levels (SPL) and their
signal variations is the decibel (dB).
The sound pressure level is nothing more than the atmospheric
pressure generated by the vibration of a sound, and measured at a certain
point. This measurement is usually made in dB SPL and we can represent it
graphically as follows:
dB SPL = 20 Log P / P ref.
Where:
P is the Sound pressure measured in dynas / cm (2)
P ref. is the reference sound pressure (0.0002 dynas / cm 2) (the
threshold of hearing)
The electrical signal level, also measured in dB, corresponds to 10

times the logarithm of the average of two power levels:
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dB = 10 Log P/ P ref.
Where:
P is the power in Watts.
P ref. is the reference power in Watts.
Electrical signal levels in decibels can be expressed in several ways:
dBm: decibels with a reference of 1 milliwatt.

dbu or dBv: decibels with a reference of 0.775 volts.
dBV: decibels with a reference of 1 volt.
THE DECIBEL (dB).
The decibel (dB) always describes a relationship between two

quantities; These quantities are related to power. The reason the decibel is
used is because it is logarithmic, which allows us to summarize small
values that would otherwise require multi-digit numbers. Also, since ear
sensitivity is logarithmic, decibel values better express the auditory
response than integers and simple power ratios. Therefore we conclude that
the decibel is integrated into professional Audio to simplify things, and not
to complicate them.
The decibel represents 1/10 of a Bel, and a Bel is the logarithm of the
ratio between two units of electrical or acoustic power. To express the
relationship between two power values (P1 and P2) in Bels we use the
following formula:
Bel = Log(P1/P2)
In the case of the decibel (tenth of a Bel), the resulting formula is:
dB = 10 Log (P2 / P1).
Example 1:
10
What is the power ratio in dB between 2 watts and 1 watt?
dB = 10 Log ( 2 / 1)
= 10 Log 2
= 10 * 0.301
= 3.01
=3
Example 2:
What is the relationship in dB between 100 watts and 10 watts?
dB = 10 Log ( 100 / 10)

= 10 Log 10
= 10 * 1
=10
The previous examples allow us to realize two interesting aspects in

relation to dB:
a) Whenever one power is twice as high as another, it represents

only 3 dB of difference.
b) Whenever one power is 10 times greater than another, it only

represents a 10 dB difference.
The decibel can also be used to calculate the relationship between

voltages, and in this case the formula to use is:
DB volt = 20 Log (E2/E1)
According to the results obtained using this formula, the relationship

in decibels between two voltages that have twice the value of each other is
6 dB (which is double that obtained in the power formula). Similarly, a
voltage that is 10 times greater than another (10:1 ratio) produces a result
of 20 dB. This discrepancy is explained using the following development:
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The ratio in dB between 100 and 10 watts is 10 dB.
The dB ratio between 100 and 10 volts is 20 dB.
If we calculate the power generated by 100 volts in an 8 ohm resistor

using the following formula:
P = E(2) / R
= 100(2) / 8
= 10000 / 8
= 1250 volts
If we calculate the power generated by 10 watts in an 8 ohm resistor:
P = 10 (2) / 8
= 100 / 8
= 12.5 watts
Now if we calculate the relationship of both powers in dB:
DB = 10 Log(P2/P1)
= 10 Log (1250 / 12.5)
= 10 Log 100
= 10 * 2
= 20 dB.
Below we present a list of values in dB considering 1 watt as a

reference:
Power in Level in dB
Watts (P2) (P1 = 1 watt)
1 0
1.25 1
12
1.6 2
2.0 3
2.5 4
3.15 5
4.0 6
5.0 7
6.3 8
8.0 9
10 10
100 20
200 23
400 26
800 29
1000 30
2000 33
4000 36
8000 39
10000 40
20000 43
40000 46
80000 49
100000 50
THE dB RELATED TO ELECTRICAL UNITS.
dBm.
The dBm is related to electrical units with 1 milliwatt as a reference.

So 0 dBm = 1 milli watt. This unit became standard in 1940 for use in
telephone lines where the resistance is 600 ohms.
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This value (0.001 watts) corresponds to the power dissipated when
0.775 volts rms is inserted into a line with 600 ohms of resistance.
If we are provided with the information that a console has a
maximum output of +20 dBm, this means that the maximum output of this
console is 100 milliwatts.
If we are told that the maximum output of a console is +20dBm into
600 ohms, this means that the maximum output load of the console is 7075
volts rms. We can infer this information using the following formula:
If: P = E(2) / R
Then: E =  P * R
E =  0.1 * 600
E =  60
E = 7.7459
E = 7.75
Now you may wonder where the value 0.1 for P comes from:
Yes: dBm = 10 Log (P2 / P1)

Then: P2 = 10  (dBm / 10) + Log P1 
P2 = 10  (20 /10) + Log 0.001 
P2 = 10  2 + (-3) 
P2 = 10 (-1)
P2 = 0.1
This unit is used for connections that require long lengths of cable,
which is generally balanced (600 ohms) as it is not susceptible to noise
caused by induction and loss of high frequencies.
dBu.
Modern Audio equipment is sensitive to voltage levels, in this

equipment its output power is not relevant, or it is only relevant in the case
of power amplifiers when they drive speakers. In that case, output watts are
the unit used instead of dB.
The dBu represents the same value as the dBm, but only if the
impedance is 600 ohms, but 0 dBu is always equivalent to 0.775 volts. This
unit was created to avoid confusion with dBV.
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The “u” after the term “dB” represents the word “unloaded”, which
means “without load”, to express voltage values in an open circuit, where
the impedance factor does not intervene.
dBV.
This is a voltage related unit, which has a reference value of 1 volt

rms. The resulting formula is:
dBV = 20 Log E2 / E1
Knowing that E1 = 1 volt.

This unit is generally used in unbalanced cables, which are generally
high impedance.
RELATIONSHIP OF dBV, dBu, dBm WITH THE

SPECIFICATIONS.
In most equipment, we will find that the inputs and outputs that use ¼
and RCA connectors have specifications given in dBV (usually –10 dBV),
and this is justified because this type of connector is used for high
impedance equipment, the which are more sensitive to voltage than power.
On the other hand, in equipment with long cable runs or environments
where protection against induced noise or leakage is important (recording
studios or live sound systems).
In some cases, we want to connect low impedance outputs to high

impedance inputs, which generally causes greater level losses, but in the
opposite case (high outputs to low inputs), the output can saturate (when
trying to compensate impedance), which would generate distortion, loss of
level, significant damage to the frequency response, and in the worst case,
even damage to the circuits. So check the specifications carefully before
making an irreversible mistake.
dBW.
It is a measure of power in watts (Watts) created by Mix magazine,

which has 1 watt as a reference power. It is implemented to reduce values
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when commenting output specifications of power amplifiers, where an
amplifier with 100 watts output represents (10Log100 / 1) 20 dBW.
A 1000 Watt amplifier = 30 dBW, etc. This unit is not officially
accepted by any organization, but is increasingly used in technical journals.
RELATIONSHIP OF dB WITH ACOUSTIC SIGNAL.
dB can be used to measure sound pressure level, which represents the

force of air pressing on the eardrum. In this case the equation used is:
DB spl = 20 Log P2 / P1
If we experiment a little with this equation, we find that if one SPL is

twice as high as another, it only represents a 6 dB difference, and if it is 10
times greater, it represents 20 dB.
Volume is a subjective value assigned to the auditory stimulus, and is
highly influenced by the frequency and absolute level of the sound.
Generally 0 dBspl represents the threshold of hearing (in a young,
undamaged ear.
Instead of only relating several SPL's to various pressures, it is better
to relate the SPL's to usual sound sources. Below we present a very useful
table for this:
dB SPL Usual situations

0 Hearing threshold (1 to 4 kHz in young people)
10 Anechoic chamber, wind noise on leaves
20 Extremely quiet recording studio, murmur
30 Silent recording studio
40 Silent auditorium
50 Average noise in suburban housing (at night)
60 Average conversation (3 feet)
70 Home monitoring average (popular music)
80 cabin of a passenger plane
90 Heavy traffic noise (5 feet)
100 Classical music in strong sections
110 Typical film studio monitoring volume (20 feet)
120 Typical audio studio monitoring volume (rock music)
130 50 HP siren (100 feet)
140 Colt 45 pistols (25 feet)
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WHAT IS RMS?
It is the abbreviation of “Root Mean Square”. It is a mathematical

expression used in Audio to describe signal levels. It is particularly
effective for describing the energy of a complex wave or a Sine wave. It
does not represent the peak value, nor the average value of peaks, but is
obtained by quadratizing all the instantaneous voltages of a waveform,
averaging the squared values and taking the square root of those numbers.
For a periodic and constant signal (such as a Sine wave) this data can
be obtained by multiplying the peak value by a constant (0.707). So, the
RMS value of a Sine wave is 0.707 times the peak value.
This value is implemented because it reflects well the performance of
an amplifier. When the terms “program levels” or “music levels” are used,
these represent very subjective values, since much depends on the nature of
that program or musical source. The RMS value of any program will much
better reflect its energy content. That is why it is better to use the RMS
value.
In electrical measurements, RMS data is also used. These are similar

but not the same as the averages.
When we want to evaluate how the value of a signal (loudness) is
perceived by the human ear, RMS values more closely represent the
sensitivity of the ear with respect to Audio energy.
VOLUME.
It is defined as power level. In terms of Audio equipment, when you

increase the volume of a device, you increase the power output. This term
should not be used to express sound intensity or magnitude of an electrical
signal.
LEVEL.
17
It is defined as the magnitude of a quantity with respect to an
arbitrary reference. It can be expressed in dBspl (with respect to 0.0002
microbars) or in dBm (with respect to 0.0001 watts)
REVENUE.
It represents the increase in power of a signal, usually expressed in

dB. Sometimes voltage increases are expressed as voltage gain, but in this
case we must be careful, because it may represent a loss rather than a gain,
depending on the relative impedance involved.
THE EAR.
The ear is a sensitive transducer that collects sound pressure waves in

the eardrum and converts them into mechanical vibrations. These
vibrations are transferred to its internal section, which acts as an amplifier
and limiter, so that they are received by the nerve terminals, the which
convert them into a stimulus called auditory sensation.
THE THRESHOLD OF HEARING.
For most people, the hearing sensation can be realized from 0 dBspl,
which corresponds to 0.0002 microbars on a barometer.
THE THRESHOLD OF HEARING.
The average SPL level that causes discomfort in most people is called
the threshold of sensation, and this occurs at 118 dBspl between the
frequencies 200 Hz and 10 kHz.
THE THRESHOLD OF PAIN.
The average SPL level that causes pain in most people is 140 dBspl in
the 200 Hz to 10 kHz range.
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PSYCHOACOUSTIC.
INTRODUCTION.
Psychoacoustics is dedicated to studying the perception of sound, that

is, how the ear and brain process the information that comes to us in the
form of sound.
PSYCHOACOUSTIC SENSATIONS.
When we hear a sound, we perceive sensations that can be classified

into three types: pitch, loudness and timbre. Pitch is the sensation that
allows us to distinguish low sounds from high sounds, more specifically, to
differentiate the sounds of a musical scale. Loudness is the sensation by
which we distinguish a strong sound from a weak one. Timbre brings
together a series of qualities by which it is possible to distinguish the
sounds of the various instruments and voices. Frequency is related to the
sensation of height, amplitude to loudness, and spectrum to timbre.
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HEIGHT.
The relationship between frequency and pitch is quite direct, with low
frequencies corresponding to low sounds and high frequencies
corresponding to high sounds. In reality, height as a psychophysical
parameter also varies a little with the intensity of the sound, that is, a weak
sound and a strong sound of the same frequency seem to have slightly
different heights. It also varies a little with the timbre. A very bright timbre
appears to be higher pitched than a duller one, even when the frequency
and intensity are the same.
SONORITY.
The sensation of loudness, that is, the strength, volume or intensity of

a sound, is, in principle, related to its amplitude. However, the relationship
is not as direct as that between frequency and height. In fact, loudness is
dependent not only on amplitude but also on frequency. Thus, at equal
frequencies we can affirm that a sound with greater amplitude is louder.
DOORBELL.
The timbre of a sound is a complex quality, which depends on several

physical characteristics. The study of the various aspects of timbre was
highly motivated by the desire to artificially reproduce the sounds of
natural instruments, as well as to create completely new timbres, giving rise
to various sound synthesis techniques. Although today electronic
synthesizers are the most widespread and expanded, the synthesis of sounds
has several centuries of history, the flute organ can testify to the efforts of
human ingenuity in this regard.
There are two approaches to timbre analysis. The first studies isolated
sounds, the second approach classifies sounds according to the source (for
example an instrument), and associates a timbral quality with each source.
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The first approach distinguishes a low sound from a clarinet, for
example, from another high sound from the same instrument. In fact, those
who do not know the clarinet, when listening to both registers (low and
high) separately, may think that they are different instruments. Two
elements intervene here: the spectrum and the envelopes.
There is a primary envelope, which determines the way in which the
general amplitude varies in time, and a series of secondary envelopes,
which correspond to the relative temporal variations of the harmonics or
partials. The primary envelope is strongly related to the way sound is
produced, and characterizes entire families of instruments. The secondary
envelopes depend on the way in which the different frequencies of the
spectrum are damped.
In electronic sound synthesizers, efforts have been made to provide

resources to control these envelopes. At first we worked with a 4-section
primary envelope, called ADSR (acronym for Attack-Decay-Sustain-
Release). Current synthesizers allow, depending on their complexity
(generally in proportion to cost) define the envelopes more precisely.
Secondary envelopes have been implemented with a multitude of
techniques, such as the use of time-varying filters, frequency modulation,
and additive synthesis.
DIRECTIONALITY OF SOUND.
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Real sounds originate from sources that are located somewhere in the
surrounding space, giving rise to two types of sensations: directionality and
spatiality. Directionality refers to the ability to locate the direction where
sound is coming from. This sensation allows us to visually locate a sound
source after listening to it. Spatiality, on the other hand, allows us to
associate a sound with the environment in which it propagates, and
estimate the dimensions of a room or hall without having to resort to sight.
Directionality is linked to two phenomena. The first is the difference

in time between the perception of a sound with the right ear and with the
left ear, because the path of the sound wave from the source to each ear is
different. Thus, a sound coming from the left will reach the left ear sooner,
simply because it is closer to the sound source.
The other phenomenon is the difference in sound pressures, also
caused by the difference between distances, so the sound that comes from
the left, the sound pressure is greater in the left ear, not only because it is
closer to the source, but also because The head acts as a barrier to sound.
MASKING.
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Among the qualities of hearing there is one that has important
consequences for hearing, and it is the fact that sounds are capable of
masking other sounds. Masking a sound means hiding it or making it
imperceptible. Masking is a phenomenon quite familiar to everyone. It
happens, for example, when we try to listen to someone who speaks in the
middle of a very intense noise: we cannot discriminate what they are saying
because their voice is masked by the noise.
It is interesting to note that masking is a property of hearing, not

sound. In good audio equipment, if we mix a very intense sound (for
example 90 dB) with a very weak sound (for example 20 dB), the speaker
output will contain both sounds in their original proportions. This can be
verified by successively isolating, using appropriate filters, one or another
sound. However, the ear will not perceive 20 dB.
Masking is, in a certain sense, a defect of hearing, but it is also a
virtue, since it allows us to get rid of a quantity of information that is
useless or difficult for the brain to process.
Finally, masking is also used unconsciously when the volume of a
music system or Walkman is increased due to the existence of ambient
noises. In this case, by raising the sound level of the music, it masks
ambient noise, allowing you to listen to the music in better conditions. In
the modern world, ambient noise is considerable, which has led society to
prefer and get used to “loud” music. This is potentially dangerous for
hearing health, since masking noise with music requires that the music
level be between 20 and 30 dB's above the noise. Thus, if the ambient noise
is 75 dB, it is likely that the Walkman user is hearing a level close to 100
dB.
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ARCHITECTURAL ACOUSTICS.
INTRODUCTION.
Architectural acoustics studies the phenomena linked to adequate,

faithful and functional sound propagation in a venue, whether a concert hall
or a recording studio. This also involves the problem of acoustic insulation.
Rooms or halls dedicated to a particular application (for example for
music recording, for conferences or for concerts) must have qualities for
that application. By acoustic qualities of an enclosure we understand the
properties related to the behavior of sound in the enclosure, such as early
reflections, reverberation, the existence or not of echoes and resonances,
sound coverage of sources, etc.
ECHOES.
The simplest phenomenon that takes place in an environment with

sound-reflecting surfaces is the echo; it consists of a single reflection that
returns to the point where the source is located at about 100 ms. (or more)
after the sound is emitted.
EARLY REFLECTIONS.
When the sound source is surrounded by several surfaces (floors,

walls and ceiling) a listener will receive the direct sound, and also the
sound reflected from each wall. The first reflections received, which are far
apart in time, are called early reflections.
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In rooms that are not too large, the first reflections are quite close in
time to each other, so that they are not perceived as an echo.
AMBIENCE.
The distribution in time of the early reflections creates the sensation

of ambience, that is, the sensation that allows the listener to auditorily
identify the space in which they are located. Blind people develop a special
ability to interpret the spatial information contained in the environment.
Architecturally, ambiance control can be achieved through a careful
design that involves tracing, on a plan of the room, acoustic “rays” similar
to those in the previous figure, carefully measuring their paths, and from
there determining arrival times. of the corresponding reflections. Nowadays
this work is done with the help of digital computers and appropriate
programs.
SOUND ABSORPTION.
25
The surfaces of an enclosure only partially reflect the sound that hits
them, the rest is absorbed. Depending on the type of material or covering of
a wall, it may absorb more or less sound.
Hard materials, such as concrete or marble, are highly reflective and
therefore poorly absorbent of sound, while soft and porous materials, such
as glass wool, are poorly reflective and therefore highly absorbent.
Absorption increases with frequency, because for high frequencies

the wavelength is small and therefore the irregularities of the surface or the
thickness of the material itself are more comparable to the wavelength. In
some cases, however, some resonance phenomenon between the material
and the wall can improve absorption at low frequencies. Acoustic treatment
is usually almost as expensive as the construction of the building.
MATERIAL Absorption coefficient  at frequency

125 250 500 1000 2000 4000
unpainted concrete 0.01 0.01 0.02 0.02 0.02 0.04
Painted concrete 0.01 0.01 0.01 0.02 0.02 0.02
unpainted exposed brick 0.02 0.02 0.03 0.04 0.05 0.05
Painted exposed brick 0.01 0.01 0.02 0.02 0.02 0.02
Lime and sand plaster 0.04 0.05 0.06 0.08 0.04 0.06
Gypsum board (Durlock) 12mm to 10 cm 0.29 0.10 0.05 0.04 0.07 0.09
Plaster on expanded metal 0.04 0.04 0.04 0.06 0.06 0.03
Marble or tile 0.01 0.01 0.01 0.01 0.02 0.02
Wood in panels (5 cm from the wall) 0.30 0.25 0.20 0.17 0.15 0.10
Chipboard panel 0.47 0.52 0.50 0.55 0.58 0.63
Parquet 0.04 0.04 0.07 0.06 0.06 0.07
Parquet on asphalt 0.05 0.03 0.06 0.09 0.10 0.22
Parquet on slats 0.20 0.15 0.12 0.10 0.10 0.07
Rubber mat 0.5 cm 0.04 0.04 0.08 0.12 0.03 0.10
Wool carpet 1.2 kg/ m² 0.10 0.16 0.11 0.30 0.50 0.47
Wool carpet 2.3 kg/ m² 0.17 0.18 0.21 0.50 0.63 0.83
Curtain 338 g/ m² 0.03 0.04 0.11 0.17 0.24 0.35
Curtain 475 g/ m² gathered at 50% 0.07 0.31 0.49 0.75 0.70 0.60
Polyurethane foam (Fonac) 35 mm 0.11 0.14 0.36 0.82 0.90 0.97
Polyurethane foam (Sonex) 35 mm 0.06 0.20 0.45 0.71 0.95 0.89
26
Glass wool (felt 14 kg/ m³ ) 25 mm 0.15 0.25 0.40 0.50 0.65 0.70
Glass wool (felt 14 kg/ m³ ) 50 mm 0.25 0.45 0.70 0.80 0.85 0.85
Glass wool (panel 35 kg/ m³ ) 25 mm 0.20 0.40 0.80 0.90 1.00 1.00
Glass wool (panel 35 kg/ m³ ) 50 mm 0.30 0.75 1.00 1.00 1.00 1.00
Open window 1.00 1.00 1.00 1.00 1.00 1.00
Glass 0.03 0.02 0.02 0.01 0.07 0.04
Spanacustic (Manville) ceiling panel 19 mm - 0.08 0.71 0.86 0.68 -
Acustidom ceiling panel (Manville) 4 mm - 0.72 0.31 0.38 0.79 -
Prismatic ceiling panel (Manville) 4 mm - 0.70 0.61 0.70 0.78 -
Profil ceiling panel (Manville) 4 mm - 0.72 0.62 0.69 0.78 -
Auratone (USG) 5/8” cracked ceiling panel 0.34 0.36 0.71 0.85 0.68 0.64
Cortega cracked ceiling panel (AWI) 5/8” 0.31 0.32 0.51 0.72 0.74 0.77
Wooden seat (0.8 m² /seat) 0.01 0.02 0.03 0.04 0.06 0.08
Thick upholstered seat (0.8 m² /seat) 0.44 0.44 0.44 0.44 0.44 0.44
People on wooden seat (0.8 m² /person) 0.34 0.39 0.44 0.54 0.56 0.56
People in upholstered seat (0.8 m² /person) 0.53 0.51 0.51 0.56 0.56 0.59
Standing people (0.8 m ² /person) 0.25 0.44 0.59 0.56 0.62 0.50
REVERBERATION TIME.
After the period of early reflections, reflections of reflections, and

reflections of reflections of reflections, begin to appear; and so on, giving
rise to a very complex situation in which the reflections become more and
more dense. This permanence of sound even after the source is interrupted
is called reverberation.
However; In each reflection, part of the sound is absorbed by the
surface, and another part is reflected. The absorbed part can be transformed
into tiny amounts of heat, or spread to another neighboring room, or both.
The reflected part maintains its sound character, and will travel within the
enclosure until it encounters another surface, in which again part will be
absorbed and another part will be reflected. The process continues like this
until most of the sound is absorbed, and the reflected sound is too weak to
be audible, that is, it is extinguished.
The word requires shorter reverberation times than music, because the
most significant part of the word are the consonants, which are both weaker
and shorter than the vowels. Consequently, with a high reverberation time
the vowels are prolonged too much, masking the consonants that follow
them, which reduces the intelligibility of the word. Music, on the other
27
hand, benefits from a considerable reverberation time, since it allows
sounds to be better spliced and small imperfections in execution hidden,
while providing a spatiality that is desirable in music.
ACOUSTIC ABSORBING MATERIALS .
It is often necessary, both in performance halls and in recording and

monitoring studios, to carry out specific treatments to optimize acoustic
conditions. This is achieved with acoustic absorbing materials, that is,
materials specially formulated to have high sound absorption. The most
economical is glass wool, which comes in two forms: as felt, and as rigid
paper. Absorption increases with thickness, and also with density. Allows
very high sound absorption. The drawback is that it must be separated from
the acoustic environment by means of protective panels whose purpose is
twofold: to protect the glass wool from people, and the people from the
glass wool (since the particles that could be released not only hurt the skin
but which when breathed accumulate irreversibly in the lungs, with the
consequent danger to health).
Another type of material is polyurethane (polyester urethane) or
melamine foams. They are materials that are manufactured faceted in the
form of anechoic wedges. This surface structure behaves like a sound trap,
since the sound that hits the surface of a wedge is reflected several times in
that wedge and in the adjacent one.
28
For acoustic treatment of ceilings, sound-absorbing ceilings based on
mineral fibers (basalt), fiberglass, cellulose fibers, cork, etc. can be used.
with various fantasy surface finishes. They are installed suspended by
means of frames at a certain distance from the slab. The greater the
separation, the better the resulting absorption, especially if some glass wool
is interspersed.
Two warnings are necessary here. The first refers to expanded
polystyrene (styrene). Although it is an excellent thermal insulator, its
acoustic characteristics are very poor.
The second warning is regarding the custom of covering ceilings with
egg boxes, under the belief that they are good sound absorbers. They are
actually not effective for this application, because they lack the necessary
porosity and volume. Perhaps the confusion originates from the similarity it
presents to anechoic wedges. They are not recommended for any serious
acoustic applications.
Floor treatment is normally carried out with rugs, which are more
effective if placed on porous plant fiber rugs (burlap, jute) or polyester. The
effect of carpets is not limited to absorbing sound, but rather they attenuate
the sounds of footsteps or objects that fall or rub against the floor (for
example, microphone cables). For the same structure, the absorption of a
carpet increases with thickness. The type of fiber that makes up a carpet
(wool, nylon) does not significantly affect its absorption coefficient.
Finally, curtains can also be used as sound absorbers, especially when
they are part of the architectural design with some aesthetic or functional
purpose. It must be taken into account that the greater the separation from
the wall, the greater the effectiveness in absorption. Porosity is also
important, since a waterproof plastic curtain does not have absorbent
properties. On the contrary, a curtain made of thick fabric, velvet, etc., will
be quite absorbent. Absorption also increases with folding, gathering or
draping, that is, the ratio between the area actually occupied by the curtain
and the area of the stretched curtain. A curtain gathered at 50% can almost
double its absorption coefficient.
29
ACOUSTIC INSULATION.
Acoustically isolating an enclosure means preventing sounds

generated within it from transcending to the outside and, reciprocally,
preventing external noises from being perceived from within.
Acoustic isolation is very important in everything that has to do with
professional sound. If the venue is a concert or show venue in which music
is played at a high sound level, it is necessary to prevent the sound from
transcending into annoying noises in the neighborhood. If it is a recording
room or a radio studio, any noise from outside will contaminate the sound
that you want to broadcast or record. This is why thick (and therefore
heavy) walls offer greater insulation than thin ones.
A more detailed analysis indicates that it is possible to obtain greater
acoustic insulation through double or multiple partitions. In other words,
given a quantity of material (for example 20 cm. of concrete thickness) we
can get more out of it if we divide it into two parts (in this case two 10 cm
walls. each one) and we separate it with an air space. If the air space is
filled with some absorbent material (glass wool) the result is even greater
insulation.
This type of structure is widely used with gypsum rock plates
(Durlock, Placo, Plador). These plates are made of plaster covered on both
sides by cellulose (cardboard). The thickness is normally about
12mm. and they are usually used in pairs separated by 50, 70 or 90 mm.
using sheet metal profiles. The space between both plates is filled with
glass wool.
30
31
The double partition concept is also used to build highly sound-
insulating windows, like the “fish tanks” that separate the control room
from the recording room of the studios. In this case, two sheets of thick
glass of different thicknesses are used (for example 6 mm. and
8 mm.), fixed to the frame using non-hardening silicone putties. Absorbent
material, such as glass wool or polyurethane foam, is placed on the inside
edges.
To prevent condensation from occurring inside due to temperature
differences, which would fog up the windows.
The following table details the values of PT (transmission loss) at

various frequencies and STC (sound transmission class), corresponding to
various materials and structures.
32
PT at frequency
Material or structure STC
125 250 500 1000 2000 4000
Concrete (90mm) 37 30 30 37 35 38 41
Concrete (140mm) 45 30 34 41 48 56 55
Concrete (190mm) 53 37 46 46 54 59 60
Concrete (290 mm) 50 33 41 45 51 57 61
Concrete (90 mm) + air (25 mm) +
Fiberglass (65 mm) + concrete 62 49 54 57 66 71 81
(90mm) + gypsum board (16mm)
Gypsum board (Durlock) (12 mm) 28 15 20 25 29 32 27
Gypsum board (Durlock) (2*12mm) 31 19 26 30 32 29 37
Gypsum board (12mm) + air (90mm) +
33 12 23 32 41 44 39
gypsum board (12mm)
Gypsum board (2*12mm) + air (90mm)
37 16 26 36 42 45 48
+ plasterboard (12 mm)
Gypsum board (2*12mm) + air (70mm) +
45 23 30 45 49 52 52
gypsum board (2*12mm)
Gypsum board (12mm) + air (20mm) +
fiberglass (50mm) + gypsum board 45 21 35 48 55 56 43
(12mm)
Gypsum board (2*12mm) + air (40mm) +
fiberglass (50mm) + gypsum board 55 34 47 56 61 59 57
(2*12mm)
Glass (6mm) 31 25 28 31 34 30 37
Laminated glass (6 mm) 35 26 29 32 35 35 43
Glass (3mm) + air (50mm) + glass
38 18 26 38 43 48 35
(3mm)
Glass (3mm) + air (100mm) + glass
54 29 35 44 46 47 50
(6mm)
Solid wood door (24 kg/ m² ) without
22 19 22 26 24 23 20
weather stripping
Solid wood door with weather stripping 26 22 25 29 25 26 28
Solid wood door (24 kg/m ² ) + air (230
mm) + hollow # 18 sheet steel door (26
49 35 44 48 44 54 62
kg/m ² ) + magnetic weather stripping on
the frame.
33
SIGNALS AND SYSTEMS.
INTRODUCTION.
The interconnection between two or more devices, such as

microphones, amplifiers, equalizers, speakers, etc.; They give rise to what
is called a system. These devices have the common characteristic that they
all receive, process and deliver signals of some type. The concept of a
signal is that of a variable magnitude over time that transmits or transports
information. In the case of sound systems, there are two main types of
signal: acoustic and electrical. The conversion between both types of signal
is carried out by means of devices called transducers (microphones,
speakers, headphones). Other types of signals involved in audio systems
are magnetic (tapes, hard drives) and optical (compact discs, fiber optic
transmission).
SIGNS.
As we have seen, all equipment, devices and systems work with

signals, that is, with variable magnitudes that transmit information,
precisely the waveform of sound.
The original signal is the sound itself as it reaches the transducer
element, that is, the microphone. The microphone converts the sound signal
into an electrical signal. The electrical signal should have exactly the same
waveform as the sound signal, the sound signal is a sound pressure, while
the electrical signal is a voltage (or voltage).
34
SYSTEMS.
A system is the result of interconnecting a set of devices with their

own entity. There are two basic reasons to think in terms of systems when
approaching the solution of a technical problem, the first is because it is
easier to subdivide the problem into several simpler subproblems, and then
solve these using more specific tools and resources. The second reason is
that the solutions to the subproblems may have other uses, or, in other
words, when solving other complex technical problems, situations similar
to some previously solved arise.
For this, there are standards that are respected by all those who intend
to offer a versatile product. In this sense, today the issue has become much
easier, but it is still necessary to verify the interconnectivity of the
equipment to ensure its optimal functioning.
BLOCKS DIAGRAM.
Block diagrams are used to graphically represent the interconnections

between the various devices. A block diagram is a drawing in which each
component of a system is represented with an appropriate symbol (for
example a triangle or a rectangle), with one or more inputs, through which
the signal or signals to be processed enter, and also with one or more
outputs, through which the already processed signal or signals are obtained.
A characteristic of block diagrams is that they are schematic.
Connection details are not presented.
The lines, which in reality are pairs of cables, are generally
represented by a single wire.
NOISE.
35
Noise is understood as any unwanted signal that overlaps the useful
signal. In sound systems there are two types of noise: acoustic noise and
electrical noise. Acoustic noise is ambient noise itself, formed by a number
of near and far sources that overlap. For example, the noise of vehicles on
the street or people talking, the noise of machines, ventilation, etc.; that
filter through defects in sound insulation.
Electrical noise originates from physical phenomena that take place
within electrical and electronic circuits.
The most important thing is to keep it below the hearing threshold,
which today is possible although expensive. Another type of electrical
noise is that which originates from magnetic media, such as tapes or disks,
which is transferred to the electrical signal. In digital systems, there is also
quantization or digitization noise.
Noise can also be classified according to its frequency spectrum.
There are continuous spectrum, discrete spectrum and mixed spectrum
noises. The electrical noise of the components is continuous spectrum, that
is, it contains all the frequencies of the audible spectrum. Ambient noise is
usually mixed. Continuous spectrum noises are combined, such as wind
noise or the combination of numerous relatively distant sources, with
noises that have specific frequencies, such as the noise of fans or other
machines.
The transformers of the power supplies, as well as the ballasts of the
fluorescent tubes, vibrate with the frequency of the power line, that is, 50
Hz, also causing audible hums. These hums can also be electrically
coupled, through the cables, which is why these must be of excellent
quality and adequately shielded (the shielding or metal covering of the
cables allows this defect to be eliminated).
DYNAMIC RANGE.
36
The dynamic range, RD, is a parameter associated with a signal that
represents the relationship between the maximum and minimum signal
level.
DISTORTION.
Distortion is the deformation of the shape of a signal.
The waveform changed but the fundamental frequency remains the

same. This implies that harmonics of the fundamental appear, which are
added to the original signal. This type of distortion is called harmonic
distortion, and is specified by a parameter called total harmonic distortion,
THD ( T otal Harmonic Distortion ), which expresses the harmonics
generated as a percentage of the original sinusoidal signal.
The audible effect of harmonic distortion is to add some brightness to
the timbre of the sine wave. In most cases it is not an unpleasant effect,
even though it alters the original signal.
When distortion becomes excessive, the device is said to clip or enter
clipping.
FREQUENCY RESPONSE.
37
A transducer receives an input signal that we can graph, then
produces an output signal that we can also graph. The differences between
the input and output graphs represent the frequency response of the
transducer. If the line represented on the output graph is equal to the input
line, the frequency response is said to be flat. A frequency response curve
graphically indicates the effect a transducer has on the pitch of a sound.
SIGNAL PROCESSING.
The first processor in the audio chain is the microphone, a transducer

capable of converting a sound signal into an electrical signal. The need for
this device arises from the fact that subsequent processing is carried out
exclusively by electronic means, which handle electrical signals.
The second signal processor is the amplifier. This component takes a
small level electrical signal and transforms it into an electrical signal of the
same waveform but with greater amplitude, that is, it amplifies it.
A third type of processor is filters. These devices pass certain
frequencies of the input signal spectrum, and block the remaining
frequencies. An example of this is the tone control (record-treble, or
record-mid-treble) and the equalizers. These are used for several functions:
to emphasize some frequencies in the input signal spectrum but which for
some reason suffer attenuations within the system; to correct room acoustic
problems; to achieve certain special effects; to reduce the total system noise
by blocking the frequency bands in which there is noise but no signal; etc
Another class of processors are compressors, expanders, limiters and
gates. The application of these allows us to accommodate the dynamic
range, that is, the relationship between the maximum and minimum level of
a signal. Thus, compressors reduce the dynamic range, without affecting
the fidelity of what is heard.
Limiters are protections designed to avoid very high peaks that would
destroy some part of the system (for example the tweeters). The expanders
allow you to recover the dynamic range, as well as reduce low-level noise,
the gates eliminate the signal when its level is below a certain threshold,
which prevents residual noise from the preceding device from appearing
during silence.
Finally, effects processors, devices that create effects such as
reverberation, early reflections, enrichment of the spectrum of a sound, etc.
The purpose of these is to give more realism to a recording or a sound
38
system, allow greater expressiveness, improve the quality of the sounds or
their perception, etc.
ELECTRICITY.
ELECTRICAL CIRCUITS.
Let's begin by defining an electrical circuit as a set of components

interconnected by means of conductive wires (cables), in such a way that
there are one or more closed loops. These components can be power
sources (for example a battery, or the home line), signal sources
(microphones, synthesizers), electrical devices (lamps, motors, resistors
and capacitors) or electronic components (diodes, transistors and integrated
circuits or chips). ).
ELECTRIC CURRENT.
Electric current is nothing more than electric charges in motion.

It is also defined as the amount of electrical charge that circulates
through a conductor, and is measured in a quantity called ampere (or
ampere), its abbreviation A.
For small currents, the unit milliamp (mA) is used, that is, one
thousandth of an ampere. For example, a current of around 0.3 A flows
through a common light bulb, that is, 300 mA.
STRAIN.
This magnitude is measured using the volt (or volt) as a unit, which is
why it is sometimes called voltage. Voltage is measured between two
points in a circuit. Thus, in the home line, the voltage between live and
neutral is 220 V. Similarly, the voltage between the positive and negative
of a common battery is 1.5 V.
39
A more specific example is the voltage between the terminals of a
microphone. In this case the levels are very small, so it is convenient to use
the millivolt (mV) equivalent to one thousandth of 1V.
IDEAL VOLTAGE SOURCE.
In reality there are no ideal sources, but some sources, such as the 220
V home line and car batteries, come close.
Voltage sources can be classified into power supplies and signal
sources. Power supplies are those that provide the energy that a circuit
needs to function. They are normally of high value, from a few volts to
several hundred volts, and constitute an inaccessible internal part of the
equipment. Signal sources, on the other hand, normally have much lower
levels, which can be less than 1 mV.
The power sources can be continuous (cells, batteries, electronic
circuit sources) or alternating (220 V home power distribution line). The
signal sources are almost always alternating.
ENDURANCE.
It is the opposition to the passage of electrons through a resistor or

resistor in a direct current circuit. Symbolized by R, it is expressed in units
of ohm ( Ω ), or its multiple, kiloohm ( ΚΩ ), equal to
1000 Ω.
Every real transducer (cable) has some resistance. Which increases
with the length of the conductor, also increases as the section is reduced.
Thus, a thick wire will have low resistance and a thin wire will have high
40
resistance. For that reason, cables intended to carry large currents must be
thick.
ELECTRICAL POWER.
Electrical power P is the electrical energy delivered to a device per

unit of time (typically 1 sec.). It is expressed in watt (or also watt), a unit
that is abbreviated W.
IMPEDANCE.
It is the opposition to the passage of electrons in an alternating current

circuit. Symbolized by Z and measured in ohm.
MICROPHONES.
INTRODUCTION.
The first element of the audio chain is the microphone, a transducer

capable of converting a sound signal into an electrical signal. We will study
some important parameters and specifications.
There are two main trends in the manufacture of microphones:
pressure microphones, which receive the signal through only one side of
41
the membrane; and gradient ones, which receive signal from both sides of
the membrane.
Microphones create a controversy in the world of Professional Audio,
the most marked difference between one engineer and another is the use of
these since there are no rules in the use of microphones, only guides and
recommendations. The only thing that is proven is that to have a decent
microphone technique we must use two elements:
1. Extensive experimentation.
2. Knowledge of engineering and theoretical behavior of

microphones.
DIRECTIONALITY.
An important characteristic of microphones is their directionality.

Due to its construction, and the principles of acoustics, the sensitivity of a
microphone varies depending on the angle with respect to its axis from
which the sound comes.
The directional characteristics of a microphone can be indicated by
means of a directional diagram or polar diagram. This type of diagram
indicates how the sensitivity of the microphone varies with the angle
between the sound source and the main axis, that is, that direction of
maximum sensitivity.
42
The directional pattern of a microphone varies with frequency,
because for high frequencies, the wavelength is small, comparable to the
size of the microphone itself, which projects acoustic “shadows” on itself
that depend on the orientation and length. wave (and therefore frequency).
The omnidirectional pattern has the same sensitivity in all

directions, so it does not need to be focused towards the source. This type
of microphone is used precisely when it is necessary to capture ambient
sound, regardless of its origin. Omnidirectional microphones have less
variation in the polar pattern with frequency, which is why they do not
“color” the sound coming from directions different from the main axis, that
is, they do not present important peaks in the frequency response.
43
Cardioid pattern microphones are quite directional, their sensitivity
being greatly reduced in the direction opposite to the main one. Due to their
directional characteristic, cardioid microphones have the peculiarity that
when the source is very close to the microphone (3 or 4 cm), the frequency
response changes, increasing the sensitivity in low frequencies.
This is called the proximity effect, and it is used by vocalists to
thicken the tone of their voice.
One of the main applications of the cardioid pattern (also called
directional or unidirectional ) is to take sound from a certain source
whose position is quite stable, such as a musical instrument, rejecting
sounds coming from other sources.
Thus, the capture of ambient noise will be considerably reduced,

since the noise comes from all directions. An omnidirectional microphone
will pick up the noise in its entirety, while a cardioid microphone will pick
up only a portion of the noise.
Another polar pattern is the figure of eight , so called because it is
shaped like an 8 . This type of microphone could also be called
bidirectional , since it is strongly directional in the two directions parallel
to the main axis. In the direction perpendicular to this axis the sensitivity is
zero, which makes it possible to eliminate noise coming from these
directions.
Like cardioids, bidirectional also exhibits the proximity effect,
increasing bass sensitivity when the source is very close to the microphone.
44
DYNAMIC MICROPHONES.
There are several mechanisms for converting sound energy into

electrical energy used in microphones. The most common are dynamic
microphones and capacitive microphones.
Dynamic microphones, also called moving coil microphones, are
made of a coil with several turns of copper wire that oscillates along a
cylindrical magnet core. The coil is driven by a diaphragm that vibrates in
accordance with the pressure variations of a sound wave.
A disadvantage of dynamic
microphones is the so-called handling noise (i.e. the noise caused by
moving or touching the microphone).
The main advantage of this type of microphone is its robustness and
tolerance to adverse operating conditions, such as variations in temperature
or humidity, high sound pressure levels, shocks and shocks, etc.; which is
why they are especially suitable for live sound.
45
Another advantage is that they do not require their own power
supplies to generate an electrical signal in response to a sound.
CAPACITIVE MICROPHONES.
Capacitive microphones (also called capacitor , condenser , or

electrostatic ) are based on the use of an electric field instead of a
magnetic field. They are made up of a very thin diaphragm (typically 5
microns thick) plated in gold, and a metal back plate that is usually
perforated or slotted.
The way to charge the capacitor plates is through external

polarization, which is achieved by connecting the microphone to a constant
voltage source through a resistor. This source can be a battery built into the
body of the microphone itself, or a remote source located in the console or
amplifier, called phantom power. This source can have a value between 1.5
V and 48 V depending on the microphone model.
Currently, a type of pre-polarized condenser microphone is widely used,
that is, with an intrinsic internal polarization, which in principle does not
require the use of a phantom source. They are called electret, and are
characterized because one of the plates contains a special insulating film in
which, during manufacturing, electrical charges have been introduced that
remain trapped in the internal structure with no possibility of escaping.
In any case, they come with an internal amplifier that requires some type of
power.
IMPEDANCE.
There are high impedance microphones (greater than 10,000 Ω; that

is, 10 ΚΩ) and low impedance (less than 500 Ω). In professional sound,
low-impedance microphones are used almost exclusively, because they are
46
less noisy, and offer fewer difficulties for wiring, especially when large
distances are involved as is often the case in live sound. The output voltage
level is, in general, very small, especially in low-impedance microphones,
which is why it is necessary to use preamplifiers to raise the voltage to the
level normally required by audio mixers (consoles). These preamplifiers
are built into the mixing consoles, and appear on the microphone inputs.
These preamplifiers should not be confused with the impedance conversion
amplifiers included in capacitive microphones (both polarized and pre-
polarized or electret).
NOISE.
In microphones there are two noise production mechanisms. The most

obvious is the capture of ambient noise. The reduction of this noise is
linked to the reduction of the ambient noise itself, and to the use of the
directional pattern to reduce noises that come from directions other than the
useful signal (a voice or instrument).
The other mechanism is electrical noise. This noise can only be
reduced (but not eliminated) by designing the microphone so that it has
very low impedance (for example 100 Ω), and also by using high quality
materials and highly refined manufacturing processes in its manufacture.
DISTORTION.
Another specification of interest in microphones is distortion. Distortion

differs from noise in that it is a deformation of the wave, while noise is an
independent signal that is added to the signal.
The distortion phenomenon normally occurs at high signal levels; the
specification is usually associated with the maximum sound pressure level
that the microphone supports. A possible specification could be:
THD: 1% at 125 dB NPS.
Another way of expressing the same thing would be:
Maximum NPS: 125 dB at 1% THD
The maximum value, unless otherwise noted, represents an

operational level, and not a security level. The difference between the two
is that the operational level is a level at which the microphone is still
47
working reasonably well, while the safety level is one above which the
microphone may deteriorate. Reducing the signal to the operating level will
restore correct operation.
PHANTOM POWER.
In capacitive (or condenser) microphones, the power supply is

specified, which can be with a battery located in the body of the
microphone itself or with a phantom power source. The outputs of the
capsule (capacitor) enter amplifiers whose sole purpose is to reduce the
excessively high impedance of the capacitor to the required levels. These
amplifiers, as well as the capsule itself, require power, which comes from
the console through certain resistors. Then follow two capacitors whose
purpose is to eliminate direct currents, and only allow audio frequencies to
pass through. The capacitors are connected (via the XLR connector) to the
cable and then to the console. The phantom supply, which is usually built
into the console today, applies the same voltage to the signal lines through
another pair of resistors.
Because not all microphones are condenser, it is logical to wonder

what happens when you connect a dynamic microphone, for example, if the
microphone is balanced (all good professional quality ones are supposed to
be), nothing happens, since that the same tension is applied to both parts of
the capsule, so the applied tension is zero. But if the microphone were
unbalanced, the voltage would be applied directly to the coil, which would
potentially be very detrimental to the microphone, and could easily be
destroyed.
CHARACTERISTICS OF SOME MICROPHONES.
HANDHELD VOCAL MICROPHONE (Hand Held).
These microphones are designed to withstand certain factors that

come into play when the microphone is used in the hand by a singer or
announcer. These factors are:
48
1. They must avoid the noise generated by handling the microphone.
When a microphone is held in your hands, friction noises are
generated, which are very annoying for the audience and can even
damage the PA equipment.
2. They must be for rough use. In this situation there is a great

possibility that the microphone will fall by accident, which is why
they are designed to withstand strong blows and falls.
3. It should have a closed polar pattern. The tendency to Feed Back
of a microphone that is constantly changing position is very great,
which is why cardioid, supercardioid or hypercardioid
microphones are used. This avoids this trend as much as possible.
4. It must have a proximity effect. Experienced singers and
announcers take advantage of this effect to color their voice. This
is especially useful when the pressure generated by the user is
light, and thus an effect similar to the compensation loudness of
homemade equipment is achieved.
5. You should avoid breath and wind noises. In this application, the
microphone is exposed to frequent gusts of wind generated by the
user or the environment (in outdoor applications), which is why it
is equipped with a metal mesh and a layer of insulating material
(foam, cloth , etc) which is placed in front of the membrane and
greatly mitigates this problem.
6. It must have an adequate frequency response. The vocal range
does not require excessive bass (below 80 Hz) or excessive treble
(above 8 KHz), which is why the capsules are designed to have a
drop in their response curve in those sectors, in some cases filters
are added. in the equipment's electronics in order to improve its
characteristics. It is also necessary that the microphone curve has
an emphasis between 1 KHz and 3 KHz. Because this frequency
range makes the voice more intelligible and improves its
performance in a musical context.
PEDESTAL MICROPHONES.
There are a large number of microphones that are designed only to be

placed on pedestals or special bases since due to their morphology they are
difficult and impossible to hold comfortably in the hand.
49
This is the case with tube microphones (which are large and heavy)
and multi-input or multi-capsule models, which are also difficult to hold
manually.
In this case, an external suspension is required to avoid vibration and
shock noises.
LAVALIERS MICROPHONES.
These microphones are designed to be attached to the user's body (tie,

shirt, collar, or any piece of clothing). They are very small to prevent them
from being visible and this gives them an excellent frequency response (the
body of the microphone does not act as a physical barrier), to achieve that
size electret membranes are generally used.
For this function, omnidirectional capsules are used, which allows the
microphone to be placed in any position without changing its response, but
this makes them first candidates in the Feed Back trend. In television
applications (where monitors are not used) the omnidirectional pattern is
ideal, but in theater or applications that do use monitors, unidirectional
versions are preferred.
Unidirectional ones have the advantage of picking up less ambient
sound and having a more constant response, but they can cause problems if
placed on the neck, since their response and sensitivity changes every time
the user moves their head.
These microphones are only used in theater, television and cinema,
since their tendency to Feed Back is exaggerated and their reception of
ambient noise makes them completely unsuitable for live concerts.
HAND SET MICROPHONES (HEAD WORN).
These microphones are used when free use of the user's hands is
necessary, such as in the case of singers, instrumentalists and machine
operators.
These should be lightweight and should reject a lot of breath noise
and plosive consonants that are generated by being located so close to the
user's mouth (less than 2 inches). Generally dynamic capsules are used but
currently condenser capsules in Diferiod or Cardioid versions are
successfully implemented.
50
CONTACT MICROPHONES.
These microphones are designed to capture the signal in a solid

medium. And they must adhere to the body of the sound source. For this,
piezoelectric elements are used. The frequency response changes
dramatically with respect to where the microphone is placed on the body.
The most recent versions are for violins, violas and bowed instruments in
general. The microphones come in the shape of a bridge, achieving
excellent response and rejection of ambient noise and Feed Back.
ULTRA-DIRECTIONAL MICROPHONES.
These microphones have the virtue of largely rejecting sounds that

arrive outside the exact axis point (of their axis). This makes them ideal for
capturing very distant sounds and they are used in applications such as
cinema and recording animal sounds.
To achieve this polar pattern, different techniques are used:
1. SHOTGUN: This technology is also called interference tube

since a long tube (with continuous holes along it) is placed in front
of the capsule which has the mission of generating phase
cancellation of sounds that are generated out of axis on the front
section of the microphone.
2. PARABOLICS: These employ a conventional capsule placed in

front of a parabola-shaped shell which concentrates all sounds
coming from the front of the microphone to the back of the
capsule except for those that hit the capsule itself. This increases
its directionality and sensitivity drastically. The problem is that its
frequency response depends on the diameter of the capsule,
existing models have a flat response in the range greater than 1
KHz and this limits their use to the recording of sounds of birds or
insects, although they are also used in transmission by television
of sporting events.
3. RIFLE: These microphones consist of an omnidirectional capsule
and several long tubes of different lengths placed in a circle in
front of the capsule, which creates delay cancellation of sounds
that do not fall in perfect axis.
4. ZOOM: These are very high-tech equipment which are connected
to the zoom factor of the lens of a film or video camera. The
microphone uses a sophisticated system to aurally focus sounds
51
that are at a distance relative to that indicated by the camera's
focusing mechanism.
AUTOMATIC MICROPHONES.
These microphones incorporate a gate into their circuit, in this way

the gate closes the microphone output when there is no SPL greater than
“x” affecting the membrane. The value of “x” is established by the
manufacturer and can occasionally be modified by the user, changing the
internal configuration through Jumpers.
They are very useful in Theaters and other applications.
DUMMY HEADS MICROPHONES.
These microphones are used only for experimental functions and

analysis of human hearing behavior. It is based on an exact replica of the
upper part of the body of a mature adult (from the shoulders up), two
specially designed microphones are used and placed inside the Dummy's
ears, with this very precise analyzes can be made of how the human body
in frequency response.
IN THE EAR MICROPHONES.
Based on the same principle as Dummy but in this case a real human
body is used.
It consists of a pair of miniature microphones specially designed to be
placed inside a person's ears. This allows us to analyze the response
changes caused by the person's movement.
STEREO MICROPHONES.
They are units that use two capsules to achieve a stereo response, they
are manufactured with the “XY” or “MS” stereo microphone patterns.
There are models whose configuration is completely variable, being able to
52
change the polar pattern, sensitivity, phase and position of each of the
capsules, making it an extremely versatile tool. These microphones offer us
two balanced outputs which we can add to mono to achieve infinite
variants of the polar pattern by changing the parameters of each capsule.
TIPS AND SOME TECHNIQUES FOR MICROPHONES.
DO N'T YOU HAVE ENOUGH GAIN BEFORE THE

REFEED?
Here's what you can do:

(in order of importance)
1. Move microphones away from speakers.

2. Move the speakers as far as possible from the microphones.
3. Move the speakers (PA) close to the public.
4. Reduce the number of active microphones.
5. Use directional microphones and speakers.
6. Eliminate acoustic reflection near the microphones.
7. Reduce the reverberation of the place with acoustic treatment.
8. Use an equalizer to reduce system gain at feedback frequencies.
TECHNIQUES FOR HANDHELD MICROPHONES.
1. Keep the microphone at an appropriate distance to balance the sound.

2. Use some low frequency roll-offs to control the proximity effect.
3. Use pop filters to control noise caused by breathing.
53
4. Do not create excessive noise when holding the microphone with
your hand.
5. Point the microphone toward your mouth and out of reach of other
sound sources.
6. have good dynamic control with the voice more than with the
movement of the microphone.
TECHNIQUES FOR LAVALIER MICROPHONES.
1. Observe the most appropriate place and orientation.

2. Use pop filter if you need to, especially with unidirectional capsules.
3. You should not breathe and/or touch the microphone, nor touch the
cable.
4. Do not turn your head too far from the microphone.
5. Turn off the lavalier microphone when using any other microphone
over the altar or pulpit.
6. Care should be taken to ensure that you can hear clearly when
different voices speak.
TECHNIQUES FOR CHOIR MICROPHONES.
1. Find the right places to place the microphones.

2. Use the minimum number of microphones.
3. Turn off microphones that are not being used.
4. Don't over-amplify the chorus.
5. Do not sing directly into the microphone.
6. Make sure they sing with a natural voice.
TECHNIQUES FOR CONGREGATIONAL

MICROPHONES.
54
1. Techniques for choir microphones (previous)
Further:
2. Use only enough level to add ambience to the recording.

3. Do not mix the microphone area with the sound to try to reinforce
the system.
TECHNIQUES FOR MICROPHONEING ACOUSTIC

INSTRUMENTS.
1. Experiment with placement to get the best sound.

2. Maintain a constant distance between the sound source and the
microphone.
3. Use shockmount if there is any noise on stage (especially with
movement or knocks on the microphone stands).
4. Do not place the microphone where it is likely to be hit by the
instrument or the musician.
5. If someone sings and plays an instrument at the same time, do not
allow the voice to be picked up by the instrument's microphone.
TECHNIQUES FOR MICROPHONEING A PIANO.
1. Experiment with microphone placement to get the best sound.

2. Use shockmount if vibration is a problem.
3. Adjust the height of the piano lid to obtain better sound and/or
isolation.
4. Listen for interference effects with multiple microphones.
5. Do not allow your voice to be picked up by the instrument's
microphone.
55
Stereo system Types
Microphone Position
To choose Microphone
Maximum Axis
Response at
XY 2 - Cardioid 135°
Space:
Coincident
Maximum axes
ORTF (Radio
Response at
and TV
2 - Cardioid 110°
Organization.
Space: Close
French)
Match (7")
Maximum axes
US
Response at 90°
(Dutch Sound 2 - Cardioid
Space: Close
Organization)
Match (12")
56
Maximum axes
2 Response at 90°
Stereosonic
Bidirectional Space:
Coincident
Cardioid
1 – Cardioid
M.S. forward;
1
(Mid-Side) Bidirectional at
Bidirectional
ends
2 – Cardioid
Parallel angles
either
A-B Space: 3 – 10
2 - omni-
feet
directional
57
Characteristic Omni- Super- Hyper- Bi-
Cardioid
s directional cardioid cardioid directional
Answer
Pattern
Polar
Angle
Of 360° 131° 115° 105° 90°
Coverage
Angle
Maximum of
---- 180° 126° 110° 90°
recazo
(null angle)
Rejection
Later
0 ~25dB 12dB 6 dB 0
(relative to
front)
Sensitivity
To the Sound
100 % 33 % 27 % 25 % 33 %
Atmosphere
(rel. Omni)
Factor
Distance 1 1.7 1.9 2 1.7
(rel. Omni)
58
SELECTION GUIDE FOR LIVE RECORDING AND
PRESENTATION.
VOICES
Live Voice Live Voice Headband Voice Ensemble
(Dynamic) (Condenser) Voice studio
Beta 58 A Beta 87 A WCM 16 KSM 32 KSM 32
SM 58 SM 87A WH 20 XLR SM 81 SM 81
Beta 57 A BG 5.1 SM 10A SM 7A SM 94
SM 57 SM 12A SM 87 A BG 4.1
BG 3.1 512 Beta 87 A
BG 2.1 BG 5.1
BG 1.1
INSTRUMENTS
Amplifier Amplifier Kick Drum Snare Aerial Toms
Bass and floor guitar
Beta 56 Beta 52 Beta 52 Beta 57 A Beta 98 D/S
Beta 57 A SM 7 A Beta 91 Beta 56 Beta 57 A
SM 57 Beta 57 A Beta 57 A SM 57 Beta 56
BG 6.1 Beta 56 SM 57 BG 6.1 SM 57
BG 3.1 SM 57 BG 6.1 BG 6.1
BG 2.1
Conga Marimba and Piano² Cymbals

High Hat² percussion²
KSM 32 Beta 98 D/S KSM 32 KSM 32 KSM 32
SM 81 Beta 56 SM 81 SM 81 SM 81
SM 94 Beta 57 A SM 94 Beta 57 A Beta 91
BG 4.1 SM 57 BG 4.1 SM 57 BG 4.1
59
Strings Bass Instruments Instruments Saxophone
Acoustic Copper Wood
KSM 32 KSM 32 KSM 32 KSM 32 KSM 32
SM 81 Beta 52 Beta 98/S SM 81 Beta 98/S
SM 94 SM 81 Beta 56 Beta 98/S SM 7 A
Beta 98/S SM 94 Beta 57 A BG 4.1 Beta 56
BG 4.1 BG 4.1 SM 57 Beta 57 A
SM 11 SM 57
Harmonic Guitar Orchestra² Record

Ambient Acoustics
520 DX “Green KSM 32
KSM 32 KSM 32 KSM 32
Bullet” (pair)
SM 81 SM 57 Beta 57 A SM 81 VP 88
(MS stereo)
SM 94 SM 58 Beta 56 SM 94 SM 81 (pair)
BG 4.1 Beta 91 BG 4.1 SM 94 (pair)
Beta 57 A SM 57 BG 4.1 (pair)
SM 57 BG 3.1
SM 11
60
AMPLIFIER.
INTRODUCTION.
The amplifier is the first purely electrical signal processing block. Its
purpose is to increase the level of signals coming from low-level
generators, such as microphones, until reaching a level suitable for a certain
application, such as driving a speaker or acoustic box (horn). Amplifiers
exist in all electronic devices or equipment, such as digital watches, remote
controls, computers, etc. We are interested in audio signal amplifiers.
REVENUE.
The small signal that needs to be amplified is applied between two

terminals called input and the already amplified signal is obtained from two
other terminals called output. One of the most fundamental parameters of
an amplifier is gain, or amplification.
Gain is often expressed in decibels (dB).
SIGNAL LEVEL.
The other fundamental property of amplifiers is the signal level they

are capable of handling. There are three characteristic signal levels: low
level, line level and power level. Low-level signals correspond to the signal
produced directly by transducers, such as microphones, vinyl record
pickups, and tape playback heads. Line level signals are the result of
applying preamplification to low level signals, but they are also the signals
produced by various equipment such as cassette decks, compact disc
players, tuners, synthesizers and other electronic musical instruments, etc. .
Finally, the power level is that required to drive the loudspeakers (speakers,
horns) or speakers.
CLASSIFICATION OF AMPLIFIER.
Amplifiers can be classified according to the signal they handle, thus

there are low-level amplifiers, or preamplifiers, and high-level amplifiers,
61
or power amplifiers. Preamplifiers are intended to bring low-level signals
to line level, which is the standard level handled by the inputs and outputs
of mixing consoles. Power amplifiers receive line level signal at their input
and amplify it to the power level.
In reality, preamplifiers are normally built into consoles or signal
generating equipment such as cassette decks, so their specifications are not
under the user's control. The same does not happen with power amplifiers,
for which various technical characteristics are specified that need to be paid
attention to.
MAXIMUM OUTPUT POWER.
First, the maximum output power is specified, which is indicated for

one or more load impedance values, normally 8 Ω or 4 Ω (since these are
the typical impedance values of speakers and boxes). acoustics). Amplifiers
incorporate overload protection, this can cause the power for 4 Ω to be less
than double that for 8 Ω.
The power delivered by the amplifier will allow us to calculate the
sound pressure level produced by the complete system.
SENSITIVITY.
The second specification of power amplifiers is related to gain. This

is sensitivity, defined as the value of the input voltage necessary to produce
maximum power. It can be specified in V or dBV.
SIGNAL/NOISE RATIO.
A third specification of amplifiers is their signal to noise ratio, S/R

(signal to noise ratio).
This specification is important when the dynamic range of the signal
is in consideration. This is defined as the difference in dB between the
maximum and minimum output level. The minimum level is, many times,
62
the noise level of the signal generator or the ambient noise captured by a
microphone and/or recorded on any medium. For an adequate selection of
an amplifier, it should be considered that its S/N ratio, for the output level
at which it will actually work, is greater than the dynamic range of the
signal to be amplified.
FREQUENCY RESPONSE.
The next specification of amplifiers is the frequency response.

Indicates the variation of gain (usually in dB) with frequency.
Nowadays amplifiers widely cover the audio frequency range, and
some even exceed it. It is not uncommon to find amplifiers that are “flat”
up to 100 KHz, which is doubtful if they provide any real benefit, because
frequencies above 20 KHz are inaudible to the human ear.
DISTORTION.
The next characteristic that must be specified in amplifiers is

distortion. Which consists of the deformation of a signal due to a non-linear
transfer and has clearly audible effects. Some distortions are favorable to
the ear, and are even added on purpose, as in the effects called aural
exciter, enharas, etc. In the case of amplifiers, there is always some
distortion. An interesting fact is that the distortion introduced by tubes is
more favorable to hearing than that produced by transistors, which is why
for some years now tube amplifiers have been reappearing on the market,
which seemed completely surpassed by those from tubes. solid state.
These amplifiers are generally much more expensive, because the

tube is a device with little diffusion in other areas.
There are two types of distortion: Total Harmonic Distortion (THD)
and intermodulation distortion (IMD).
Total harmonic distortion refers to the deformation that a perfect
(pure) sine wave experiences when passing through an amplifier. The result
of this deformation is the appearance of harmonics of the original
frequency of the sinusoid, that is, in addition to the original amplified
sinusoid, a residue appears formed by its successive harmonics.
Intermodulation distortion originates from the mutual interference
that occurs between two sinusoidal tones of different frequencies added
63
together in the same channel (not to be confused with what would be
channel separation).
Although it is often not given due attention, intermodulation
distortion is much more harmful to the sound signal than harmonic
distortion. Indeed, the harmonic distortion of an isolated musical sound
tends to reinforce some harmonics, giving greater brightness to the sound.
When two or more sounds are presented, however, intermodulation
distortion produces tones that are not harmonically related to any of the
original sounds, producing a noticeable and unpleasant effect.
Most amplifiers today have IMD values less than 0.1%, and some
register values much lower even.
INPUT IMPEDANCE.
Another specification is the input impedance, which is the impedance

that is measured externally at the input terminals.
Typically the input impedance of amplifiers is in the range between
10 KΩ and 50 KΩ, and that of consoles is a few hundred Ω.
In amplifiers with balanced inputs, two input impedance values are
specified, one for balanced inputs and one for unbalanced inputs, with the
balanced value being twice the unbalanced value.
CUSHIONING FACTOR.
In amplifiers, the output impedance is not generally specified,

although it is common to provide an equivalent data, which is the damping
factor. It is the ratio between the nominal load impedance and the actual
output impedance.
Current amplifiers achieve damping factors of several hundred. This
is important to ensure that the speaker impedance does not significantly
modify the real voltage on the speaker, for a similar reasoning to that of the
impedance of the signal source and the amplifier input.
SEPARATION OF CHANNELS.
64
Another specification that is usually given in the case of stereo
amplifiers is channel separation (crosstalk), also called crosstalk. This
specification describes to what extent a signal appears at the output of an
unexcited channel as a consequence of a signal applied to the input of the
other channel. The way to determine it is to apply signal on one channel
and nothing on the other.
OTHER FEATURES.
Finally, there are a series of accessory specifications, among them are

the input and output connectors. Typically, inputs for professional
equipment are of the XLR type, which can be balanced, unbalanced, or
switchable between both arrangements. Also used, in some cases, ¼" (6.35
mm) Jack type connectors for the three-way balanced case, also called TRS
( Tip – Ring – Sleeve , that is, tip – ring – sleeve) and for the two-way
unbalanced case called TS ( T ip – S leeve: tip – sleeve). Another connector
used is the RCA type, which is unbalanced. In general, amplifiers provide
more than one possibility for greater flexibility of use, although the XLR
connection is preferred.
The output connection is not made through connectors but through a

terminal block in which the ends of the cables are directly adjusted, of the
type called binding post (there is no elegant translation into Spanish; it
could be translated as binding post). Another specification is the
consumption of electrical energy.
Dimensions and weight constitute another specification since
amplifiers with similar characteristics can vary significantly in size and
weight.
It is convenient to know that lighter amplifiers offer advantages when
moving and installing, since they use forced ventilation for power
transistors or integrated circuits using electric fans, which generate noise
that, in certain applications, is noticeable and annoying. The solution is to
place the amplifier in a separate room with adequate acoustic insulation.
65
SPEAKERS AND SPEAKERS.
INTRODUCTION.
To complete a minimal acoustic system that is functionally complete,

a transducer must be added to the microphones and amplifiers that
transform electrical energy back into acoustic energy. Examples of this are
speakers, headphones and earphones. The most widespread speakers are
moving coil speakers, both for low and high frequencies.
CLASSIFICATION BY FREQUENCY RANGES.
Both in high fidelity sound (good quality sound for family

consumption) and in professional sound (superior quality sound for
66
recordings or shows) it is common to use speakers that include two or more
speakers that cover different frequency ranges. Thus, for low frequencies,
that is, less than 500 Hz, the so-called woofers are used (whose direct
translation would be “barkers”), speakers whose diameter varies between
8" (20.3 cm) and 18" (45.7 cm) (although the most common is between 12"
and 18"). For medium frequencies, between 500 Hz and about 6 KHz, the
formerly called squawkers are used, whose typical diameter is between 5"
(12.7 cm) and 12" (30.5 cm). Finally, for high frequencies, above 1.5 KHz,
and sometimes above 6 KHz, so-called tweeters are used.
In high-power professional sound, the speakers have a single speaker,
and one or more boxes are placed for each frequency range, with
characteristics optimized for said range.
MOBILE COIL SPEAKERS.
It is made up of a magnetic circuit, formed in turn by a base or rear

plate with a cylindrical central core or pole mounted on its center, a
permanent magnet in the shape of a large washer, and a front plate in the
shape of a smaller washer. Between the central pole and the previous plate
there is an air space called the
air gap, over which there is a
powerful magnetic field.
The coil is housed in said air
gap, which is mounted on a
paper tube that communicates
it with the cone.
67
As the coil is immersed in a magnetic field, when electric current
circulates through it, a force is generated that gives it movement. This
movement is transmitted to the cone or diaphragm, and this then acts as a
kind of piston, propelling the air out or in depending on the polarity applied
to the coil. This process generates successive waves of compression and
rarefaction of the air that propagates as sound.
COMPRESSION EXCITERS.
For high-frequency speakers, a variant of the previous structure is

used, called a compression driver. The name is due to the fact that the
exciter generates very high sound pressures, which are then brought to
normal values by a horn, which operates as an acoustic impedance adapter.
If an impedance adapter is not used, the radiated acoustic power
would be much lower and the system would lose performance.
As in direct radiation speakers, there is a diaphragm driven by a coil
immersed in the magnetic field of a magnet, this diaphragm is dome-
shaped instead of being conical. Below the diaphragm, there is a phase
correction element, also in the shape of a dome, but with internal
perforations that communicate with the throat of the exciter (driver). This
makes it possible to compensate for the different distances that the sound
must travel from the different points of the diaphragm to the throat,
avoiding sound cancellations.
68
SPEAKER BOXES.
As we said, the speakers are mounted in acoustic boxes, the purpose

of which is to improve the characteristics of sound radiation, as well as
facilitate maneuverability and protect the drivers.
We see that if at a certain moment the cone moves outwards, there
will be a compression of the air in front of the speaker and a decompression
of the air behind it. This creates what is called an acoustic dipole, and leads
to an irregular directional pattern, as well as lower sound performance.
Adding an acoustic box, or sonodeflector, or baffle allows you to correct
this problem.
69
There are several types of baffles. The conceptually simplest baffle
consists of mounting the speaker flush with a wall over a hole drilled in it,
so that compression and decompression waves cannot mix. This type of
baffle is called infinite baffle (or infinite sound deflector), and it allows the
entire wave radiated by the speaker to be used. Theoretically it is one of the
best systems, for practical reasons its application is generally not feasible,
since it would require an inconveniently large unused space behind the
wall.
The second type of baffle is the closed one. This speaker uses a box
covered on the inside with absorbent material, so that its interior behaves
like an open space. The result is similar to that of an infinite speaker.
The third type is the open baffle or ventilated baffle, which is the
most used for bass boxes. There are also several types of ventilated baffles.
The simplest, called a bass reflector, two waves are radiated from this
speaker. The first is that created by the compression waves of the external
or front face of the cone, radiated directly. The second is that created by the
decompression wave of the internal or rear face of the cone, which exits
through the opening or mouth of the baffle. If this wave came out
immediately, because it was in counterphase to the compression wave (that
70
is, opposite), it would cancel with it, giving a very weak sound. But it is
made to travel a certain distance before leaving, so that when it leaves, the
other wave has already become decompression, and then the two are in
phase, reinforcing the sound.
Within the bass reflectors, there is a variant to improve the very low
frequency response that consists of making the sound travel a longer path
inside the box through a labyrinth. Another variation is to add a tube into
the opening, called a tuning tube, which adds a resonance to the box.
71
SENSITIVITY.
The next specification is sensitivity, and it is related to the sound

pressure level that can be obtained from the speaker with a certain power. It
is defined as the sound pressure level at one meter distance (on the axis)
when an electrical power of 1 w is applied. Sometimes it is specified
directly as sound pressure level (SPL) at 1m and 1w, without using the
word “sensitivity”.
It is not so simple to determine the sound pressure level at a distance other

than 1m, since it does not depend only on the baffle but on the acoustic
characteristics of the environment where it is used.
When the loudspeaker is operated in an acoustic environment with
reflections, such as any normal room or hall, the sound pressure is the
72
result of the direct field (sound coming from the loudspeaker) and the
reverberant field (sound coming from the multiple reflections). At a
distance of 1m we can accept that the direct field predominates, which is
why the true sound pressure level practically coincides with the sensitivity,
but the same does not happen at much greater distances.
FREQUENCY RESPONSE.
The frequency response of individual loudspeakers should be

distinguished from the frequency response of a loudspeaker, whether it
consists of a single loudspeaker or several loudspeakers covering various
ranges. The frequency response is a graph that indicates how the sensitivity
of the speaker or baffle varies with frequency. Of all the audio components,
the speaker is probably the most imperfect, and therefore the frequency
response is usually more irregular than that of the microphone and much
more than that of the amplifier.
DIRECTIONALITY.
The sensitivity of a loudspeaker also fluctuates with the direction and

to this is added the cabinet's own interference, especially noticeable at high
frequencies, all of which gives rise to a certain directional pattern. In reality
73
there is a horizontal and a vertical directional diagram, since the baffles are
not symmetrical. In both cases the respective diagrams correspond to
measurements carried out in an anechoic chamber (without echo).
It is interesting to note that high-frequency speakers, due to their

horns, are very directional.
The coverage angle or beam width is usually specified, that is, the
angle that can be covered with a drop in sensitivity no greater than
6 dB.
74
For this reason, it is recommended that they focus directly on the
public and without obstacles, such as columns, decorative elements, etc. In
those cases where the audience occupies a greater angle than the coverage
angle, clusters are used, that is, groups of speakers oriented in such a way
that each one covers a part of the audience. An example is theaters with
high seats, balconies or gatherings. In that case, one pair of speakers will
cover the lower stalls, another the upper stalls, and so on. Of course, proper
mounting must be provided to achieve this non-horizontal orientation.
Another important consideration is to ensure that the low frequency
and high frequency boxes of each channel are concentrated in the same
place. This is to avoid modifying the sense of power directionality of the
sound. If they are too far apart, a feeling that the source is diffuse could be
created, which impairs the intelligibility of the word and the music.
FREQUENCY PROCESSORS.
INTRODUCTION.
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To begin, let us remember that sound waves are pressure variation
phenomena. The air in the place where a piece of music is heard is
compressed and decompressed by the movement of the speaker
diaphragms. In a free space in which there was no obstacle between the
speaker and our ears, the sound would reach it as it was emitted by said
speaker. In a closed space things are very different since moving air
particles collide with objects, walls, etc.; Each of these shocks modifies the
sound wave in at least two parameters: its direction and its intensity.
Its direction varies because the surfaces act on the sound. Part of the
energy with which the wave arrives is lost in that sound reflection, so that
its intensity is lower than before the collision; The amount of energy
absorbed by the object with which the sound wave collides depends on its
nature. Thus, while marble returns more than 90% of the incident energy,
certain materials such as rubber and fiberglass are capable of retaining,
under certain conditions, up to 85 or 90% of the sound energy that reaches
them. The set of conditions that occur in a certain location with respect to
the behavior of sound waves is usually known as the local acoustics.
It must be understood that it will be difficult to find two venues with
identical acoustics, so if the same sound system is heard in two venues the
auditory sensation will not be the same, it will sound different in one and
the other since the waves will be modified. .
Obviously there should only be one way for the equipment to sound
good, it will be the one in which the original sounds are faithfully
preserved.
Other factors that also threaten perfect hearing are: noise, static,
humming, etc. Unfortunately, it is not always possible to avoid these
drawbacks.
Most current equipment incorporates two tone controls, one for bass
and one for treble; In some cases they incorporate a third control that
regulates the amplitude of the medium frequencies. In general, room
acoustics tend to introduce progressive loss or gain with frequency, both in
the bass and treble, so that these tone controls are sufficiently effective.
The perfect adaptation between the equipment and the listening room
can be achieved if it were possible to govern the level of the signals for
each specific frequency value. This is practically impossible, as there
would be an incredible amount of controls since the human hearing
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spectrum goes from 20 Hz to 20,000 Hz; that is, 19,980 different
frequencies.
The problem would be simplified if we think that the human ear does
not respond linearly with frequency, but rather does so logarithmically;
That is, the step from 100 to 150 Hz (50 Hz difference) seems identical to
that between 1000 and 1500 Hz (500 Hz difference), for this reason the
keyboard of a piano is divided into octaves (octave is the distance or
difference that exists between a certain frequency and its double, for
example: La 4 = 220 Hz, La5 = 440 Hz, La 6 = 880 Hz, etc.).
We have thus arrived at the reason for the existence of frequency
processors, also called equalizers, whose mission is to match or equalize
the response of the system with the listening rooms, so that hearing is free
of appreciable differences between the ideal and the real.
Although there are a large number of equalizers, they all pursue the
same objective: to effectively correct the frequency response curve of the
audio chain; microphones, speakers, etc. These settings are affected
according to the taste of the engineer or producer.
It is important to know the main elements that make them up:
FILTERS.
The most important elements that an equalizer is made up of are the

so-called filters.
A filter is a system in which a signal of fixed amplitude and
frequency is introduced; a signal with the same frequency is obtained at the
output and whose amplitude can vary depending on the type of filter. Thus,
a filter acts by amplifying or attenuating the amplitude, although it
maintains the frequency of the input signal.
All slopes developed by the filters are measured in decibels per
octave (dB/oct).
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Among the main filters are:
 High pass filter (high pass filter, HPF)

 Low pass filter (low pass filter, LPF)
 Band pass filter (BPF)
 Band rejection filter
 Narrow band filter (notch filter)
PASS FILTER – HIGH AND PASS FILTER – LOW.
The high-pass filter (HPF) consists of a circuit that acts as an

eliminator of frequencies lower than a certain cut-off or pass frequency. A
low-pass filter (LPF) will be exactly the opposite, that is, it attenuates any
signal with a frequency higher than the cut-off or pass frequency.
These two filters are also called cut – off filters. That is, a high-pass
filter allows the passage of frequencies higher than the chosen one or cuts
off the frequencies lower than this; A low-pass filter allows the passage of
frequencies higher than the chosen one or cuts off frequencies higher than
this one.
Filters can also be called high-pass: short-low.
(low cut), and the filters pass – low: short – high (high cut).
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PASS AND BAND REJECTION FILTER.
These filters are obtained from a high pass and a

low-pass, and consists of a filter that rejects or passes any frequency
between two cuts. They are applicable only to frequencies within the range
indicated by the frequencies determined by the high-pass and low-pass
filters (HPF – LPF).
Within the band pass and rejection filters is the central
frequency, the main basis of interest for equalization systems.
NARROW BAND FILTER.
The narrow band filter (notch filter) acts by attenuating a very narrow
range of frequencies. It is used to suppress any disturbance that occurs at a
certain frequency point.
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QUALITY FACTOR.
Intimately linked to the central frequency is the so-called

quality factor (Q) of a circuit. It is a measure of the quality of the filter, this
factor is useful when dealing with band-pass filters, since its value will be
greater the narrower this bandwidth is.
That is, at high values of Q, the filter will affect a few

frequencies close to the center frequency, while at low values of Q a
greater number of frequencies close to the center will be affected.
The quality factor of an equalizer can be obtained by dividing
the center frequency by the bandwidth. Bandwidth is understood as the
frequency range that is –3dB to the sides of the center frequency (as can be
seen in the filter graphs). This calculation is summarized in the following
formula:
Q = _______ Center frequency_____________

Upper frequency – Lower frequency
Below is a comparative table of Q values with respect to their

equivalent values in octaves:
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Q Octaves
8.65 1/6
5.76 1/4
4.32 1/3
2.87 1/2
1.90 3/4
1.41 1
0.92 3/2
0.67 2
0.40 3
Once the characteristics, elements and operation of the filters

are known, some of the most common types of equalizer will be described.
STAGED EQUALIZERS (SHELVING).
These equalizers appeared when they wanted to modify the less

audible frequency areas (bass and treble). They consist of a single filter that
acts at the extremes of the frequency spectrum captured by the human ear.
The normal operating frequencies of these equalizers can be 50
to 100 Hz for the lower zone and 10 to 15 KHz for the upper zone.
GRAPHIC EQUALIZERS.
The transition from a single filter (with a single control) to a

complete equalizer (with several controls, one for each operating
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frequency) was made by placing a series of band-pass filters in parallel, so
that the sound is divided into several bands, each one can be regulated
independently of the others.
This is how the graphic equalizer was born, it receives its name
from the ease of visualizing the position of its controls (linear
potentiometers on which the amplification or attenuation can be directly
read), these equalizers constitute a highly requested method of tone control
in which the spectrum The auditory system is divided into very narrow
bands; filters are generally based on octave intervals or fractions thereof.
The band has an individual slider control that provides an
increase or decrease in amplitude.
These devices provide an excellent method of equalizing the
frequency responses of equipment and rooms.
Graphic equalizers are the most widespread and can be found
from 5 to 33 or more controls.
Remember that the audible spectrum is between 20 Hz and 20
KHz, it runs around 10 octaves (9.96). Therefore, the most typical graphic
equalizer is the one-octave one, where we find 10 control points, one for
each octave. In more complete systems, 1/3 octave equalizers, in which
each octave is divided into 3 controls, having approximately 30 controls.
Below are the operating frequencies of the most common graphic
equalizers:
 2 Octaves: 62.5, 250 Hz; 1, 4, 16KHz.

 1 Octave: 31.2, 62.5, 125, 250, 500 Hz; 1, 2, 4, 8, 16KHz.
 1/3 Octave: 20, 25, 31.5, 40, 50, 63, 80, 100, 125, 160, 200, 250,
315, 400, 500, 630, 800Hz; 1, 1.25, 1.6, 2, 2.5, 3.1, 4, 5,
6.3, 8, 10, 12.5, 16, 20 KHz.
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PARAMETRIC EQUALIZERS.
Parametric or variable parameter equalizers are very useful since it is

possible to make very precise and selective adjustments. It is common in a
parametric equalizer to have different sets of filters, allowing many parts to
be modified at the same time, which makes adjustment a little more
complicated.
A parametric equalizer not only has variable amplitude cutting or
boosting, it also has a variable frequency selector, which acts on different
areas of the audio spectrum, and it also has a control that adjusts the
bandwidth that will be cut or boosted.
The benefit of parametric equalization is greater freedom and
flexibility in adjusting the response curve.
Probably the most difficult parameter to understand regarding
parametric equalization on the Q.
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The number of values that Q can cover in parametric equalizers can
range from a section of the audio spectrum smaller than a semitone to
several octaves.
It is important to mention some suggestions to facilitate its
adjustment:
 To apply it in mixes, it is convenient to start with a very low Q

setting, adjusting the range of frequencies you want to change and
then cutting or increasing the gain very subtly until the appropriate
tonal color is achieved; then the desired bandwidth is increased.
 In the case of individual tracks, you start with a high Q and with the
gain increased exaggeratedly, then you choose the frequency to
change; Once found, the amount of increase or gain is adjusted and
at the same time the Q control is moved until the appropriate tonal
color is found.
SEMI-PARAMETRIC EQUALIZERS.
Many equalizers found on mixers use an adjustable frequency system,

based on a much simpler parametric design known as semi-parametric. In
these, the frequency and the amount of cut or boost are variable while the
width of the filter is fixed.
To achieve a better result with equalization equipment, it is essential
to know the frequencies that comprise the auditory spectrum.
FREQUENCY SPECTRUM.
VERY LOW FREQUENCIES (16 – 60 Hz).
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These frequencies give the musical program the feeling of power,
especially if they occur suddenly, if they occur continuously or with
emphasis they mask the musical passage and dirty it, they should be used
with discretion.
They have little musical content.
LOW FREQUENCIES (60 – 250 Hz).
Contains the root notes of most instruments, too much reinforcement

in this band can make the passage boomy.
LOW MIDDLE FREQUENCIES (250 – 2000 Hz).
This band delivers body and richness to the sounds. Excessive

reinforcement of this area can produce sounds with nasal or telephone
effects; this reinforcement can cause auditory fatigue in the listener.
MEDIUM FREQUENCIES (2 – 4 KHz).
These frequencies provide intensity, presence and definition. This band is

the most important for the recognition of voice and instruments; Misuse
can achieve confusing results and cause auditory fatigue.
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HIGH MEDIUM FREQUENCIES (4 – 6 KHz).
They are the frequencies responsible for the clarity and transparency
of the voice and most instruments. The increase in the equalization of this
area produces the same effect on our hearing as if the level had been
increased by 3 dB. Attenuation of these frequencies produces transparent
and distant sounds.
HIGH FREQUENCIES (6 – 16 KHz).
This band is used to control the brightness and also the clarity of the
sounds. Excessive reinforcement can produce crystalline, metallic sounds
and hisses in vowels and s.
VERY HIGH FREQUENCIES (16 – 20 KHz).
In this last section of the spectrum the high ends are controlled,
creating sharp sounds and generating hiss. This band has little musical
content.
The ideal is to know in the most precise way the range of frequencies
in which the instrument to be treated develops.
Frequencies
Instrument Harmonics
fundamental
Drum 30 – 147Hz 1 – 6KHz
Snare 100 – 200Hz 1 – 20KHz
Saucers 300 – 587Hz 1 – 15KHz
Piano 27 – 4196Hz 5 – 8KHZ
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Electric bass 41 – 300Hz 1 – 7KHz
Double bass 41 – 294Hz 1 – 5KHz
Cello 65 – 698HZ 1 – 6.5KHz
Viola 131 – 1175Hz 2 – 8.5KHz
Fiddle 196 – 3136Hz 4 – 15KHz
Electric guitar
82 – 1319HZ 1 – 3.5KHz
(amplifier)
Electric guitar
82 – 1319Hz 1 – 15KHz
(direct)
Acoustic guitar 82 – 988Hz 1 – 15KHz
Tuba 49 – 587Hz 1 – 4KHz
Trombone 73 – 587Hz 1 – 7.5KHz
French horn 87 – 880Hz 1 – 6KHz
Trumpet 165 – 988Hz 1 – 7.5KHz
Bassoon 62 – 587Hz 1 – 7KHZ
Clarinet 1658 – 1568Hz 2 – 10KHz
Oboe 261 – 1568Hz 2 – 12KHz
Flute 261 – 2349Hz 3 – 8KHz
low voice 87 – 392Hz 1 – 12KHz
tenor voice 131 – 494Hz 1 – 12KHz
loud voice 175 – 698Hz 2 – 12KHz
soprano voice 274 – 1175Hz 2 – 12KHz
Below are various aspects about the control of some frequency zones
in some musical instruments:
Depth 60-80 Hz, body 100 Hz, stiff sound 300-800 Hz,
Drum:
attack (click) 2-6 KHz.
Snare: Body 200-240 Hz, clarity 5-7 KHz.
Depth 240 Hz, roughness 1-3 KHz, attack 5 KHz,
Air Toms:
brightness 10 KHz.
Depth 80-120 Hz, roughness 1-3 KHz, attack 5 KHz,
Floor Toms:
brightness 10 KHz.
Setbacks
Clanck 200 Hz, brightness 7-12 KHz.
and dishes:
Resonance 200-240 Hz, presence and slap
Congas, bongoes:
5KHz.
Depth 60-80 Hz, boomy sound 600 Hz, attack and
Electric bass:
presence 1-2.5 KHz, string noise 3 KHz and up.
Electric guitar: Body 100-240 Hz, squawk 600 Hz, presence 2-3 KHz,
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hiss 6 KHz and up.
Depth 80-120 Hz, body 240 Hz, clarity and presence
Acoustic guitar:
2.5-5 KHz, bright 7-10 KHz.
Strings: Body 120-240 Hz, brightness 7.5-10 KHz.
Body 120-240 Hz, warmth 500 Hz, harshness 3 KHz,
Metals:
stridency 7-7.5 KHz, key noise 10 KHz and up.
Electronic
Depth 80-120 Hz, body 200-250 Hz, presence 2.5 KHz.
organ:
Depth 80-120 Hz, presence 2-2.5 KHz, attack and
Acoustic piano:
clarity 10 Khz.
Body 100-150 Hz, nasal sound 500-1 KHz, presence 3-
Male voices:
5 KHz, hiss 6 KHz and up.
Voices Body 200-250 Hz, nasal sound 500-1 KHz, presence 3-
feminine: 5 KHz, hiss 6 KHz onwards.
The best way to equalize is based on always comparing the natural

signal with the equalized one. It is also essential to know the limitations
that the signal will have later if the recording is intended to be marketed.
DYNAMIC PROCESSORS.
This type of processor allows you to have control over the dynamic
range of the signal. Among its main exponents are compressors, expander
limiters and noise gates. No automatic amplitude controlling device can
provide the aesthetic judgment and finesse necessary for effective sound
control programming. However, automatic amplitude control systems
contribute to the control of sudden variations and unpredictable sources,
protection of the equipment from possible over-modulation, as well as
greater use of signal dynamics.
COMPRESSOR – LIMITER.
These types of processors help control the general dynamic range of a

signal, and thus work at higher levels, obtaining a better signal-to-noise
ratio.
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Providing protection against voltage overloads, avoiding distortions
and intervening in the achievement of certain widely used musical effects,
proving to be a very useful tool in the creative aspect.
Compressors and limiters themselves are processors that reduce the
dynamic range. Compression is the name given to the process of gain
reduction that is more or less continuous. Limiting responds to a sudden
reduction in the signal.
Fundamentally, these processors are characterized because their
output gain can be preset without depending on the input level, that is, their
gain will remain constant even if the input level of the device varies within
a range of values called compression – limitation margin; reducing high
levels to the set point without altering the rest of the processed signal.
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The main problem is that there is no standard setting of
parameters, but rather they must be analyzed and adjusted by the user in
each use and according to the needs of the audio signal.
The limiter is a device through which a program can pass
without alterations in the signal until it reaches a critical value. If the input
signal rises above the threshold value, the system gain is automatically
reduced so that the output cannot rise significantly above the threshold
value.
The compressor is similar to the limiter, because when the starting

point is exceeded the system gain is reduced; but its action is less dramatic,
so an increase in the input value above the critical value produces a small
increase in the output. The gain reduction control can be adjusted.
NOISE GATES.
Noise gates are essential processors in current audio work. The

mission of these devices is the elimination or attenuation of a signal when
the input level is below the determined threshold.
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This will allow establishing a threshold point that is above the level of
background noise or parasitic and unwanted sounds, eliminating them,
while the desired signal will exceed the marked threshold, crossing it
without problems.
Percussive signals, due to their brevity and rapid decay, are the
easiest to process, while signals with a lot of sustain and long decay are
impossible to process; When the signal decays and exceeds the preset
threshold, it will be abruptly cut off.
It must be understood that no device is intelligent enough to
distinguish between an acceptable musical signal and noise and unwanted
signal. Like compressors and limiters, there is no general standard for their
application. The result always depends on the relationship between two
parameters.
EXPANDERS.
The most common way these devices operate is by expanding the

signal level downward when it is below the threshold, so the signal will be
proportionally reduced creating a wide dynamic range.
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The most important parameters generally present within dynamic
processors are:
THRESHOLD.
It is also called the starting point, the point at which the processor will
begin to work. In the case of compressors and limiters, all signal that
exceeds the threshold level will be processed; while in expanders and gates,
only signals that are below the threshold will be processed. The threshold is
variable to find the appropriate point in each case, this setting is given in
dB.
RATIO.
With this parameter it is possible to regulate correspondence between

the input level and the output level.
In the case of compression – limitation the ratios are between 1:1, 2:1,
3:1, 4:1 ......10:1, up to ∞:1 A ratio of 1:1 will indicate that the processor
will not act, a ratio of 2:1 will mean that only half of each signal that
exceeds the threshold will be output, while a 3:1 ratio will only obtain a
third of the entered values. The approximate reduction system is 10:1,
higher values will indicate limitation, implying high levels of reduction.
In the case of expanders, the ratio will work inversely to that of
compressors – limiters. A 1:2 ratio will deliver double the values that fall
below the threshold, a 1:3 ratio will deliver triple the values and so on.
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ATTACK TIME (ATTACK).
This parameter determines the characteristic and size of the signal

that will exceed or fall below the threshold before being processed. The
attack time can be defined as the time the processor needs to begin
performing its function. This time is usually adjustable and is given in
millisecond values. The attack time is set in such a way that it is not so fast
as it can alter the figure. Slow attack times are useful when a high degree of
processing is required.
Below are some values within the attack times of compressors and
expanders.
COMPRESSOR ATTACK TIMES.
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EXPANDER ATTACK TIMES.
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RECOVERY TIME
(RELEASE).
It is the time it takes for the device to return to its normal level once
the compression, limitation or expansion carried out has ceased. Recovery
time is used to make gain variations as less noticeable.
Recovery times are adjustable from a few milliseconds to several
seconds.
Its use requires great care since rapid changes in the recovery change
will cause an unpleasant gain change and as a consequence an effect of
abrupt level fluctuations, while slow changes will cause the processor not
to have time to recover before the next signal exceeds the threshold,
causing inconsistent processing.
In some systems the attack and recovery time is not adjustable by the
operator; The manufacturer deliberately selects the times in order to
facilitate the task for the user, providing fast times for transient peaks and
slower times for continuous levels.
Below are some of the values within the recovery times of
compressors and expanders.
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COMPRESSOR RECOVERY TIMES.
EXPANDER RECOVERY TIMES.
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GAIN (GAIN).
The function of the gain may be to counteract the change in level, or

to set an appropriate level for the next stage within the audio chain.
An important function within dynamic processing is the key or side

chain connection, which works as a detection system that allows providing
criteria to the operation of the processor. That is, the processor will operate
only when an external signal connected to the detector is present in the
device; when the signal is not present, the processor will not act.
Using this detection system it is possible to create various auxiliary
systems for dynamic processing such as de-esser , ducking , etc.
The de-esser consists of a device that raises the high frequencies of a
signal and then attenuates them using a compressor, thus eliminating
hissing frequencies.
They are normally used in voice processing to eliminate sibilance (s,
sh, tsch, etc.). Although they can be found in independent units, it is also
possible to create them by an equalizer and a compressor – limiter.
The equalizer adjustment is carried out by attenuating the frequencies
below 3 KHz. From this frequency onwards, the upper bands must be
increased by approximately 6 to 10 dB per octave. It is important to keep in
mind that the hissiest frequencies are around 8 KHz.
The compressor setting should have an approximate ratio of 6:1 or
more, with fast attack and recovery times. The threshold will be adjusted
so that the attenuation only acts when the hissing signal to be eliminated is
present.
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Ducking is a device commonly used in applications where it is
desired to automatically superimpose one signal on top of another. The
function consists of activating the compressor – limiter through an external
signal. That is, the compressor – limiter is inserted into the music signal,
while the voice microphone signal is connected to the detector (key or side
chain) of the compressor – limiter. Thus, when the voice microphone level
exceeds the marked threshold , the music level will be automatically
reduced, allowing the voice to be heard more clearly.
This system is also used between instruments such as bass and bass
drum, where the bass drum takes the place of the voice and the bass takes
the place of the music; thus achieving a relationship with greater impact.
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MIXING CONSOLES.
INTRODUCTION.
In a practical system, it is common for several devices to intervene at

the same time. For example, there could be signal sources such as
microphones, synthesizers, or compact disc players; processors, such as
compressors, equalizers and reverberators; monitoring or sound
reinforcement systems, consisting of an amplifier and speakers; and,
finally, recording equipment such as an analog multitrack recorder or a
DAT.
The last device we will study is the mixing console, also called a
mixing console or mixer. This equipment can range from a simple 4-
channel mixer to a complex 48 or 56-channel digital console.
SPECIFIC FUNCTIONS OF A CONSOLE.
The primary function of a mixing console is to provide the sum of

various electrical signals, each attenuated or amplified from its original
level by a factor adjustable by the operator.
There are several reasons why it is generally preferable to sum signals
electrically rather than acoustically:
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1. The electrical adjustment of the levels of the various signals is
much simpler, since it is done by means of sliding potentiometers.
An acoustic adjustment would imply a great flexibility of the
musicians to achieve a careful dynamic balance between the
various parts, which can be a very great demand (especially in pop
music, and not in classical music).
2. The acoustic adjustment would involve stereophonic recording
with a pair of distant microphones, which would capture less
signal but equal or more ambient noise, worsening the signal/noise
ratio. Nowadays it is much easier to combat electrical noise than
acoustic noise.
3. Electrical adjustment can be performed on signals from

multichannel recording, making adjustments possible after
recording. If the dynamic balance were done acoustically and the
result was recorded in stereo, it would no longer be possible to
modify or correct said balance.
4. If one of the sound sources is a synthesizer or other electronic
instrument, it is preferable to directly process the electrical signal
that it generates. If it were first transformed into sound (through a
monitoring system) and then returned to the electrical signal
through a microphone, the signal/noise ratio would be worse.
5. The electric mix allows the possibility of recording in play back,
that is, adding another instrument (or voice) later on a base made
up of several previously recorded instruments.
Large consoles usually have a considerable number of input channels

(for example 24), some of which are monophonic and others stereophonic,
and a smaller number of output channels (for example 6). Each output
channel is the superposition of some input channels with corresponding
level settings.
Output channels are often called groups, or also submasters. In
general, there is a stereo output channel on which all signals can be mixed,
including those from the submasters. It is called master, or main mix.
Another function of the consoles is the possibility of incorporating
effects into the signals. Some effects are connected in series, meaning that
the entire signal passes through them. The consoles provide insert
connections on each input channel for this purpose.
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Other effects are connected in parallel, so that part of the signal is
processed and part is not. For this, auxiliary sends (send) and returns
(return) are provided.
Finally, the vast majority of consoles allow equalization (generally
simple, that is, 2 or 3 bands) on each input channel, and sometimes also on
the output (in this case there are usually 7 or more bands).
In addition to the previous functions, there are others of an
administrative nature, which facilitate the operator's work in terms of level
adjustments, error location, connection flexibility, versatility, etc. These
functions are carried out through the following elements:
1. VU meters , that is, analog level indicators (using moving needle

instruments) or quasi-analog, using LED bars (light-emitting
diodes) or LCD (liquid crystal display). In general, it is provided
with a single stereophonic VU meter, switchable by means of
buttons between several sources (individual inputs, output). In the
most complete consoles you can make a VU meter for each
channel, in addition to one for each output.
2. Monitoring outputs for the control room.
3. Mute buttons, which allow one or more channels to be silenced at
the operator's discretion.
4. Solo buttons, which allow you to listen, also as desired, to one
channel at a time (or more than one, if several buttons are
pressed).
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STRUCTURE OF A MIXING CONSOLE.
The main functions of a console are based on a few basic concepts such as:
amplification, attenuation, filtering and superposition of electrical audio
signals. However, the complexity of its topology (that is, the internal
structure of connections), as well as the large number of variants presented
in commercially available consoles, makes its understanding difficult. For
this reason it is convenient to describe some simple block diagrams, to
which we will add more elements.
In this example it is a console with 4 input channels and 2 output channels
(right and left). Each input channel has a line input and a microphone input.
TRS connectors are used for the line input and XLR for the microphone
input, in both cases with a balanced connection to reduce noise. In some
consoles both signals can coexist. In others, you can select by means of a
button which of the two inputs is active.
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The line input goes directly to a trim, while the mic input goes
through a preamp first. The purpose of level adjustment is to give
uniformity to the average level of the various input signals. This setting
provides a gain of up to 60 dB for the microphone input and up to 40 dB
for the line input.
After the level adjustment, a sliding potentiometer called a fader
(pronounced feíder; in Spanish, attenuator) appears on each channel, with
which the proportion in which said channel will be mixed with the others is
adjusted. The fader provides a gain between - ∞ and 10 dB.
The level adjustment allows you to work with an appropriate signal
level for the rest of the circuit, that is, neither too small as to have a poor
signal-to-noise ratio, nor so high as to cause any part of the circuit to go
into saturation.
Continuing with the signal path, you reach the pan setting (pan pot =
panoramic potentiometer = panoramic potentiometer). This setting splits
the signal into two parts: one goes to the right channel and the other goes to
the left channel.
When the setting is in the center position, the signal goes equally to
both channels. The purpose of this control is to virtually locate in space the
source that corresponds to each channel. In practice, to achieve highly
realistic special effects, panning must be complemented by adding a delay
on the weakest channel.
The signals from the right channel of the pan setting are directed to
the right mixer and those from the left channel to the left mixer.
A mixer is simply a signal adder, which adds all signals in equal
proportion. If you want one signal to appear in the final mix at a higher
level than another, the adjustment must be made using the faders of the
respective input channels.
Finally, the output of each mixer passes through a main fader, which
in turn allows the level to be adjusted independently on each output
channel. If the output channels designated as right and left are actually used
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for stereophonic sound, both main faders should be adjusted evenly so as
not to distort the stereo image that is assigned to each signal by panning.
EQUALIZERS.
The next element to add is an equalizer. The most rudimentary

version consists of a bass and treble tone control, but most consoles have at
least three bands: bass, midrange and treble. The central band is usually
semiparametric, allowing the center frequency to be adjusted between two
extremes, or parametric, also allowing the bandwidth or quality factor Q to
be adjusted. In some more complex consoles two central parametric bands
are provided, and in others all four bands are parametric. Finally, there are
consoles where the equalization section is a complete octave graphic
equalizer.
The equalizer is inserted between the level adjustment and the
channel fader.
The frequencies of the different bands are not standardized, although
it is common to find certain sets, for example, 100 Hz for the bass, 1KHz
for the mids and 10 KHz for the highs, or 80 Hz for the lows, 2.5 KHz for
the mids and 12 KHz for treble. In the case of parametric or semi-
parametric media, the frequency range is usually between 100 Hz and
10KHz. In parametric media the value of Q can vary between 0.5 and 10 or
more, although it is advisable to restrict the setting to a maximum of three
to avoid the ringing and hissing sounds that accompany very narrow
bandwidth filters (or Q very high).
In addition to the equalizer, consoles usually have low-frequency
high-pass filters (40 to 100 Hz), which are optionally inserted before the
equalizer, using a selector, to eliminate very low-frequency noises (hums,
engine noises, footsteps, etc.). There may also be a low-pass, high cutoff
frequency to reduce high-frequency noise in signals that do not contain
such frequencies.
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INSERTION CONNECTIONS (INSERTS).
One of the features that give power and versatility to the consoles is
the possibility of adding processing, through external equipment, to the
signals they receive. Insert connections provide the means to interleave
effects and other serial processors.
Y insert cables are used, terminated at one end in a TRS plug type
connector and at the other end in two TS plug connectors. By inserting the
TRS plug into the insert connector of the console, the internal connection
between the level adjustment output and the equalizer input is interrupted,
and these points are derived outwards through the TS connectors,
respectively directed to the input and output of the external processor.
These insert connections allow, for example, the use of compressors
or gates on specific channels as well as –essers, antipop, etc.
Th
e
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insert connections seen here are located practically at the entrance, after
level adjustment. It is possible to incorporate these types of connections at
other points in the signal path, even after mixing.
AUXILIARY CONNECTIONS.
Auxiliary connections are divided into send and return connections.

An auxiliary send is an output obtained by adding, in an auxiliary mixer,
the signals from the input channels.
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These signals first go through gain adjustments that allow them to be
mixed in different proportions than those used for the main mix. An
auxiliary return is a normally stereo input, which after a gain adjustment
enters the main bus.
AUXILIARY SHIPPING.
Regarding auxiliary sends, there are two possibilities: 1) that the

signals to be mixed are taken before the channel fader, and 2) that they are
taken after. In the first case you have the auxiliary pre fader, and in the
other auxiliary post fader.
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The auxiliary pre fader takes the signals before the respective faders,
it is not affected by the adjustments made to the channels with a view to the
main mix.
The auxiliary post fader is used specifically for parallel effects. The
signal sent to the effects processor is in this case affected by the fader, so
the processed signal will increase or decrease along with the raw signal.
On many consoles the aux sends can be switched between post fader
and pre fader. This allows greater versatility, since it leaves the operator the
choice between one possibility and another.
AUXILIARY RETURNS.
Normally the auxiliary returns receive the signal that returns from an
effects processor, and send it to the main bus. Since many effects have
stereo output (even if they have mono input), the auxiliary returns are
usually stereo.
GROUPS OR SUBMASTERS.
Large consoles usually have several outputs called groups (sometimes

also subgroups or submasters).
When there are several groups, on each input channel there are
routing selectors that allow determining which group or groups the signal
of said channel will be directed to. These selectors are located after the
channel pan setting, and select pairs of groups, for example: 1-2, 3-4, so
that partial mixes are stereophonic.
The group signal passes through a group fader (which is an overall
group volume control), from which the group output is obtained. This
output can be used for recording to a multitrack recorder, or can be added
to the main mix via a group pan setting.
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110
PHANTOM POWER.
Most current consoles provide a 48V phantom supply on their

balanced XLR microphone inputs, to bias capacitive microphones. This
source can generally be connected or disconnected globally using a switch,
although in the most complete consoles the connection is independent on
each channel. While dynamic microphones do not require power.
MONITORING.
Monitoring consists of the propagation of the signal resulting from a

mix for listening, either by the operator or by the musicians. There are two
types of monitoring: monitoring for the operator, which must sound exactly
like the main mix (or submixes, as the case may be), and monitoring for the
performer, which differs from the previous one in that it normally
emphasizes some parts that facilitate its execution. or interpretation. In the
latter case, the signals from the pre-fader auxiliary sends are taken.
Operator monitoring is typically provided in two forms: a control
room output, which drives the monitor speakers through a medium-power
external amplifier, and an amplified headphone output. A volume control is
provided for these signals.
MUTE SELECTORS.
There are other selectors that fulfill administrative functions within

the mixing job. The first is the mute selector, which is used to silence the
selected signal without altering the rest. This is achieved by disconnecting
the signal from the main bus and the monitor bus.
Muting can be applied to more than one channel.
SOLO SELECTORS.
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Another common control on consoles is the solo, which allows you to
mute all channels except those in which the corresponding selector button
has been pressed. In a certain sense it performs the inverse function of the
mute, and it should be noted that the solo selection prevails over the mute
selection on the same channel.
There are two types of solo: the pre fader solo (pre fader listen, PFL,
or also cue), which takes the channel signal before passing through the
fader, and the post fader solo (solo in place, SIP), which takes the signal as
it is going to be fed to the main bus for mixing. The single pre fader is used
as a guide during level adjustment of the channel input signal. The single
post fader, to isolate a certain channel as it will appear in the mix. It is
possible to select more than one.
VUMETERS.
The consoles have a measuring instrument called a vumeter. The

name comes from the acronym for Volume Units, that is, volume units.
There are actually three scales for VU meters in use by the various
console manufacturers, which consist of calling the levels –10 dBV, +4
dBu and 0 dBu respectively, 0 VU. In all cases, the reference used must be
verified in the user manual.
PRESENTATION OF THE INPUT CHANNELS.
The input channels have a quite characteristic presentation, although

there are many variants depending on the manufacturer and model.
As can be seen, the vertical arrangement of the controls, adjustments
and selectors approximately follows the succession of blocks through
which the signal passes.
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At the top we find the input signal level adjustment
potentiometer. Next is a selection button for a low-
frequency high-pass filter (50 Hz) to remove or reduce
very low-frequency components. Below the previous
section is the equalizer section. The example shows a
parametric equalizer in which the low cut frequency is
100 Hz, the high cut frequency is 10 KHz and the mid
frequency is adjustable, as well as its quality factor Q.
Below appears the level control section of the
auxiliary outputs, which in this example are 4. Next to
each level setting is a button that allows you to switch
between post fader and pre fader connections. This
provision is not always available.
Sometimes some auxes are pre-fader and some are
post-fader, other times some are switchable and some
are not.
Then we find the panning section, with some
addressing selectors. These switches allow you to direct
the panned signal to the left and right channels of the
main mix, to output groups 1 and 2, or to 3 and 4. The
solo and mute buttons are also found here. The last one
eliminates the signal coming from the corresponding
channel from the mix. The solo selector, on the other
hand, primarily affects the solo bus and the monitoring
signal for the control room, so it does not affect the main
mix.
Finally, there is the channel fader, that is, a sliding
potentiometer that allows you to adjust the level of the
signal corresponding to said channel for the purposes of
the main mix or group mixes. The fader is graduated in
dB.
Many manufacturers prefer to graduate the faders
of their consoles from 0 to 10, without such a scale
representing any type of standard measurement. The
same observation applies to the rest of the controls or
gain adjustment.
(except in the equalizers, where dB grading has been
imposed).
EXIT SECTION.
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This section includes the group faders, the main fader, the level
settings for the auxiliary sends and returns, the VU meters, the control
room monitoring volume adjustment, and a series of routing selectors
linked to the groups. , auxiliary returns and monitoring.
The vu meters are usually found at the top of this section. In general,
there is at least one VU meter for each group, and a stereo pair for the main
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mix, although in the most economical consoles a single stereo pair is
usually used, assignable to various outputs by means of selector buttons.
At the bottom, parallel to the input channel faders, the group and
main mix faders are located; each group has a pan setting.
In addition, each auxiliary mode has selectors that allow the
corresponding stereophonic signal to be input to the main bus and/or pairs
of group buses.
The headphone connection is also located in this part of the console,
which is accompanied by a volume control that affects both channels
equally.
Finally, a selector has been included that establishes whether the solo
mode is PFL (pre fader) or SIP (post fader).
It has also incorporated its own volume control for the solo, which
makes it possible not to alter the general monitoring level every time you
want to listen to an individual channel.
CONNECTIONS FOR TAPE RECORDER.
Many consoles have input and output connections for tape, cassette,
or DAT recorders. The tape in inputs, coming from the play outputs of the
recorder, in some cases act as supplementary input channels, and in other
cases simply as inputs for the control room monitoring system. The outputs
for the recorder (tape out) usually repeat the main outputs, which makes it
possible to record the result of a complete mix in DAT.
When you want to record in multitrack, you use the insertion sends of
the channels, interspersing if necessary some processors, such as external
equalizers, compressors or gates. Group submixes can also be recorded
using the group outputs.
Some consoles have direct post-fader outputs of the input channels,
which allow the signals to be recorded in multitrack after passing through
the filters, the equalizer and the channel fader. While this provides greater
flexibility, sometimes it may be more convenient to record and play back
via the insert connection, with the outputs of the multitrack recorder being
able to be sent via the insert return.
Proceeding with the mixing of the recorded material exactly as if they

were the original live signals.
CONSOLE SPECIFICATIONS.
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There are some obvious specifications, such as the number of input
channels, the number of auxiliary connections and groups. Even the block
diagram of the internal connection, which is not a specification in itself,
can inform us about the suitability from the connectivity point of view of a
certain model for the type of work to be performed with the console.
It is extremely important that the specification be provided attaching
the conditions under which it is measured or determined. It must be taken
into account that many commercially available consoles completely lack
this information.
DISTORTION.
The first specification is distortion. In general, total harmonic

distortion, THD, and in some cases intermodulation distortion, IMD, occur.
The minimum data required is the following:
1. the frequency of the sinusoidal test signal. Normally it is 1 KHz,

but it would be desirable to also have the distortion values at
other frequencies, for example 100 Hz and 10
KHz.
2. The point where the signal is injected. Generally it is the line
input of a channel, but it could be an auxiliary return or an insert
connection.
3. The location of the faders. Normally at their nominal point (0
dB), although sometimes it is stipulated that they are located in
“typical positions”, which is ambiguous.
4. The input and output level. It should be a high value like 15 or
20 dBu. In many cases the distortion is indicated for a nominal
output level, for example 4 dBu. This is very useful, since
distortion is a problem at high levels, and not at low levels.
NOISE.
The second specification of a console is noise. This can be specified

in several ways: as a noise voltage level at the output in dBu, as a
signal/noise ratio, as a level referred to the nominal level, as noise
equivalent to the input in dBu or dBm. Since a console has several outputs,
the specification may vary depending on which output it refers to. The
measurement conditions that you need to know, at a minimum, are:
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1. Output on which the noise is measured (can be a main output, a
group output, a direct channel output (post fader), an aux send,
an insert send, tec)
2. Position of the corresponding channel and output faders. The
noise when all channel faders are at their minimum is always
lower than when they are all at their nominal position, or,
worse still, at their maximum.
3. Position of the equalizer controls (they should be the central
positions, since otherwise the noise in a certain band could be
accentuated or reduced)
4. Frequency band of the filter used to measure noise. Typically,
it should cover the range from 20 Hz to 20 KHz.
OVERLOAD MARGIN.
The next specification is the headroom. This data is provided with

respect to the various console outputs. In general, the main outputs have the
greatest overload margin.
FREQUENCY RESPONSE.
The frequency response is defined in the same way as for other

equipment.
Within the measurement conditions it is important to indicate:
1. In what position are the equalizer controls located. Normally,
they should be flat (0dB)
2. Whether or not the high-pass and low-pass filters have been
interleaved.
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3. The signal level for which the measurement has been carried out,
for high frequency and high level signals, distortion appears due
to the fact that the mixing amplifier has a limit as to the speed
with which its output can vary. This implies that for high level
signals the upper cutoff frequency is reduced.
4. The type of input (line in, microphone in, aux return, etc.) and
output (aux send, main output, etc.)
The main disadvantage of an excessive frequency response is that

noise could be added to the mix buses that, despite being outside the
audible range, would unnecessarily increase the overall level of the mix.
INDICATORS.
The various light indicators are usually detailed, such as LEDs

indicating solo and mute, overload (clipping or saturation), power on,
phantom source connection, auxiliary send type, etc.
OTHER SPECIFICATIONS.
The remaining specifications are of an administrative nature, such as

dimensions, weight, type of mounting (rack, table), environmental
conditions of operation and storage (temperature and humidity range), type
of power supply and electrical energy consumption, etc. These data do not
provide anything new from a functional point of view, although they can
constitute important decision elements when purchasing a console.
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CONCLUSION.
Since all the signals present in a sound system sooner or later pass
through the console, its specifications have a decisive influence on the final
product: the mix. For this reason, special attention must be paid to these
specifications.
Finally, both for live and studio sound, care must be taken to
interconnect all the components of the system, since it is too easy to make
mistakes that considerably degrade the overall performance of the system.
CABLES AND CONNECTORS.
INTRODUCTION.
A cable is probably the least expensive item in a stereo system. 90%

of equipment failures are caused by cables, so if we worry about the quality
of microphones, amplifiers and speakers, we must also worry about the
quality of the cables (even if this means a lot of expense).
A high price does not guarantee a good product. There are big
differences between cables and connectors that look the same.
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We can ask ourselves the question about what type of cable to use:
simple, double, with or without mesh; with braided, wrapped or foil mesh.
This chapter specifies some functions and construction differences with
respect to various cables and connectors for professional use.
ELECTROSTATIC AND ELECTROMAGNETIC

INSULATION.
The insulation (mayan) of cables for line or microphone signal is

essential. The purpose of the mesh is to exclude electrostatic fields by
intercepting its charge and draining it to the ground so that it does not add
to the signal of the conductors.
The shorter the wavelength of the noise, the more easily it penetrates
the mesh, which is why the best possible wiring for permanent installations
is one that uses insulation based on a foil wrapped around the cable and not
a braided mesh.
Consequently, we can intuit that cables with braided or coiled mesh
are the ones most used to connect microphones and instruments in a
system. Cables with coiled mesh are the most flexible, but have a tendency
to open, which worsens their noise resistance.
The best way to prevent electrostatic and microphonic noise is to use
cables constructed of dialectically stable materials and tightly interlocked
cables well covered by rubber.
Magnetic fields cancel only with balanced lines and increasing the
physical distance from the source.
Ground loops also generate noise in the cable, which is induced by
current flow in the cable. In this case the only solution is to properly
ground the system.
PRINCIPLES OF BALANCED CIRCUITS.
In a balanced circuit the two conductors carry the same signal, but
with opposite polarities. In a balanced input, both conductors have the same
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potential difference with respect to ground and the input is designed to
recognize only the voltage difference between both conductors.
Then any noise or electrostatic interference which will have the same
polarity in both conductors will be canceled by the input circuit.
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CABLES WITH ONE OR TWO CONDUCTORS.
Single conductor and mesh cables are used for unbalanced circuits.
These unbalance balanced circuits. Two-conductor cables and mesh are
used for balanced circuits.
It is best to avoid two-conductor cables in unbalanced circuits
because this will simply increase the cable's capacitance (energy stored in
an electric field).
BLACK CABLES AND HORN WIRES.
We know that we should never use cables without cables for

microphones or instruments. These are used in speaker cables where the
signal is very high and the resistance in the speakers is very low.
The diameter of the conductors must be larger to withstand the high
currents involved.
The higher the current, the greater the magnetic field and
consequently the greater the sensitivity to low frequency loss caused by the
inductance of the cable. Speaker cables should preferably not be twisted,
because this increases inductance.
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MULTIPAIR AUDIO CABLES (SNAKES).
When we have an audio console placed in a remote position from the

sound source, a large number of microphone cables must travel a relatively
long distance. If we use individual cables in this case, the result would be
expensive, cumbersome, etc. In these cases we use the so-called Snakes.
The Snakes end on the console side with an XLR connector and on
the other side, they can end in a box with a chassis XLR connector.
They usually not only bring information to the console, but also bring
information from the console to the remote point (like when we bring the L
and R master outputs of the console to the amplifiers that are on stage).
When we have cables that carry different types of signals
(microphone, line, speakers and AC) running in parallel for long distances,
the risk of Cross Talk is very great, so we must try to separate them as far
away as possible. It is preferable that if they meet at a point it is
perpendicular. If they must necessarily go together, they must be grouped
and tied separately.
A Snake must have tension reinforcement both on the side of the box
and on each of the XLR connectors on the other end.
Snakes must be treated with care, avoid sharp kinks, and carefully
coiled and uncoiled, as a broken cable on a Snake is practically irreparable.
The use of special reels is recommended, which increase the durability of
the Snake, and speed up its installation and removal.
CONNECTORS.
A connector must be: easy to use, difficult to accidentally disconnect,

and must not: introduce resistance into the circuit or allow interference.
If a system is installed with permanent intentions, and will not be
constantly disconnected or moved, then the best option is not to use
connectors, but to make direct contacts using screws in the chassis of the
devices. This type of connection has minimal resistance, and does not cause
resistance due to aging of the metal of the connectors, in addition, it is
difficult to disconnect accidentally.
Every time a Plug is inserted into a Jack, some resistance is added to
the system, and although this is little, with the passage of time, dust and
corrosion this resistance increases. When connectors are joined and
unjoined regularly, they tend to self-clean, preventing resistance from
corrosion or dust.
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TELEPHONE PLUG.
These connectors are easy to attach to the cable and relatively

inexpensive, except in the case of the pure bronze military version which is
very expensive.
There are two types of Plugs, the TRS and the TS, where T is the tip
of the Plug, R is the ring or ring and S is the sleeve or sleeve. The TS
versions are used for unbalanced circuits and the TRS versions for balanced
or stereo circuits (which is why they are also called Stereo Plug).
The most popular version of the Plug is the ¼ inch (6.29 mm) but
there are other versions such as the Bantam TT, 0.173 inches (commonly
used in studio patches).
The ⅛ inch mini Plug (commonly used in laptop headphones) and the
1/16 inch sub mini Plug (used in micro cassette recorders and 6 to 12 volt
DC adapters).
One of the disadvantages of the Plug is that it generates noise when it
is connected, this is caused when the tip of the Plug touches the chassis of
the Jack and an electric shock occurs.
Another problem is that the shell can break if it is made of plastic, so
it is preferable to use those with a metal shell.
The Plug is prone to damage if pushed sideways while connecting,

which can be avoided with 90 degree connectors.
The Plugs do not have a safety system, so they can be accidentally
disconnected.
There are very advanced high continuity versions developed by
manufacturers such as Neutrik and Switchcraft, but they are three times
more expensive than conventional models.
PHONO CONNECTORS (RCA).
Invented by the Radio Corporation of America, they were originally

developed for internal connections in radios and televisions. They became
very popular in home audio equipment because they were economical, easy
to connect and very small in diameter.
This familiar connector has a pin in the center of a shell, which is
why they are also called Plug pin.
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Wide cable cannot be used with this type of connector because it has
a very small diameter.
These connectors generate great resistance over time while
connected.
If you have to use this type of connector for professional use, they
must always be gold-plated and must be rotated eventually to clean the
connection.
XLR CONNECTORS.
These connectors were introduced to the market by Cannon (which is

why they are called that way) and there are multiple types. The one used in
professional audio is the XLR 3, which has three connection pins and an
external metal shell which can be connected or not to the cable gland.
The XLRs have a lock that is activated when they are connected, and
you have to press a piece of the connector to be able to disconnect them.
Pin 1 is always connected to ground so that when connected it does not
produce an electric shock.
These connectors support large diameter cables and have excellent
anti-strain protection (both outward and sideways). The pins are difficult to
touch, so the contacts cannot easily become dirty.
BANANA CONNECTOR.
This type of connector is exclusively used for speaker cables, because

it easily supports high voltage flows without presenting greater resistance.
It is very easy to disconnect and generally has a side flap that indicates the
negative of the connection.
The Jack of this type of connector generally has a side hole through
which the cable can be inserted and can then be tightened with a plastic nut.
This type of connection is recommended for permanent installation.
SPEACKON EP4 AND EP8 CONNECTOR.
Designed by Neutrik, it is a connector that is beginning to become the

industrial standard for connecting speakers.
It is very robust and resistant to pressure. It has a voltage lock based
on the fact that it must be inserted into the Jack and then rotated 45 degrees
for contact to be made, this avoids any possibility of accidentally
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disconnecting it. There are versions with 4 and 8 contacts (EP4 and EP8),
which makes it ideal for bi-amplified speakers. It is definitely the best
option for high-consumption professional speakers, since it supports very
large diameter cables (#10).
DIRECT BOXES.
When it comes to amplifying instruments such as keyboards, the

direct signal is obtained from the line output of the instrument, but the
impedance of these devices and the output level are very high, this can
damage the output circuit of the instrument or affect the response. signal
frequency through the cable capacitance and the distortion generated by the
circuit.
In this case we use a transformer called a direct box (DI), which
lowers the impedance, the output level (from line to microphone) and
provides effective isolation against ground loops and other electromagnetic
noise.
MONITORING SYSTEM.
The platform from which praise is directed and preaching is delivered

is the forum or platform. This is where musicians flow in their gifts. But
there is an extremely important technical factor: the monitors. If there are
problems, the musicians will not perform well, the preacher will probably
wear out his voice, since there is no good frontal reference.
It is recommended that the main speaker system be placed in front of
the microphones to avoid feedback (feed back).
There are two ways to send the referral signal to the forum. The first
is to use an auxiliary output from the mixer to a graphic equalizer
(preferably 30 bands) then to an amplifier and at the end of the chain place
the monitors, with the speaker pointing in front of the musician or speaker,
forming a equilateral triangle.
For musicians and singers, other monitors need to be placed. Their
placement must be strategic so that they can be shared. The thing to always
remember is that these are with the speaker pointed at the back of the
microphones to avoid a chain of signal feedback. This is the way used at
the amateur level, or when there is a low budget.
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The second way is how it is used at a professional level. This consists
of having a console or monitor mixer, which is capable of handling several
mixes output to monitors (from 6 or more). This must have the same
number of channels or inputs as the room one.
It also has equalization per channel and insertion points (in-out) to be
able to insert processors (compressors, reverb, graphic equalizers, etc.). To
feed the signal to the monitor console you need a Snake or Medusa Splitter,
Split the signal in two), that is, the Snake box will have two output cables
for each input, and one output is connected to the monitor console. room
and another connects to the monitor console.
With this we will obtain total independence from the main console.
This means that we will not affect the equalization of the main system, and
we can give each musician, through an amplifier with a monitor, the mix
that best suits him, for example: the pianist can be given a mix in which his
main volume is the piano, in the background the singer's voice. For the
drummer it will be a different mix with greater sound power, and so in each
monitor, the objective is to provide comfort when playing and not to have it
be a competition of who can be heard more.
The monitor engineer is in charge of controlling the mixes and doing

everything possible so that everyone gets what they want. This makes the
job of the monitor engineer very difficult, since he practically must be able
to produce such a number of mixes that we can easily say that he is giving
6 or more concerts at the same time for the most demanding listeners, who
are singers and musicians.
Therefore, a monitor engineer must meet a series of conditions and
virtues that qualify him for this job:
1. Be well organized to prepare the assembly and placement of
each monitor.
2. Have a kind character and infinite patience to overcome
unforeseen events, without losing technical and psychological
control.
3. Possess a very high capacity for visual attention to capture the
signals that different musicians send you while they play,
asking you to raise or lower an instrument.
4. Have excellent reflexes to react before the dreaded feed back
occurs that the singer is about to cause by pointing his
microphone at his monitor, while leaning or lowering his hands.
5. Have a good memory to locate, in fractions of a second, the
button that needs to be moved instantly.
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6. Finally, he must remain lucid, show great fortitude to transmit
confidence to the musicians, know how to admit his mistakes
and correct them immediately and have a very high capacity for
analysis of what is happening in all stages of the electroacoustic
chain to prevent, predict and avoid audible disasters. on the
forum or platform, which could overshadow the work of the
praise group.
GLOSSARY.
TO
A: Abbreviation of ampere.
Sound absorption: (sound absorption) Action carried out by every
surface to a greater or lesser degree, absorbing and eliminating part of the
sound energy that affects it.
Feedback: (feed back) Electroacoustic feedback phenomenon
between a speaker and a microphone that gives rise to self-sustained hisses
or hums.
Chord: (chord) In music, any superposition of two or more sounds.
Acoustics: (Acoustics) 1. discipline that studies sound in its various
aspects. 2. Set of characteristics of an environment that determine how
sound behaves in it.
Impedance adaptation: (impedance matching) 1. Broadly speaking,
any relationship between the impedances of a source and a load that
maximizes the magnitude of the signal is most important for an application.
For example, if maximum voltage is desired, the adaptation implies that the
load impedance is high with respect to that of the source. 2. Specifically,
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the relationship between the impedances of a source and the load that
maximize power transfer.
AES/EBU: Two-channel digital communication protocol, using XLR
connectors.
Acoustic insulation: (acoustical insulation) Action carried out by any
dividing partition between two environments, preventing to a greater or
lesser extent the passage of acoustic energy from one side to the other (it
may include inaudible, ultrasonic and subsonic waves).
Sound insulation: (sound insulation, soundproofing) Action carried
out by any dividing partition between two environments, preventing to a
greater or lesser extent the passage of sound energy from one side to the
other.
Algorithm: (algorithm) 1. Calculation procedure (especially used in
digital signal processing) to perform a specific function (filter, add effects,
delay, etc.). 2. In synthesis of sounds by frequency modulation, a certain
structure of interconnection of modulators.
Alias: (alias) See alias frequencies.
Aliasing: A type of distortion that occurs when the sampling theorem

is not satisfied (that is, when in a signal to be sampled there are frequencies
beyond half the sampling frequency). It consists of the appearance of alias
frequencies, that is, frequencies that do not exist in the original signal.
Speaker: (loudspeaker) Transducer that converts electrical signals
(generally power) into sound.
Direct radiation speaker: (direct radiator) Speaker whose
diaphragm is directly exposed without a horn or other coupler.
Height: (pitch) Sensation of greater or lesser gravity or sharpness of
the sound.
Ambience: (ambience) Auditory sensation through which it is
possible to have an idea of the characteristics of an environment: size,
position of the walls, etc.
Amper: Unit of electrical current. It is abbreviated to A.
Amplifier: (amplifier) Device to increase the level of an electrical
signal.
Controlled amplifier: (controlled amplifier) Amplifier whose gain can be
controlled with an electrical variable (for example a voltage).
Amplitude: (amplitude) Maximum or peak value of a signal.
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Spectrum analysis: (spectrum analysis) Measurement whose purpose
is to determine the spectrum of a sound or signal, that is, the frequencies it
contains and their respective amplitudes.
Spectrum analyzer: (spectrum analyzer) Device to perform
spectrum analysis.
Analog: (analog) It is said of any device that processes a physical
magnitude using an analogy that uses another physical magnitude. For
example, it processes sound by handling electrical signals analogous to the
sound to be processed.
Analog / a: (analog) 1. Quality of a physical variable to evolve over
time in the same way as another. 2. System, device or signal whose
operation or temporal evolution is the same as that of another.
Bandwidth: (bandwidth) Difference between the maximum and
minimum frequencies that a band-pass filter or any device allows to pass
through.
Harmonic: (harmonic) Harmonic sound.
Attack: (attack) 1. In a sound, the initial stage during which the
amplitude of the sound increases until it reaches a maximum value. 2. In a
dynamics processor, effects processor, etc.; stage during which one goes
from the non-active situation to the active situation.
ATRAC: Minidisc psychoacoustic compression system.

Audio: 1. Audible frequency range. 2. Techniques and technologies
applied to sound processing and reproduction.
Digital audio: (digital audio) A series of techniques and technologies
for the processing, storage and reproduction of audio signals based on the
conversion of the electrical signal into numbers.
Audio frequencies: (audio frequencies) Audible frequencies
(between 20 Hz and 20 KHz).
Baffle: Acoustic box.

Closed baffle: (closed baffle) Hermetic acoustic box.
Ventilated baffle: (open baffle) Acoustic box with a hole or tube that
communicates its interior with the environment.
Filter bank: (filter bank) A set of filters (usually digital) that
subdivide an entire range of frequencies into subbands.
Band: (band) An interval of frequencies.
Audio band: (audio frequency band) Frequency interval between 20
Hz and 20 KHz.
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Beat: (beat) Variation in the amplitude of the wave resulting from
superimposing two signals of similar frequencies.
Battimento: (beat) Italian word that means shake or pulsation.
Biamplification: (biamplification) Technique by which the signal is
divided into its low and high frequency components before being amplified
by two amplifiers that feed the low and high frequency speakers
respectively.
Binary: (binary) It is said of any magnitude that assumes only two
possible values.
Binding post: A type of connector typically used in amplifier output
connections, it consists of press-fit cable glands.
Bit: Binary digit.
Shielding: (shielding, screening) Element used to prevent the exit
(entry) of electric or magnetic fields from (a) a device, cable, etc.
Electrostatic shielding: (electrostatic shielding) Shielding that
prevents the emission of electric fields from a cable, or their entry into the
cable. Normally it is a mesh that covers or wraps the insulated conductor
that carries the signal, electrical energy, etc. Sometimes it is a metal tape
(for example aluminum) wrapped around it.
Magnetic shielding: (magnetic shielding) Shielding that prevents

magnetic fields from a transformer, motor, etc. from transcending. In
general it is a cover made of ferromagnetic material.
Telescopic shields: (telescopic shields) In a balanced line, shield
connected to the circuit ground only on one side. It is used to prevent
ground loops.
Coil: (coil) Winding of enameled wire generally around a core of
ferromagnetic material. When passed through by an electric current, it
creates a magnetic field.
Horn : Acoustic device to adapt the high acoustic impedance of a
compression exciter to the low acoustic impedance of air, and thus improve
performance.
Ground loop: (ground loop) A closed path that is generated between
a conductor connected at more than one point to the physical ground, and
to the ground itself. It behaves like an antenna, picking up especially low-
frequency noises that produce hum in the system.
Bus: Bus line. In a console, a set of lines that transport various
signals to the inputs of an adder or mixer. In a digital system, a set of lines,
each of which contains one bit.
Byte: 8 bits.
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C: Abbreviation for capacity.
Head: (head) Device that contains the electromagnetic transducers
that allow recording, playing or erasing a magnetic tape.
Erasing head: (erasing head) Head by means of which every vestige
of a signal previously recorded on a magnetic tape is eliminated.
Recording head: (recording head) Head by means of which a signal
is recorded on tape.
Reproduction head: (playing head) Head through which a signal
previously recorded on magnetic tape is reproduced.
Rotary head: (rotary head) Head mounted on a rotating drum that is
used in digital video and audio tapes to increase the relative speed between
the head and the tape, extending the bandwidth of the signal to be recorded.
Cable: Electrical conductor insulated by a plastic covering, generally

formed internally by fine strands of copper.
Shielded cable: (shielded cable) Cable that is inside a copper or
tinned copper sheath that completely covers it (or a coiled metal tape),
isolating it from electrostatic and electromagnetic interference. Externally,
the mesh is covered by a plastic wrap.
Coaxial cable: (coaxial cable) Shielded cable formed by an insulated
conductive tube inside which there is another insulated conductor.
Acoustic box: (baffle) Enclosure to avoid interference between the
compression wave (front) and the decompression wave (rear) of a direct
radiation speaker.
Direct box: A device that uses an impedance adapter and a balancer
to allow the connection of certain unbalanced, high-impedance signal
sources (electric guitar, for example) to a microphone input.
Sound field: (sound field) Way in which sound varies or is
distributed in time and space.
Near field: (near field) Direct field.
Direct field: (direct field) Part of the sound field formed by all those
waves or wave fronts that have not yet experienced any reflection.
Electric field: (electric field) Action at a distance of an electric
charge, consisting of the attraction of charges of different signs or the
repulsion of those of the same sign.
Far field: (far field) Reverberant field.
Free field: (free field) Sound field in which there are no obstacles for
sound waves.
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Magnetic field: (magnetic field) Action at a distance created by an
electric current or by a magnetized material consisting of the appearance of
forces on mobile charges or on other ferromagnetic materials.
Reverberant field: (reverberant field) Part of the sound field formed
by those waves that have suffered at least one reflection.
Channel: (channel) 1. In a console, each of the subsystems that
process independent signals. 2. In a MIDI system, each of the parts receives
independent messages.
Cannon: A type of balanced connector typically used for microphone
or line level signals. Also called XLR.
Capacity: (capacity) Magnitude C associated with a capacitor equal

to the accumulated electrical charge divided by the applied voltage: C = Q /
V. It is measured in farads, F.
Capacitor: (capacitor) Electrical component formed by two
conductive plates facing each other and separated by air or other insulating
material. Each board is connected to a terminal. When a voltage is applied
between its terminals, it accumulates opposite charges between its plates.
Cardioid: (cardioids) Type of microphone whose directional
diaphragm is shaped like a heart, with maximum sensitivity forward and
minimum sensitivity backward.
Load: 1. (load) Any component, device, equipment, etc. That it
connects to a source, be it signal or power. 2. (charge) Electrical charge.
Electric charge: (electric charge) Basic physical quantity of
electricity, abbreviated Q. It can be positive or negative. Two bodies with
opposite charges attract each other, and bodies with equal charges repel
each other.
CD: Compact Disc (English acronym). Optical technology (laser)
disc for high quality digital audio playback.
CD-ROM: ROM memory that uses a compact disc as a medium. It is
used in computing to store programs or data, including sound files in
formats. WAV.VOC.MP3, etc.
Chorus: Chorus. Effect consisting of a vibrato superimposed on the
dry signal.
Cycle: (cycle) Referring to a periodic or quasiperiodic wave, each
portion that repeats.
Hysteresis cycle: (hysterics cycle) Closed loop that illustrates the
relationship between the magnetizing force and the magnetic field when the
former varies alternately between a maximum and a minimum.
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Electric circuit: (electric circuit) interconnection of electrical components
forming one or more closed loops.
Coverage of a source: Angle that a sound source, such as a speaker,
a horn, etc., covers on either side of its axis of symmetry.
Sound absorption coefficient: Fraction of the sound energy incident
on a material or surface structure that is absorbed (sometimes expressed as
a percentage.
Compression: (compression) Action of reducing the dynamic range
of a signal.
Data compression: (data compression) Reduction of the digital

information that needs to be stored compared to its original version, in
order to save storage space. In general, it is based on taking advantage of
some characteristics of the signal to be stored (for example, eliminating
inaudible components of the sound).
Psychoacoustic compression: Compression of digital audio data that
takes advantage of the masking effect to reduce the required information.
Compressor: (compressor) Equipment that receives a signal and
reduces its dynamic range based on parameters adjusted by the user.
Gate: (gate) Device that closes the input when it does not reach a
level comparable to the known minimum of the signal, and therefore it is
presumed to be noise. Improves the apparent dynamic range of the signal,
“cleaning up” silences.
Capacitor: (condenser) Capacitor.
Conductor: (conductor) Any material that easily allows the
circulation of electric current. Metals are generally good conductors,
especially silver, copper and gold.
Connectivity: Quality of a system that allows variety and flexibility
in the connections between its constituent parts.
Connector: (connector) Device that allows a safe and reliable
connection between two conductors. Some types of connectors that are
very popular in Audio are RCA, TRS, TS, XLR.
Ground connection: (ground) Connection between the reference or
circuit ground and the ground or metal chassis of a device.
Ground connection: (safety ground) Connection between the mass
or metal chassis of a piece of equipment and the physical ground. It is used
for safety reasons, and in a well done installation it is done through the
center prong of a three-prong plug.
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Balanced connection: (balanced connection) Type of connection of
three conductors, one of them connected to ground and the others with
opposite voltages. It is used to increase the noise immunity of low-level
signals, such as microphones.
Insertion connection: (insert) In a console, connection to interleave a
serial processor.
Console: (mixing console) Generic name of a device with several
input and output channels in which the mixing is carried out in different
proportions, and after passing through various processors, of various
signals from microphones, synthesizers, etc. .
Analog/digital converter: (analog to digital converter) Device that

converts values of an electrical signal into numbers, generally binary.
Digital/analog converter: (digital to analog converter) Device that
converts numbers, generally binary, into values of voltage or electric
current.
Consonance: (consonance) Auditory sensation evoked by two
superimposed sounds that do not cause pulsations.
Chorus: (chorus) Effect that consists of a vibrato superimposed on
the dry signal.
Current: (current) Electric current.
Electric current: (electric current) 1. Circulation of electric charges
through a conductor. 2. Due to abuse of language, intensity of electric
current.
Eddy currents: (Foucault currents) Swirling currents that occur in a
conductive ferromagnetic material in the presence of a varying magnetic
field. They dissipate power and reduce the effective magnetic field at high
frequencies.
Cross-over: Frequency divider filter.
Passive cross-over: (passive cross-over) Frequency divider included
in multi-way speakers (two or three speakers) that uses coil filters and
capacitors.
Active cross-over: (active cross-over) Active frequency divider
(filters contain low-level amplifiers and capacitors, but not inductances) for
bi- or multi-amplification.
Frame: (frame) In digital audio, a packet of bits that is taken by the
encoders of error correction systems.
Quasiperiodic / a: (quasi periodic) Said of a phenomenon, signal,
etc. That T is repeated from time to time with small variations, for example
the sound of a piano, which changes very little from one cycle to the next.
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Anechoic wedges: Surface finish in the form of wedges of an
acoustic absorbing material in order to increase the effective surface and
therefore its absorption coefficient.
Transfer curve: (transfer graph) Graphic representation of the
relationship between the input and output of a system, device, etc.
Fletcher and Munson curves: /Fletcher and Munson graphs)
Graphic representation of the combinations of frequency and sound
pressure level that produce the same sensation of loudness as each other
(see loudness level).
DAT: (digital audio tape) Digital audio tape (acronym in English). Tape
format somewhat smaller than the traditional cassette for recording high-
quality digital audio. Suitable for masters.
dB: Abbreviation for decibel.
dBA: Sound level unit, that is, the equivalent sound pressure level
after amplifying a filter that behaves similarly to hearing.
dBm: Power level unit referred to 1 m W.
dBu: Voltage level unit referred to 0.775 V.
dBV: unit of voltage level referred to 1 V.
dbx: Supplementary noise reduction system for use in magnetic
recording. It acts by linearly compressing by a factor of 2 in decibels
during recording, and then expanding by the same factor during playback.
DCC: (digital compact cassette) Digital compact cassette (acronym in
English). Tape format similar to traditional cassette that allows recording
and playback of digital audio and also playback of traditional analog
cassettes.
Decibel: conventional unit assigned to the logarithmic expression of a
magnitude or relationship of magnitudes.
Decibelimeter: (sound level meter) Sound level meter.
Phase shift: Phase difference between two signals.
Spectral density: Magnitude used for continuous spectra, rather than
the amplitude of the individual components, which would be too small. It is
proportional to the power per unit of bandwidth.
Demagnetization: (demagnetization) Decreasing cyclic process in
which the remaining magnetism is finally reduced to 0.
Error detection: (error detection) Operation carried out in digital
recording or transmission systems by which the presence of an error is
recognized, even when it cannot be corrected.
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Crosstalk: (croostalk) Channel separation.
Diaphragm: (diaphragm) 1. In a microphone, a thin sheet that vibrates
in accordance with the sound wave. 2. In a speaker, a conical or dome-
shaped piston that, when vibrated by the action of the exciter (motor
portion), generates sound.
Block diagram: (block diagram) Simplified diagram that represents

the constituent parts of a system and their interconnections.
Directional diagram: (directional pattern) Diagram that represents
the directional characteristics of a device (microphone, speaker, antenna,
etc.).
Polar diagram: (polar pattern) Type of diagram used to draw the
directional diaphragm, formed by concentric circles graduated in dB and
radial line segments.
Digital: Said to be any system or device that operates by processing
numbers, which may or may not represent specific physical magnitudes.
For example, a computer processing numbers that in some cases represent
symbols (word processor) and in other cases signals (sound card).
Directional: (directional) Characteristic of some devices
(microphones, speakers) of presenting a marked directionality.
Directionality: (directionality) Characteristic of sound in terms of the
direction from which it comes or towards which its source is located.
Addressing: (routing) In a console, selection (using a keypad) of the
bus or buses to which the signal from an input channel or an auxiliary
return will be directed.
Hard disk: (hard disk) Computer disk with a large capacity for
storing digital information. It is also used to record digital audio.
Dissonance: (dissonance) Auditory sensation evoked by two
superimposed sounds that cause pulsations.
Distortion: Alteration of the waveform of a signal.
Harmonic distortion: Distortion that occurs on a sinusoidal signal,
consisting of the appearance of harmonics.
Distortor: (distortion, overdrive) An effect, usually for electric
guitar, that distorts the signal by clipping.
Phase distortion: (phase distortion) Modification of the phase
relationships between the harmonics or partials of a signal. In monophonic
audio signals, phase distortion is not important, because the ear is
insensitive to absolute phase. In stereophonic signals, non-identical phase
distortions in both channels can alter the spatial image.
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Intermodulation distortion: (intermodulation distortion) distortion
that occurs when the sum of two sinusoidal signals of different frequencies
is applied to the input of a device, consists of the appearance to a greater or
lesser extent of all the additions and subtractions between the harmonics
(multiples) of both frequencies.
Frequency divider: (cross-over) Filter that separates the low

frequency and high frequency components to send them respectively to low
and high frequency speakers, or to amplifiers that will deliver power to low
and high frequency speakers.
Dither: Low-level noise (typically equivalent to an error less than
one bit) added to an analog signal prior to its sampling and digitization.
Qualitatively improves digitizing noise, eliminating distortion-like
components that occur in low-level signals.
Voltage divider: (voltage divider) Connection that results from
applying a real voltage source (that is, one that has a certain non-zero
internal resistance) to a load, the consequence of which is that the voltage
actually applied to the load is reduced.
DNR: (abbreviation in English for Dynamic Noise Reduction
System) Non-complementary noise reduction system. It operates during
playback by reducing the bandwidth of a low-pass filter when there is no
high-frequency signal, preventing the passage of non-maskable high-
frequency tape noise.
Dolby: Supplementary noise reduction system used in magnetic
recording. It works by compressing the treble during recording, and
expanding it during playback. There are three variants: Dolby A
(professional), Dolby B (consumer) and Dolby C (intermediate).
Driver: Driver of a loudspeaker, especially (but not only) of high
frequency.
AND
Echo: (echo) Reflection of sound that takes more than 100 milliseconds to
return to the source.
Equalization: (equalization) Action of correcting or compensating
the frequency response of a system to make it flatter.
Equalizer: (equalizer) Complex filter made up of several sections or
bands, in each of which the gain or attenuation can be adjusted as required
within certain margins.
138
Band equalizer: An equalizer that divides the frequency spectrum
into logarithmically equal bands (for example octaves or thirds of an
octave).
Graphic equalizer: (graphic equalizer) A particular type of band
equalizer in which the controls are vertical (sliders) and parallel, so that
their position gives a clear graphic idea of the frequency response of the
equalizer.
Paragraphic equalizer: (Para graphic equalizer) A sophisticated
equalizer that combines the multiplicity of bands of the graphics with the
versatility of adjusting each of them of the parametrics.
Parametric equalizer: A general equalizer with a few bands in
which, in addition to the gain or attenuation, the center frequency of each
band and its bandwidth or, equivalently, its Q, can be continuously
adjusted.
Semiparametric equalizer: An equalizer in which it is possible to
continuously vary the center frequency of the band but not its Q.
Effect: The result of any processing of an audio signal other than
transduction, amplification, compression, and filtering for corrective
purposes (equalization).
Antenna effect: (electrostatic pickup) Capture of electrical noise by a
cable, by capacitive or electromagnetic coupling.
Doppler effect: Apparent increase in frequency of an approaching
sound source or apparent decrease in frequency of a receding source.
Hass effect: (hass effect) Effect obtained by applying two short
pulses to both ears through headphones, with a delay between one and the
other that gradually increases. Below 0.6 ms, the sound image corresponds
to a single source that moves toward the ear excited first. Then the source
remains close to said ear but widens. Above 35 ms the sound image
corresponds to two sources.
Electret: a type of pre-polarized capacitive microphone, that is, it
does not require power between its plates. Usually they do require power
for the impedance reduction amplifier.
Energy: (energy) The most important physical quantity. It is a
magnitude associated with various that has the particularity of being
conserved and transformed. It assumes various forms. For example, in a
resistor, electrical energy is transformed into an equivalent amount of
thermal energy (heat); In a speaker, 1% to 15% of the electrical energy is
transformed into sound energy and the rest into heat, etc.
Acoustic energy: (acoustic energy) A form of mechanical energy
related to vibrations of air or other media.
Kinetic energy: (kinetic energy) Energy associated with the
movement of objects.
139
Electrical energy: (electric energy) Energy accumulated or
transported through electrical charges.
Mechanical energy: (mechanical energy) Energy associated with
movement and elasticity.
Potential energy: (potential energy) Energy capable of being

transformed into movement, for example the energy that accumulates by
sharing a spring or air.
Sound energy: (sound energy) Acoustic energy related to signals
with frequencies between 20 Hz and 20 kHz.
Thermal energy: (thermal energy) Energy related to heat. When you
heat an object, it gains thermal energy.
Masking: (masking) Psychoacoustic phenomenon by which a weak
tone becomes inaudible in the presence of another more intense tone close
in frequency.
Air gap: Small slot that separates two poles of a magnetic device
such as a recording head.
Send: (send) Any signal line that leaves a console to an external
device.
Auxiliary send: (auxiliary send) In a console, signal output from one
of the auxiliary mixers. It is used as a signal for monitoring or for parallel
effects.
Envelope: (envelope) Relatively slow evolution of some parameter of
a sound, typically its amplitude or level.
Spatiality: (spatiality) Characteristics of sound related to its
interaction with the space in which it is confined.
Specification: (specifications) Set of technical data on the operation
and other characteristics of the equipment.
Spectrum: (spectrum) Amplitudes of the various sinusoidal
components of a sound (and in some applications, also the phases)
Harmonic spectrum: (Harmonic spectrum) A discrete spectrum in
which all frequencies are multiples of a fundamental. It corresponds to
periodic signals.
Continuous spectrum: (continuous spectrum) A spectrum in which
the frequency components are so close to each other that it is not possible
to discriminate them.
Discrete spectrum: A spectrum in which the frequencies present in the
signal are clearly distinguishable.
Spectrogram: (spectrogram) 1. Plot with the frequency on the horizontal
axis and the spectrum values on the vertical axis. 2. Graph with frequency
on the vertical axis, time on the horizontal axis and intensity represented by
different colors or different shades of gray.
140
Inharmonic spectrum: A discrete spectrum whose frequencies are
not harmonically related to each other (i.e., they are not multiples of any
fundamental frequency)
Mixed spectrum: (mixed spectrum) A spectrum in which all frequencies

exist, but some stand out noticeably.
Structure: (structure) In relation to any device, equipment or system,
the way in which the different parts that make it up are organized
internally.
Gain structure: (gain structure) In a console, description, generally
graphic, of the gain adjustment ranges of the different stages that constitute
it.
Stage: (stage) In an electronic signal processor, each of the parts
through which the signal passes successively between the input and the
output.
Exciter: (driver) 1. In a loudspeaker, the driving unit, which can be
electrodynamic, electrostatic or piezoelectric. 2. In an amplifier, the stage
before the power stage.
Compression driver: (compression driver) Tweeter driver that
generates a very high sound pressure, which is reduced through the use of
horn-shaped adapters.
Expander: (expander) Device that increases the dynamic range of the
signal it receives.
Damping factor: Ratio between the load impedance of an amplifier and its
internal impedance.
Fader: (English word, pronounced feíder) Attenuator. It is used in
consoles as a level adjustment that is assigned to a signal in a mix.
Phase: (phase) Angle between the peaks of two periodic signals of
equal frequency, taking the equivalence 1 period ≡ 360º
Ferromagnetism: Branch of magnetism that studies the magnetic
properties of some materials.
Ferromagnetic: (ferromagnetic) Property of certain materials, such
as iron, neodymium, etc., to present very marked magnetic properties.
Optical fiber: (optical fiber) Glass fiber (coated with an opaque
sheath) through which digital information is transmitted by means of light
pulses. It is extremely immune to noise and allows high switching and
141
therefore transmission frequencies. It is applied to interconnect digital
audio equipment.
Figure of eight: (figure of eight) Type of bidirectional microphone
that has maximum sensitivity forward and backward and zero sensitivity to
the sides.
Filter: (filter) Device capable of selecting parts of a signal based on

their frequencies.
Anti-aliasing filter: (anti Aliasing filter) Low-pass filter that
eliminates all frequencies from a signal to be sampled that do not comply
with the sampling theorem.
Smoothing filter: Filter used in the reconstruction of a digital signal
to reduce digitization noise.
Digital filter: (digital filter) Filter obtained by performing numerical
operations on the previous samples of a digital signal, being able to obtain
a frequency response similar to that of any analog filter. It has the
advantage that it is possible to significantly reduce phase distortion.
“Notch” filter: (notch filter) Filter that blocks the passage of a
certain frequency. It is used, for example, to eliminate hum from the 220 V
line frequency.
High pass filter: (high pass filter) Filter that allows frequencies
greater than a cutoff frequency to pass and prevents lower ones from
passing.
Low pass filter: (low pass filter) Filter that allows frequencies lower
than a cutoff frequency to pass and prevents higher ones from passing.
Band pass filter: (band pass filter) Filter that allows the frequencies
between a lower cut-off frequency and a higher cut-off frequency to pass,
and prevents the rest from passing.
Comb filter: (comb filter) Filter that eliminates all frequencies
multiples of another frequency.
Flanger: Effect consisting of adding two signals with delays
modulated by push-pull signals, characterized by the suppression of some
frequencies that vary periodically with the modulation frequency.
Phoneme: (phoneme) Minimum acoustic portion of spoken language
with its own identity. For example, vowels and consonants.
Waveform: (waveform) Shape of the oscillogram of a signal.
Formants: (formants) Characteristic resonances of a voice or
instrument, which determine its timbre.
Frequency: (frequency) It is a periodic signal, number of cycles or
periods per unit of time.
Alias frequencies: (alias frequency) Frequencies that appear when
the sampling theorem does not hold (see Aliasing, sampling theorem)
142
Center frequency: (center frequency) Geometric mean between the
upper and lower cut-off frequencies of a band-pass filter (that is, the square
root of the product of both)
Cutoff frequency: (cutoff frequency) It is a filter, limiting frequency
between a pass band and a cutoff band.
Sampling frequency: Frequency with which samples are taken in a
sampling process. Also called sampling rate.
Fundamental frequency: (fundamental frequency) In a periodic
signal, simply its frequency, that is, the frequency of its first harmonic.
Wave front: Part of the wave in which the pressure is maximum.
Power source: (power source) Voltage source responsible for
providing the electrical energy that a circuit requires to function.
Voltage source: (voltage generator) Electrical device that maintains a
voltage independent of the load that is connected between its two terminals.
Real voltage source: (real voltage generator) Voltage source that has
an internal resistance or impedance, and that consequently varies its voltage
as the load resistance varies.
Signal source: (signal generator) Any signal generator or device
capable of delivering a signal to other devices.
Phantom power: (phantom power) Voltage source used to power the
capsule and impedance reduction amplifier of capacitive microphones
through the same cables that carry the signal.
Sound source: (sound source) Device, object, person, etc. That
generates sound.
Fundamental: (fundamental) In a periodic signal, its first harmonic.
Coercive force: Magnetizing force opposite a given state of
magnetization required to bring the magnetic field to 0.
Electromotive force: (electromotive force) Voltage that appears in a
circuit when the magnetic field that passes through said circuit varies.
Magnetizing force: (magnetizing field) Magnitude proportional to
the magnetic field in air created by a current or a magnetic pole.
Leakage: Spurious path for the circulation of electrical currents
generated by some failure in the insulation, dirt, humidity, etc.
Dynamic range: (dynamic range) Dynamic range.

Gain: (gain) Relationship between the output signal and the input
signal in an amplifier, often expressed in dB. In this case it is the difference
between the output and input levels.
143
Throat: (throat) Exit of a compression exciter, where the horn is
applied.
Group: (group) On a console, a subset of the input channels, selected
to obtain a submix.
Headroom: Overhead margin.

Hertz: Unit of frequency, equal to one cycle per second.
Hysteresis: (hysteretic) 1. Magnetic phenomenon that involves a
delay between the magnetizing signal and the magnetic field in a
ferromagnetic material. 2. In a gate, it differentiates between the closing
and opening thresholds, the purpose of which is to prevent repetitive
closing and opening when the signal is very close to the noise level.
Hz: Abbreviation of Hertz.
Yo
IMD: (intermodulation distortion) Intermodulation distortion (acronym in

English); a specification of said distortion, obtained by exciting the system
with a 60 Hz tone superimposed on another 78 kHz tone of 4 times smaller
amplitude and expressing the effective value of the generated partials as a
percentage of the effective value of the original tones.
Impedance: (impedance) Concept similar to resistance, with the
difference that it varies with frequency, and also introduces a phase shift
between voltage and current.
Acoustic impedance: (acoustic impedance) Ratio between sound
pressure and the speed with which air particles vibrate (not to be confused
with the speed of sound waves).
Input impedance: Impedance “seen” by a source between the input
terminals of a device.
Output impedance: (output) Internal impedance of the output of a

device when it acts as a signal source for a subsequent device.
Internal impedance: (internal impedance) Equivalent impedance of
a real voltage source.
Insertion: (insert) 1. Action of interleaving a processor on a signal
line. 2. Insert connection.
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Channel Insert: (channel insert) Insertion connection that can be
made on each channel after corresponding level adjustment.
Group Insert: Insertion connection into a group signal before
feeding it to the main mix bus.
Insert: Insertion connection (English word).
Electric current intensity: (electric current intensity) Amount of
charge that passes through a conductor per second.
Sound intensity: (sound intensity) Sound energy that passes through
a surface per unit of time and per unit of area.
Intermodulation: (intermodulation) Mutual modulation between two
sinusoidal tones.
Internet: Global computer network that allows the transfer of
information of all types (scientific, academic, cultural, commercial).
Interval: (interval) In music, difference in pitch between two notes. It
is expressed as the number of notes of the scale that are between them,
including them.
J.
Javelin: (grounding rod) Conductive rod or pipe that is buried a couple of

meters into the earth to obtain an effective grounding connection.
Faraday cage: (Faraday's cage) Metal covering around a cable, connector,
device, etc.; which prevents electric fields from emanating from it, or from
receiving them from outside.
kHz: Abbreviation for kilohertz.

kilo: Prefix that precedes the name of a unit and multiplies it by 1000.
kilohertz: Unit of frequency equal to 1000 cycles per second.
kiloohm: Unit equal to 1000 Ω.
kΩ: Abbreviation for kiloohm.
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K3: Third harmonic distortion. Symbology used on magnetic tapes.
LCD: (liquid crystal display) Liquid crystal display (English acronym). A

type of display with very low consumption (much less than LEDs) widely
used in professional sound equipment. It may be backlit.
LED: (light emitting diode) Light emitting diode, widely used in all
types of modern electronic equipment such as light signaling.
LFO: (low frequency oscillator) Low frequency oscillator (acronym
in English).
Limiter: (limiter) A device that, without distorting the waveform,
automatically adjusts the gain so that the signal cannot increase above a
certain level.
Line: (line) 1. Any conductor that carries electrical energy or signal.
2. Referring to the general power supply, 220 V home energy distribution
conductors (or 380 V three-phase in high-power systems). 3. Any medium
level audio signal input or output.
Omnibus line: (bus) In a console, a set of lines that transport various
signals to the inputs of an adder or mixer.
Field lines: (field lines) Lines that follow the direction of the
magnetic field and whose concentration is proportional to the field.
Flux lines: (flux lines) In a magnetic field, field lines.
Lines of force: (force lines) In a magnetic field, field lines.
Spectral lines: (spectral lines) In a discrete spectrum, each of the

frequencies that compose it.
log: Abbreviation for logarithm.
log 10 : Abbreviation for decimal logarithm.
Logarithm: (logarithm) Mathematical function with the property of
growing in equal increment when the variable is multiplied by a constant.
Another important property is that the logarithm of a product of two
numbers is equal to the sum of the logarithms of each of said numbers. The
logarithm is useful for representing numbers with many digits in
compressed form. For example log 10 1,000,000 = 6
Logarithm to base 10: (base ten logarithm) Decimal logarithm.
Wavelength: Distance in space between two successive peaks of a
periodic sound wave.
146
M
mA: Abbreviation for milliamp.

Magnetic: (magnetic) Said to be materials with magnetic properties.
Magnetism: (magnetism) Branch of Physics that studies magnetic
fields.
Remanent magnetism: Residual magnetic field of a ferromagnetic
material after magnetically exciting it and removing the excitation.
Overload margin: (headroom) In an amplifier, console, etc.;
difference in decibels between the maximum level of the output signal
before saturation and the nominal or average level.
Dynamic range: (dynamic range) Dynamic range.
Mass: (chassis ground) 1. A point on the metal chassis of a device
from which a grounding conductor is taken
Master: Signal (generally stereophonic) resulting from the main mix
of a console.
MD: Abbreviation for minidisc.
Sound level meter: (sound level meter) Instrument that measures
sound level.
Mixture: (mix) Sum of the signals present on a bus line (bus).
Mixer: (mixer) A special type of amplifier with several inputs that add the
signals present at said inputs.
Auxiliary mixer: (auxiliary mixer) Mixer to which signals from an
auxiliary bus arrive.
Microphone: (microphone) Transducer device that transforms sound
signal into electrical signal.
Capacitive microphone: (condenser microphone) Microphone based
on the variation in capacity of a condenser formed by a fixed plate and a
diaphragm that reacts to sound waves.
Dynamic microphone: (dynamic microphone) Moving coil
microphone, based on the generation of electrical voltage by a coil that
moves in a magnetic field.
micro: Prefix placed before the name of a unit that divides it by
1000000.
micro volt: Millionth of a volt.
micro ampere: Millionth of an ampere.
milli: Prefix that precedes the name of a unit and divides it by 1000.
147
millivolt: thousandth of a volt.
milliwatt: One thousandth of a watt.
Minidisc: Digital recording and playback format based on optical or
magneto-optical technology. Uses psychoacoustic compression.
Common mode: (common mode) In a balanced line or in a
differential amplifier, equal voltages that appear in both conductors
superimpose the signal.
Normal modes: (normal modes) Resonant frequencies of an
enclosure, room, hall, premises, etc.
Modulation: (modulation) Relatively slow variation of any
parameter of an electronic system.
Amplitude Modulation: (amplitude modulation) Periodic variation
of the amplitude of a signal.
Phase modulation: Periodic variation in the frequency of a signal.
MOL: (abbreviation in English for “maximum output level”)
Maximum output level, parameter of a magnetic tape that represents the
maximum signal level to be recorded with a third harmonic distortion (K3)
of less than 3%.
Monitor: Speaker used to listen to a mix by the sound engineer in the

control room or the musician on the stage or recording room (in the latter
case, headphones are preferably used).
Monitoring: (monitoring) Listening to a mix for verification,
adjustment or support for musical interpretation.
MPEG: Abbreviation for “Motion Picture Expert Group”. A series of
standards developed by said group, such as MP3, for psychoacoustic
compression of digital audio. It is used to transfer digital audio over the
Internet and also to store large amounts of music in a format such as CD-
ROM. Allows compression up to 12 times (with some degradation of the
original signal).
MP3: Compressed digital audio file format widely used on the
Internet.
ms: Abbreviation for millisecond.
Sampling : Action of taking a sample of a signal at regular intervals.
Multiamplification: (multiamplification) Similar to biamplification
but with three or more bands.
Multimeter: (multimeter) Instrument to measure voltage, current,
resistance and optionally other electrical parameters.
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mV: Abbreviation for millivolt.
mW: Abbreviation for milliwatt.
μ: Abbreviation of micro.
Neutral: (neutral) In the home distribution line of electrical energy,

conductor connected to ground in the power plant.
Line level: (line level) Average signal level, shared by the inputs and
outputs of most devices used in sound. Corresponds to average signal
values between 245 mV and 24.5 V.
Sound pressure level: (sound pressure level) 20 times the decimal
logarithm of the sound pressure divided by the reference pressure, that is,
20 μPa. It is expressed in dB.
Power level: (power level) 10 times the decimal logarithm of the
power divided by a reference power that must be specified in each case (see
dBm).
Sound power level: 10 times the decimal logarithm of the sound power
divided by a reference sound power equal to 10 (-12) W
Loudness level: Psychophysical magnitude that compares the
loudness of sine tones with that of an equally loud (loud) 1 kHz sine tone.
They are expressed in phon. For example, a tone has 60 phon if it is equally
loud as a 1 kHz tone of 60 dB sound pressure level.
Voltage level: 20 times the decimal logarithm of the voltage divided
by a reference voltage that must be specified (see dBu and dBV).
Sound level: (sound level) Average sound pressure level interposing
filter A, which has a response similar to that of hearing. It is expressed in
dBA.
NPS: (SPL) Abbreviation for sound pressure level.
Core: (core) Ferromagnetic material introduced inside a coil to
increase the magnetic field caused by the current that circulates through it.
Laminated core: (laminated core) Core divided into thin sheets to
reduce eddy currents.
EITHER
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Octave: (octave) Musical interval corresponding to multiplying the
frequency by 2.
Central octave: (central octave) Octave that includes all the sounds
of the chromatic scale between the C frequency 261.63 Hz
(center DO) and the SI frequency 493.88 Hz.
Ohm: Unit of resistance, equal to one volt divided by one ampere. It
is abbreviated Ω.
Ω: Abbreviation for ohm.
Omnidirectional: (omni directional) Type of microphone that is
equally sensitive in all directions.
Sound wave: (sound wave) Disturbance of the equilibrium pressure
of the air, which propagates away from the source that originated it.
Wave: (wave) 1. Disturbance that propagates in a medium. 2. Due to
abuse of language, waveform.
Square wave: (square wave) Wave whose waveform alternately takes two
values, generally opposite, remaining equal times at each value.
Quasi-symmetrical wave: (quasi symmetrical wave) Wave that in
each cycle has two parts of similar shape and duration but of opposite sign.
Sawtooth wave: Triangular wave in which the rise is much faster
than the fall, vice versa.
Sine wave: (sine wave) Wave whose waveform is a sine wave.
Symmetrical wave: (symmetrical wave) Wave form that in each
cycle has two parts of the same shape and duration but of opposite sign.
Stationary wave: Wave in an enclosure or room between two
parallel walls, which comes and goes, reflecting itself again and again.
Triangular wave: Waveform in which the value increases linearly
from a negative value to a positive value, and then decreases linearly back
to the original value, this cycle repeating periodically.
Low frequency oscillator: (low frequency oscillator, LFO)
Oscillator commonly used for modulations, its frequency generally being in
the range of 0.1 Hz to 30 Hz.
Oscillogram: (oscillogram) Graph on a system of axis with time on
the horizontal axis and the magnitude of a signal on the vertical axis. It
represents the temporal evolution of the signal.
Oscilloscope: (oscilloscope) Measuring instrument that allows
viewing the Oscillogram of a voltage signal on a screen.
150
Q
Pa: Abbreviation of Pascal.

Pan: Contraction of “panoramic”. Potentiometer used for panning.
Connection panel: (patch bay, patch panel) Board with multiple
connectors organized in the form of a matrix, on the back of which there
are semi-permanent connections with the various components of a sound
system, and on the front of which the specific connections are made. It
allows you to simplify the connection work in the daily work of a studio.
Panning : Action of sending the same signal in different proportions to two

stereophonic channels, stimulating the sensation of directionality of the
sound. It is usually complemented with delays to achieve more realistic
effects.
Pan pot: Contraction of “pan pot.”
Speaker: (speaker, loudspeaker) Loudspeaker.
Pascal: Internationally adopted unit of pressure, equivalent to
approximately one hundred thousandth of atmospheric pressure.
Polar pattern: (polar pattern) Directional diagram (poorly correct
translation).
Periodic: (periodic) Said of a phenomenon, signal, etc.; which is
repeated without variations from time to time T.
Period: (period) 1. Time T that it takes to complete a cycle in a
periodic signal. 2. For abuse of language, cycle.
Magnetic permeability: Ratio between the magnetic field in a
material and the magnetizing force that creates it.
PFL: (pre fader listen) Pre fader only. On a console, a type of solo
selector that allows the channel signal to be monitored regardless of the
ratio in which it is sent to the main or group mixer. It is used during
adjustment of the input signal level to the channel.
Phaser: Effect obtained by adding the dry signal to the signal filtered
with a high-Q bandpass filter whose frequency is controlled with an LFO.
Piezoelectricity: (piezoelectricity) Phenomenon that occurs when
certain crystals deform, by which a tension is generated at their ends. It is a
reversible phenomenon, meaning that when tension is applied, the glass
deforms.
Track: Each sector of a magnetic tape in which a signal is recorded,
generally independent of those recorded on other tracks. The tracks are
parallel.
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Play back: Recording technique consisting of recording a second
voice, instrument, etc.; on a second track while playing a previously
recorded track.
Polarization: (bias) Superposition of a direct or alternating current
with the signal to obtain a more linear transfer curve. It is used in every
amplifier and to improve the dynamic range of the signal recorded on
magnetic tape.
Polycarbonate: (polycarbonate) Transparent material with great

dimensional stability with which compact discs are manufactured.
Post fader: On a console, any signal (especially those going to the
aux sends) that is taken after the channel fader.
Power: (power) Energy transferred per unit of time. It is measured in
watts.
Acoustic power: (acoustic power) Acoustic energy transferred or
radiated per unit of time. It is measured in watts.
Electric power: (electric power) Electric energy transferred per unit
of time.
Maximum power: (maximum power) 1. In an amplifier, the
maximum power that it can deliver without saturating, protection acting or
being destroyed, as the case may be. 2. In a loudspeaker, the maximum
power that can be delivered without being destroyed (see, maximum
average power, maximum program power and maximum peak power).
Maximum average power: (maximum average power, maximum
RMS power) Maximum power to be delivered to a speaker without being
destroyed or degraded by overheating.
Maximum program power: Maximum power of a typical musical
signal to be delivered to a speaker. It takes into account the fact that in a
typical signal the maximums occur only from time to time and due to
thermal inertia they do not heat the speaker excessively.
Maximum peak power: (maximum peak power) Maximum power to
be delivered to a speaker without it being disconnected, that is, without the
coil hitting the bottom of its path or moving out of its position.
Maximum RMS power: (maximum RMS power) Incorrect
designation of the maximum average power; The RMS rating applies to
currents or voltages, not powers.
Sound power: (sound power) Sound energy transferred or radiated
per unit of time. It is measured in watts, but should not be confused with
the electrical power delivered to a speaker or acoustic channel. In general,
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the power radiated by a speaker is a very small fraction (between 1% and
15%) of the electrical power supplied to it.
Potentiometer: (potentiometer) Variable resistor used for various
adjustments. It can be rotary (for example the input level control of a
console) or slider (for example the channel fader of a console or the band
settings of a graphic equalizer).
Preamplifier: (preamplifier) Amplifier to raise the level of a low-level

signal such as signals from microphones, pickups, playback heads, etc.; up
to the line level.
Pre delay: It is a reverberation, delay between the direct signal and
the first reflection.
Pre fader: On a console, any signal (especially those going to the aux
sends) that is taken before the channel fader.
Sound pressure: (sound pressure) Difference between the
compression or decompression pressure due to a sound wave and the
equilibrium pressure (atmospheric pressure in the absence of sound).
Processor: (processor) Any device that makes some change in an
electrical signal.
Effects processor: (effect processor) Complementary equipment for
a sound system that introduces effects into the signal.
Grounding: (safety ground) Connection of the mass of a piece of
equipment to the physical ground, usually by means of a javelin (see).
Q: 1. In a band-pass filter, the ratio between the center frequency and the
bandwidth, also called the quality factor. 2. Abbreviation for electrical
charge.
Quasiperiodic/a: See Quasiperiodic/a.
Fifth: (fifth) Musical interval obtained by multiplying the frequency
of a sound by 3/2.
A: Common abbreviation for resistance.
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Rack: Modular metal frame with a standard width of 19” (=48.26
cm) on whose side metal rails the different processors, effects, connection
panels, and even small consoles provided to be mounted on it are screwed.
The standard height of processor cabinets is a multiple of 1.75” (4.45 cm).
Direct radiation: (direct radiation) Emission of sound through the
diaphragm of a speaker without interposing an adaptation by means of a
horn.
Dynamic range: (dynamic range) Difference in dB between the

maximum and minimum levels of a signal.
Extended range: Quality of certain speakers to respond relatively
flatly over a fairly wide range of frequencies. They are not used in
professional sound.
RCA: A type of shielded cable connector that is usually used for line
signals from generally non-professional equipment (although many
professional equipment also incorporates them for versatility).
Feedback: Reinjection of part of the output of a device or system into
its input.
Clipping: (clipping) Severe effect of saturation of an amplifier or
other device, which consists of the highest peaks of the signal being limited
to a constant value, although the input signal continues to vary (increase).
Crossover network: (crossover) Frequency divider.
Frequency divider network: (crossover) Frequency divider.
Noise reducer: (noise reduction system) Device that improves the
signal-to-noise ratio in an analog recording system.
Complementary noise reduction: (complementary noise reduction
system) Noise reducer that encodes the signal during recording, raising low
levels, and decodes it during playback, returning the signal to its original
dynamic range.
Non-complementary noise reduction system: A noise reducer that
acts only during playback, reducing gain or bandwidth when the signal is
so low that it can be assumed to be tape noise.
Bass reflex: (bass reflex) Ventilated acoustic box in which low-
frequency sound is internally delayed by a half-period, reinforcing the main
wave instead of interfering with it.
Reflection: (reflection) Reflection of a sound on a surface, returning
to the environment limited by that surface.
Early reflections: (early reflections) First reflections of sound on the
surfaces of a room, before successive reflections become too close in time.
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Late reflections: (late reflections) Sound reflections that occur in
close proximity to each other, giving rise to the new phenomenon of
reverberation.
Compression ratio: (compression ratio) In an audio compressor, the

ratio at which the level that exceeds the threshold is reduced.
Signal to noise ratio: (signal to noise ratio) Conscious between the
effective value of the signal and the effective value of the noise, frequently
expressed logarithmically in dB. In this case it can be calculated as the
signal level minus the noise level.
Relay: (release) 1. In a sound, the final stage during which the
amplitude of the sound is reduced until it disappears. 2. In a dynamics
processor, effects processor, etc.; stage during which one moves from the
active situation to the non-active situation.
Performance: (efficiency) Percentage relationship between the
power delivered by a device and that consumed by its power supply. In the
case of a loudspeaker, the relationship between the effective acoustic power
and the electrical power.
Resistance: 1. (resistance) Magnitude associated with a resistor, the
greater the more difficult it makes the passage of electric current, equal to
the ratio between the applied voltage and the current that circulates through
it.
2. (resistor) By abuse of resistor language.
Internal resistance: (internal resistance) Equivalent resistance of a
real voltage source.
Resistor: (resistor) Component of electrical circuits that makes the
passage of electric current more or less difficult.
Resonance: (resonance) In any vibratory or oscillatory physical
system, a phenomenon by which at a certain frequency the response is
greater than at other nearby frequencies.
Frequency response: (frequency response) Relationship between the
output and input of any system as a function of frequency.
Delay: (delay) Modification of a signal by which it appears delayed
by a certain time with respect to the original signal.
Return: 1. (return) Any signal line that returns to a console from
external equipment. 2. (monitor) Name sometimes used for the stage
monitor.
155
Auxiliary return: (auxiliary return) In a console, stereo signal input
from an external device and which is returned to the main mix or submixes
after a previous level adjustment. It is used for effects or parallel
processors.
Insert return: On a console, monophonic input from an external
device inserted into the signal path for serial processing.
Reverb: Short for reverberator, widely used informally to refer to the

processor or its effect; also review.
Reverberation: Permanence of a sound in an environment after its
source has been extinguished due to multiple reflections.
Reverberator: (reverberate, reverb) Device that artificially delivers
reverberation to a signal.
RMS: (root mean square) Effective value.
Noise: (noise) 1. An unwanted signal, often coming from intrinsic
sources involved in the physical amplification process.
2. A signal that has a continuous spectrum.
Acoustic noise: Ambient noise due to the superposition of a
multitude of nearby distant sources.
Digitization noise: Noise that appears in a digital system as a result
of approximating an exact sample with a limited number of steps.
Electrical noise: (electrical noise) Noise generated intrinsically in
electrical and electronic components. In general it grows with temperature.
White noise: (white noise) Signal whose spectral density is constant
with frequency (all frequencies appear in the same proportion).
Brownian noise: (Brownian noise) Signal whose spectral density
decreases with the square of the frequency, and therefore has a large low
frequency content.
Pink noise: (pink noise) Signal whose spectral density decreases with
frequency. It has the particularity that its energy is the same in each octave
band. It is used as a test signal in various acoustic tests.
Thermal noise: (thermal noise) A type of electrical noise generated
by any electrical conductor. It increases with increasing resistance value
and with increasing temperature. For example, a 150 Ω resistor generates,
at room temperature, a thermal noise of 0.22 μV.
156
Yes
s: International abbreviation of second.

Anechoic room: (anechoic chamber) Room acoustically treated to
almost completely eliminate reflections or echoes, creating a free field
situation. It is used to measure the directional diagram of various sources or
microphones.
Control room: (control room) Room in which most of the equipment for
both live and studio sound is located, controlled by an operator or sound
engineer. In general it is acoustically isolated from the rest of the facilities.
Direct output: On some consoles, the output of a channel, usually
after the channel fader, that can be used by external equipment without
altering the signal going to the mixer.
Saturation: State of a device or system in which an increase in
excitation no longer produces a further increase in response.
Half period: (half period) In a symmetrical waveform, half a period.
In a quasisymmetric waveform, each part is similar in shape but opposite.
Sinoid: (sinusoid) Mathematical function that represents the
evolution in time of a simple oscillation (for example that of a pendulum).
Any periodic wave is the superposition of a series of sine waves of
frequencies multiples of a fundamental frequency.
Sensitivity: (sensitivity) 1. In a microphone, the relationship between
the generated voltage and the sound pressure (v/p), also expressed in dB
referred to 1 V/Pa. 2. In an amplifier, voltage ratio to apply at the input to
obtain the specified power. 3. In a loudspeaker or loudspeaker, the sound
pressure level obtained 1 m in front of it when 1 W is applied to it.
Signal: (signal) A physical quantity varying in time that transmits or
transports information.
Dry signal: (dry signal) referred to effects processors, unprocessed
signal.
Channel separation: (croostalk) Difference in voltage levels
between the outputs of the unexcited channel and the excited channel in a
stereophonic or multichannel system.
157
Electric shock: (electric shock) Accident that occurs when the human
body comes into contact with two different points of tension, for example
the live power line and the earth. The consequences can be serious,
depending on the time of exposure and the stress to which it was subjected.
It can be avoided by grounding.
SIP: (in place solo) Post fader only. On a console, a type of solo
selector that allows monitoring of the channel signal as it is routed to the
main or group mixer. Used during adjustments for mixing (including
panning).
System: (system) Any connection of entities that process signals,
influencing each other to a greater or lesser extent.
Only 1. On a console, a selector that disconnects the rest of the
channels not selected as solo from the monitoring bus. 2. Solo part of a
piece of music.
Sound: (sound) Wave that propagates in air, water and other media,
whose frequency is between 20 Hz and 20kHz.
Harmonic sounds: (Harmonic sounds) Each of the sinusoidal
components of frequencies multiples of the fundamental that make up a
periodic signal.
Masking sound: (masking sound) Mask sound.
Mask sound: (masking sound) Sound that masks another.
Partial sounds: (overtons) Each of the sinusoidal components that
make up a direct spectrum signal.
Sound deflector: (baffle) Acoustic box.
Loudness: (loudness) Sensation that allows us to distinguish weaker
sounds from stronger ones.
Mute: (mute) 1. Selector that disconnects the signal on a specific
channel of a console. 2. In various acoustic instruments, a device that
reduces the loudness of the sound emitted, while modifying its timbral
quality.
Sustain: (sustain) In a basic envelope of an electronic synthesizer, a
stretch during which the amplitude of the wave remains constant until the
corresponding note is turned off.
Squawquer: Speaker for reproducing mid frequencies, typically
between 500Hz and 6 kHz.
Subgroup: (subgroup) Another name for a console group.
158
Submaster: Another name for a console group.
Submix: (sub mix) Secondary mix obtained by adding the signals of
a group.
Subsystem: (subsystem) A part of a system, made up of one or more
blocks, that performs a defined and reasonably autonomous function within
the larger system.
Subsound: (sub sound) Inaudible acoustic waves with frequencies
lower than 16 to 20 Hz.
Subsonic /a: (subsonic) Relating to subsound.
Subwoofer: Speaker for the reproduction of very bass sounds,
typically below 100 Hz.
SVHS: Abbreviation in English for “Super Video Home System”.
Tape format for high-quality video recording, also used by several models
of multitrack digital recorders.
Voltage: (voltage) Electrical magnitude measured between two points of

an electrical circuit, equivalent, in hydraulics, to pressure.
Fourier's theorem: Property (mathematically provable) by which
every periodic waveform can be considered as a superposition of sinusoids
of frequencies multiples of a fundamental frequency.
Tester: (multimeter) Multimeter.
THD: (total harmonic distortion) Total harmonic distortion (acronym
in English); a harmonic distortion specification. It is determined by
measuring the effective value of the harmonics generated by the device at a
sinusoidal input, and expressing it as a percentage of the effective value of
the fundamental.
Attack time: Duration of the attack stage of a sound or the operation
of a processor.
Release time: Duration of the release stage of a sound or the
operation of a processor.
Reverberation time: (reverberation time) Time it takes for a sound
to drop 60 dB below the level it had when its source was interrupted.
Earth: (earth, ground) 1. A point on a circuit connected to earth

ground by a buried conductor (javelin). 2. In a circuit, a point with respect
159
to which all voltages in the circuit are referred. Its voltage is conventionally
0 V.
Timbre: Sensation that allows sounds to be distinguished by their
source.
Tone: (tone) 1. Periodic sound. 2. Due to abuse of language,
sinusoidal periodic sound. 3. Musical interval formed by two semitones.
Topology: (topology, topography) Internal structure of connections
of strongly connective devices such as mixing consoles.
TRS: (tip-ring-sleeve) A ¼” isoaxial (on a single axis) plug
connector for stereo or balanced signals, with a ground (or common)
contact around it and two ring-shaped and tip respectively.
Transduction: Action of converting a signal from one form of
energy to another (for example, from sound to voltage).
Transducer: (transducer) Device that transforms a signal from one form of
energy to another.
Transformer: (transformer) Device formed by two windings of
insulated conductive wires that are wound around a ferromagnetic core (in
general). It serves to: 1) reduce or increase (depending on the case) the
voltage and 2) to achieve impedance matching between a source and a load.
Tremolo: (tremolo) Effect that consists of modulating the amplitude
of a signal.
Trim: Level adjustment of the input signals of a console. It is used to
match signals from different sources.
TS: (tip-sleeve) A ¼” isoaxial (on a single axis) plug connector for
monophonic or unbalanced signals, with a ground contact around and a tip.
Tweeter: Loudspeaker for reproducing high-pitched sounds, typically
above 1500 Hz.
OR
Ultrasound: (ultrasound) Inaudible acoustic waves with frequencies

greater than 20 kHz.
Ultrasonic /a: (ultrasonic) Relating to ultrasound.
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Threshold: (threshold) In dynamic processors such as compressors,
gates or expanders, limit level between the range in which the processor
acts and the range in which it does not act.
Threshold of hearing: (threshold of hearing) The minimum level of sound
pressure required to evoke a sound sensation. Normally it is close to 0 dB.
Pain threshold: (threshold of feeling) The level of sound pressure
that begins to produce ear pain. It is usually close to
120 dB.
V: Abbreviation of Volt.
Peak value: (peak value) Maximum value that a signal reaches in
each cycle.
Effective value: (RMS value) Hypothetical constant value capable of
producing the same average power as that produced by the variable signal.
Peak to peak value: Difference between the maximum positive value
and the minimum negative value in each cycle of a signal.
VCA: (voltage controlled amplifier) Voltage controlled amplifier.
Speed of sound: (sound speed) Speed of propagation of the sound
wave. It is approximately 345 m/s
Constant linear velocity: (constant linear velocity, CLV) Criterion used in
magnetic tapes and compact discs, it consists of maintaining a constant
speed of movement of the information recorded in front of the reading unit.
VHS: Abbreviation in English for “Video Home System”. Tape
format for video recording.
Vibrato: Effect that consists of modulating a signal in frequency or
phase.
Live: (hot) In the home distribution line of electrical energy, the
conductor not connected to ground in the power plant.
volt: Unit of electrical voltage. V is abbreviated.
Vumeter: (vumeter, volume indicator) Analog or quasi-analog

indicator (LED bar) for the signal level at various points of a system
(console, compressor, etc.)
W: Abbreviation for Watt.
161
Watt: Unit of power. In the case of electrical power, it corresponds to
the power delivered in a 1 Ω resistor when 1 V is applied to it.
Woofer: Speaker for bass sound reproduction, typically below 500
Hz.
XLR: Generic name for a connector for balanced signals. Also called
Cannon, which is the brand under which it was introduced to the market.
XY: Configuration of two microphones in quadrature (90º to each other) to
make stereophonic shots.
AND
And insert cable: Cable terminated at one end by a TRS connector and at
the other by two TS , used to insert an effect or serial processor into the
insert connections of consoles or other equipment.
Z: Abbreviation for impedance.
BIBLIOGRAPHY
1. Davis, Gary; Jon, Ralph. “The Sound Reinforcement Handbook”

(2nd Edition) Hal Leonard Publishing Corporation. Milwaukee,
USA. 1990 (412 pages).
2. Davis, Don; Davis, Carolyn. “Sound System Engineering”
(2nd Edition) SAMS, Carmel, USA, 1994 (665 Pages).
3. Everest, Frederick Alton. “The Master Handbook of Acoustics”
162
(2nd Edition) Mac Graw Hill (TAB books) Blue Ridge Summit,
USA,
1989 (366 pages).
4. Shure Guide to “Audio System for Houses of Worship” USA, 1988
(59 Pages).
5. Shure 75th Anniversary Special Edition Magazine, USA, 2000 (66
Pages)
In addition to the previous books, other publications have been used

from which only fragments have been extracted, so it was not considered
useful to include them here.
Likewise, numerous brochures and equipment specifications were
consulted, which can be obtained free of charge from distributors or the
manufacturer itself.
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CONTENT.

Chapter 1. Physical Acoustics.

Chapter 3. Architectural Acoustics.

Chapter 4. Signals and Systems.

Chapter 8. Speakers and Acoustic Boxes.

Chapter 9. Frequency Processors.

Chapter 10. Dynamic Processors.

Chapter 11. Mixing Consoles.

Chapter 12. Monitors.

Chapter 13. Cables and Connectors.

When you speak or play an instrument, a

Sound pressure waves are cyclical and we can represent them

If we follow the trajectory of the circle at a constant speed, setting the

Apart from the frequency with which a wave repeats, it has a

1 and 2 same frequency different amplitude.

331.4 m/s 0  C + 0.6 m/s * 1  C

This means that every time the temperature increases by 1  C we

21  C * 0.6 m/s = 12.6 m/s

The speed of sound in feet/sec is 1130 peis/sec.

It is the distance that a cycle of any frequency needs to develop

For example, if we want to know the wavelength of 300 hertz at 21 

 = 344 m/s / 300 Hz = 1.14 m.

For example, if we want to know the period (T) of 300 Hz:

T = 1 / 300 Hz = 0.00333 seconds.

This means that at 21  C 1.14 m. They are traveled in 0.00333

For it to exist out of phase, only 1  phase shift is required; in this

SOUND PRESSURE LEVELS.

The ear operates in a 10 (13): 1 energy range which is an extremely

dB SPL = 20 Log P / P ref.

The electrical signal level, also measured in dB, corresponds to 10

Electrical signal levels in decibels can be expressed in several ways:

dBm: decibels with a reference of 1 milliwatt.

THE DECIBEL (dB).

The decibel (dB) always describes a relationship between two

dB = 10 Log (P2 / P1).

dB = 10 Log ( 100 / 10)

The previous examples allow us to realize two interesting aspects in

a) Whenever one power is twice as high as another, it represents

b) Whenever one power is 10 times greater than another, it only

The decibel can also be used to calculate the relationship between

DB volt = 20 Log (E2/E1)

According to the results obtained using this formula, the relationship

If we calculate the power generated by 100 volts in an 8 ohm resistor

If we calculate the power generated by 10 watts in an 8 ohm resistor:

Now if we calculate the relationship of both powers in dB:

Below we present a list of values in dB considering 1 watt as a

THE dB RELATED TO ELECTRICAL UNITS.

The dBm is related to electrical units with 1 milliwatt as a reference.

Yes: dBm = 10 Log (P2 / P1)

Modern Audio equipment is sensitive to voltage levels, in this

This is a voltage related unit, which has a reference value of 1 volt

Knowing that E1 = 1 volt.

RELATIONSHIP OF dBV, dBu, dBm WITH THE

In some cases, we want to connect low impedance outputs to high

It is a measure of power in watts (Watts) created by Mix magazine,

RELATIONSHIP OF dB WITH ACOUSTIC SIGNAL.

dB can be used to measure sound pressure level, which represents the

If we experiment a little with this equation, we find that if one SPL is