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Audio Fundamentals
Sound and Audio
Sound Physics
The perception of sound is a mechanical disturbance of molecules within a surrounding elastic medium, such
as air. Sound energy travels through all phases of matter; i.e. gases, solids, or liquids. This molecular
disturbance propagates through the given medium at a rate related to the close proximity of its own molecule
structure, or density. The molecular disturbance is a type of kinetic energy known as acoustic energy.
Sound is the result of two or more objects colliding with sufficient force to direct the resultant energy as an
acoustic wave from the point of contact. Some of that energy is converted to heat, not sound.
Ultimately, the portions of sound energy converted to heat become greater than the resulting transmission of
force between molecules and the sound energy dissipates to an imperceptible level.
The process of hearing a sound relies on the mechanical disturbance propagating through all of those forms of
matter, not only in our environment, but even within the human ear; for example, from air through a solid,
through a liquid, and ultimately to the nerve endings in our ear that transmit impulses to the brain where the
disturbance is interpreted as a sound.
Sound Propagation
Can sound propagate through the vacuum of space? The answer is no. Only in the presence of an acoustic
energy medium will sound waves propagate. For all practical purposes, no molecules of liquid or gas exist in
the vacuum of space to transfer the kinetic energy created when two solid objects collide. Therefore, in space,
the collision of two objects results only in the local conversion of energy to heat. Here on Earth, our
perception of sound results mostly from mechanical vibrations traveling through air. In this process, a loud rise
in sound level creates an incoming positive pressure wave of air molecules much like waves rising on the
ocean. The corresponding drop in sound level results in less wave pressure behind the original wave, like the
valley between ocean waves. This region of low sound pressure is called a rarefaction. Successive pressure
waves followed by rarefactions constitute a continuously perceived sound level.
Speed of Sound
The speed at which sound vibrations travel depends on the closeness of the molecules within the medium in
which they travel. Generally, where molecules are close together, or dense, vibrations travel faster because
one molecule travels less distance in order to disturb the molecule next to it. Therefore, sound travels fastest
through dense solids, like steel, versus gases, like air, where the distance between molecules is much greater.
It is necessary to consider the speed of sound propagation depending on the medium. In addition,
temperature affects sound propagation speed. While the speed difference may be small within solids of
varying temperature, air temperature has a significant impact on sound speed. Dense, cold air molecules are
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much closer together than warm air molecules. For this reason, the speed of sound in air must be specified
with respect to temperature and altitude, which affects air pressure.
Threshold of Hearing
The worldwide agreed-upon reference for sound pressure measurement is sea level at a temperature of 68° F,
or 20°C. This condition equals 1 atmosphere of pressure. One atmosphere of pressure is equal to about
100,000 newtons per square meter or about 14.5 pounds per square inch.
The newton, an international unit of measure for force, equals about 0.225 English pound of force. The human
ear is so sensitive that, on average, it can perceive sound pressures as small as two one-hundred-thousandths
of a newton per square meter, or 0.00002 newton/sq. m, or commonly known as 20 micro-Pascals (µPa); a
very low sound pressure indeed! At about 4 KHz, humans may perceive sound at one-half that level or 10 µPa.
This region of sound pressure is known as the threshold of hearing.
Threshold of Pain
At the extreme of the human hearing sensitivity range is the threshold of pain. Very loud sounds, which are
perceived but not painful, may be as large as 64 Pa. Comparing the two extremes, the range of human sound
pressure sensitivity is a ratio of 6.4 million to one. The range is so large that working with sound pressure
levels is daunting without some easier method of managing the intensity of sound levels.
The Decibel
Since our perception of loudness is logarithmic and closely related to the power level required to get work
done, an easier method for working with the wide range of sound pressure levels we perceive is to use ratios
based on logarithms. The decibel is used to describe the ratio of two logarithms that reduce calculation of
sound pressure levels to simple arithmetic. The term dB (SPL) is used to represent the useful range of Sound
Pressure Levels.
Using decibels, the number range 0 dB to 130 dB represents the normal range from the threshold of hearing
through the threshold of pain. Decibels provide a more convenient realm for communicating and
understanding sound pressure levels. More will be discussed on the decibel scale a little later.
Not long after his death, Bell Telephone Laboratories honored Alexander Graham Bell, its founder and
inventor of the telephone, by renaming the primary unit of audio signal transmission loss in his honor. The
Transmission Unit (TU), as it was then called, became the “Bel”; representing a perceived sound level loss of
fifty percent over one mile of standard telephone cable.
As applied to human hearing, a system exhibiting a perceived sound level loss of fifty percent has one Bel of
loss and, likewise, a system perceived twice as loud has one Bel of gain. Although the Bel is still used, it is
considered too large and cumbersome when calculating and describing our perception of sound levels and the
signal power loss ratio of most electronic systems. The prefix “deci”, meaning one tenth, was added to create
the decibel (dB), or one tenth of a Bel. Therefore, the perception of a doubled sound pressure level equals 10
dB (SPL).
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Human Hearing
Vibrations propagating through air encounter two key components of the outer ear: the pinna and the concha.
While each has its own sensitivity to differing frequencies, together these components embody an acoustic
effect, which assists in collecting and localizing sounds. Before reaching the eardrum, sound travels through
the auditory canal which happens to be resonant around 4 kHz; a frequency within the region containing many
components of human speech. The eardrum, or tympanic membrane, transmits sound vibrations from the
external environment into the middle ear via three small coupling bones. This system of bones couple sound
vibrations to the middle ear where the cochlea, a curled, fluid-filled resonating chamber containing the
auditory nerves, converts acoustic vibrations into electrical energy for the brain. The shape and length of the
cochlea along with the auditory nerves within it determine our perception of sound pitch or frequency.
Frequency and Sine Waves
Sine waves can be modeled mathematically and are unique in that they exist as a pure sound, or note, without
harmonics. A sine wave that becomes distorted in any form, such as passing through inadequate or improperly
adjusted audio equipment, contains harmonics. Harmonics are integer multiples (i.e. 2nd, 3rd, 4th, 5th, etc.)
of the original fundamental tone frequency. Any repeating sound or tone that is not a sine wave contains
harmonics of the fundamental frequency. The frequency of a sine wave, tone, or complex sound wave is the
time required to complete one cycle of the wave. Historically, we simply called the frequency “cycles per
second.” To honor the German physicist Heinrich Rudolf Hertz, a pioneer in electromagnetism, the units for
frequency was renamed “Hertz”, or abbreviated “Hz”; where: 1 Hz = one vibratory cycle per second.
Wavelength Formula
Knowing any two of the quantities mentioned previously allows for calculation of the third quantity.
From the earlier discussed table showing the speed of sound with respect to gases, liquids, and solids, we
know that the speed of sound at sea level (at 20° C) is 1128 ft/s or 344 m/s.
Therefore, for any frequency of sound, say 1000 Hz, we can calculate the length of one wave by algebraically
modifying the previous formula as follows:
This gives us our answer of 1.128 feet:
Wavelength @ 1000 Hz = 1128 ft/sec ÷ 1000 Hz = 1.128 feet
Audio
The term “audio” is used to more precisely define sound in mechanical and electronic terms. For our purposes
in AV, audio is used to describe the electronic transmission of sound.
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To accomplish this, the first step is to transform the audible movement of air (sound) into an electrical signal.
A device designed to convert sound to electrical energy or vice versa is called a transducer. In the case of an
audio system, microphones perform the function of transforming sound to electrical energy.
Once in the electronic realm, audio can be amplified, mixed, equalized, digitized, recorded, transmitted and
transformed back into acoustical energy.
The Decibel Scale
Previously, we established the origin of the term decibel along with the useful hearing range in decibels;
namely 0 to 130 dB. The decibel is a unit used to describe the ratio of two power levels, pressure levels,
voltage levels, current levels, etc. So, it may be applied to a wide range of both acoustic and electronic sound
pressure and sound power situations.
In AV, the decibel is an important way of describing the amount of signal present within the electronic realm
or the perceived loudness of a sound in the acoustic realm.
A decibel is 10 times the logarithm of a ratio of two power levels. This is expressed as:
Where voltage gain or loss is concerned, the calculation involves the square of the voltage. The result is that
the logarithm of the two voltage levels being compared must be multiplied by 20 instead of 10. This is
expressed as:
In order to maintain a proper reference of understanding for any ratio in dB, an extension is often added.
Following are the typical extensions used for audio calculations in dB:
dB in volts = dBV
dB in sound pressure = dBSPL
dB in power = dBm (where the m = 1 milliwatt into a specified impedance)
Decibel Measurements
Decibel ratios are logarithmic. The sound pressure level for the threshold of hearing is expressed as 0 dB. A
sound measured at 10 dB has 10 times greater sound pressure level. However, a 10 dB SPL increase is
perceived by humans as approximately twice as loud. The jump from a quiet conversation at home (65 dB) to
the peak sound level at a convention (85 dB) is a ten-fold increase, but we would perceive it as about four
times louder.
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Common Decibel Scales
Audio manufacturers use the decibel (dB) scale extensively to specify product performance. Understanding
decibel relationships is essential to understanding audio specifications. Here are some widely used decibel
scales in audio:
● dBSPL (Sound Pressure Level) — Scale used to express acoustic energy, referenced to 20 micro
Pascals (0.00002 Pa).
● dBu (Unloaded) — Widely used scale to express audio levels of professional equipment,
referenced to 0.775 Vrms.
● dBV (Volts) — Scale used to express audio levels of consumer equipment, referenced to 1.0
Vrms.
● dBFS (Full Scale) — Scale used with digital audio equipment, referenced to the point at which
the signal will distort.
Decibel Changes
A common use of the decibel is in referring to intensity changes for the same sound. These changes refer to a
relative change based on the original sound. An absolute level in decibels for the sound would be expressed
in “dbSPL” where the perceived sound would be compared to the threshold of hearing which is 20µ Pascals or
0 dBSPL. In conversation, the “SPL” is often dropped when speaking of sound intensity. In general:
● 1 dB is the smallest change perceivable.
● 3 dB is a change noticeable to most listeners.
● 10 dB is a change that doubles or halves the intensity.
Comparative Sound Levels in dBSPL
Decibel sound pressure levels of various sound sources vary depending on the particular item and distance.
Over time, continuous noise levels above 85 dB can damage hearing. Noise levels above 140 dB can cause
damage after just one exposure.
Here are some common decibel levels seen in AV:
● 0 dB is the threshold of hearing or the lowest pressure the human ear can hear.
● 130 dB is the threshold of pain or the highest pressure level the human ear can tolerate before
damage can occur.
● +4 dBu is the nominal signal level of professional audio equipment.
● -10 dBV is the nominal signal level of consumer audio equipment.
Note: Many products intended for professional applications use this level if using RCA
connectors.
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VU Meters
Within the audio signal path, it is necessary to monitor audio levels and prevent signal distortion. The Volume
Unit (VU) meter is designed to provide a visual representation of the signal in a manner similar to how the
human ear will perceive it.
VU meters are commonly found in mixers, signal processors, and some amplifiers and are available in both
analog and digital varieties. It is important to keep in mind that the VU meter provides an average volume
readout and not fast peak response. Note the differences between analog and digital VU measurements for
the same sound.
Audio Signal Levels
We will discuss the different audio signal levels for commonly used audio products in our industry.
Common audio products provide signals with a variety of signal strengths referred to as output voltage levels.
These levels vary greatly from a few millivolts for microphones to over 100 volts for amplifiers. Audio signal
level ranges are described by the source’s nominal output level, such as microphone level, line level, or
speaker level.
Microphone Level
Mic level is the nominal signal output of a standard microphone. Mic level has a very low voltage – .775
millivolts (-60 dBu).
Since this signal level is so low, a preamplifier (preamp) is required to boost the signal to line level before it
can be used by other audio processing equipment and advanced down the audio signal path. This preamp is
either an internal circuit or external component in AV.
Line Level
Line level is the operating signal level for AV systems. Line Level is the nominal level that is produced by
devices with a prerecorded signal.
Line level devices include but are not limited to:
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DVD
CD Player
VCR
Tuner
Tape Deck
Output signal characteristics on typical consumer equipment include:
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-10 dBV (Unbalanced)
316 millivolts or .316 Volts
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Output signal characteristics on typical professional equipment include:
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+4 dBu (Balanced)
1.228 Volts
Signal processing equipment such as equalizers and feedback eliminators operate at line level. In addition, the
output from mic preamps and the inputs to power amplifiers are at line level.
Headphone Level
Headphone, or headset level, is the audio level found on portable devices and general purpose computer
sound cards. This signal level is higher than mic level and slightly lower than consumer equipment line level.
Characteristics include:
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-20 dBV or 100 millivolts
Around 16 ohm impedance
It should be noted that the level and impedance must be matched by the source equipment in order for the
headphone to produce accurate sound.
Speaker Level
The audio level coming from the output of a power amplifier is called speaker level.
Higher voltages are required to drive speakers. The voltage for a loud signal coming out of a 100-watt
amplifier might be around 25-30 volts.
A unit of electrical power used to indicate the rate of energy produced or consumed by an electrical device is
called a watt. A unit of electrical resistance, transmitting a current of 1 amp when subjected to a potential
difference of 1 volt, is an ohm. When working with power amp outputs and speakers it is more common to
discuss levels in terms of power measured in watts with respect to impedance measured in ohms.
For example, an amplifier with an output listed as “75 watts per channel with an 8-ohm load” can provide 75
watts on each output when an 8-ohm speaker is connected.
Audio Signal Transmission
Audio signals can be transmitted either as Analog or Digital. Analog audio signals are modulated electronic
versions of transduced acoustic vibrations. Digital audio signals are a data stream of numbers representing
converted acoustic vibrations.
For analog signals, a transducer (such as a microphone) changes acoustic energy into electrical energy.
Another transducer (such as a speaker) changes the electrical energy back to acoustic energy. We will take a
more detailed look at microphones and speakers later in the document. Digital signals require an analog-todigital, or A/D, converter to change or “digitize” the signal into numeric values. Both transmission methods
have inherent benefits.
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Analog Audio Transmission
Analog audio signals can be transmitted in one of two ways, unbalanced or balanced.
The transmission of an unbalanced audio signal is over two wires: signal and ground. This is all that is required
to transmit audio and is common in short cable runs and less professional applications where noise is less of a
problem. Noise is any unwanted signal that adversely affects the quality of the picture or sound.
A balanced audio signal is transmitted over three lines: positive signal, negative signal, and ground. The
addition of this third line provides a way to reject noise.
Like analog signals, digital signals also use balanced or unbalanced connections. Digital transmission is also
becoming more prevalent in fiber optic installations.
Unbalanced Audio Connections
An unbalanced audio signal is the standard signal type found on consumer devices such as a PC, a DVD player,
and a MP3 player. The signal is carried on two conductors, or wires, with the following properties:
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Signal (hot)
Ground (return)
Common connectors used to carry an unbalanced audio signal include:
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RCA - Center pin (signal) and Shield (ground)
3.5 mm Mini Stereo – Tip (left signal), Ring (right signal), and Sleeve (ground)
Captive screw – Pin 1 (left), Pin 2 (n/c), Pin 3 (ground), Pin 4 (right), Pin 5 (n/c)
This can vary by manufacturer.
Unbalanced Audio Signals and Noise
Unbalanced signals are very susceptible to noise.
Extraneous electromagnetic interference – EMI or radio frequency interference – RFI may be picked up and
become part of the signal due to the fact that there is no reference to differentiate between the signal and the
noise.
Balanced Audio Connections
Balanced audio signals are the standard signal type found on most professional devices including switchers,
interfaces, microphones, professional television cameras, etc. The balanced signal is accomplished through the
use of three conductors with the following nomenclature:
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Signal (hot/+)
Return (cold/-)
Ground (shield)
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Common connector types used for balanced audio signals include:
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1/4” phone TRS connector - Tip (+), Ring (-), Sleeve (ground)
XLR Plug (XLR jack shown at right) - Pin 1 (ground), Pin 2 (+), Pin 3 (-)
Captive screw – pin 1 (left +), pin 2 (left -), pin 3 (ground), pin 4 (right +), pin 5 (right -)
The pin configurations listed may vary by manufacturer. Note that captive screw connectors and plugs can be
used in both balanced and unbalanced audio signal applications.
Balanced Audio Signals and Noise Rejection
Balanced signals are very useful due to their ability to reject extraneous noise. The balanced signal is split into
two components that travel out of phase with each other. Extraneous EMI/RFI is picked up on both
components and becomes part of the signal. One signal component is reversed at the destination thus
canceling the noise on both conductors.
Use an Audio Converter product that can accommodate balanced or unbalanced signals for universal
compatibility to provide signal matching between balanced and unbalanced audio equipment. A good
converter may also work as a quality line driver for sending and receiving audio signals over long distances.
Gain is a general term for an increase in signal power or voltage produced by an amplifier. User controls
include variable gain adjustments and a gain compensation switch to maintain consistency between consumer
and professional audio levels.
Audio Program Sources
Program Sources
Program sources are the originating point in the audio signal path.
Equipment in this category is grouped into the two source types based on their typical output level:
● Mic Level Equipment
● Line Level Equipment
We will take a quick look at both of these followed by an extended look at microphones.
Program Sources - Mic Level Equipment
A microphone, as mentioned in an earlier topic, is a transducer – which converts one form of energy into
another. Microphones have a very low output that must be boosted to line level with a preamplifier, or
preamp, in order to be used farther along in the audio signal path.
Microphone connections typically use XLR connectors except for special high-end recording mics and modular
mics (such as wireless).
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Program Sources: Line Level Equipment
As mentioned in an earlier topic, line level is described as the nominal level that audio systems operate at
before amplification. Therefore, sources which already feature an output signal at this level are often referred
to as line level sources.
These sources include, but are not limited to:
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Wireless microphones
Cassette decks
MP3 Players
CD and DVD Players
Video Tape Players
Tuners
Blu-ray Players
While professional versions of these devices have balanced +4 dBu (1.228 V) outputs on XLR connectors, it’s
not uncommon to be asked to integrate models with unbalanced connectors such as RCA.
Types of Microphones
In order to convert acoustic energy into electrical energy, the acoustic energy must be captured. This is what a
microphone does. Microphones are classified based on how they capture sound.
There are two primary types of microphones:
● Dynamic
● Condenser
There are advantages and disadvantages to both types of mics which aid in determining the right application.
Dynamic Microphones
Dynamic microphones utilize a circular diaphragm inside the microphone’s capsule that moves back and forth
as it reacts to changes in sound pressure. As it moves, a small coil attached to the back of the diaphragm
moves in and out of a magnetic field creating an electrical signal. This moving coil design is popular for its
durability and low cost.
A variant of the dynamic design is the ribbon microphone, a classic design that employs a corrugated ribbon
strip suspended in a magnetic field in place of the diaphragm. Ribbon microphones tend to be more fragile
than their diaphragm cousins but remain popular for detailed reproduction of sound.
Condenser Microphones
Condenser microphones utilize a pair of circular plates, one fixed and one that moves in relation to changing
air pressure. One plate is energized by an external voltage source in order to create capacitance that reacts to
acoustic pressure changes.
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The changing capacitance creates the electrical signal. This signal from the capsule is very low (.000775 V) and
must be amplified to bring it up to at least consumer line level (0.316 V).
The external voltage to polarize the capsule and power the microphone’s electronics is provided by phantom
power – electrical current that travels from mixer or mic preamp to the microphone along the mic cable.
While generally more expensive and more fragile than dynamics, these microphones are highly regarded for
their sound quality.
Electret Condenser Microphones
A variant of the condenser mic commonly found in the industry is the Electret Condenser.
This microphone uses a pre-polarized capsule and only needs a small amount of voltage to operate the
electronics. These capsules are used in lectern mics, lavaliere (“clip-on”) mics, and head worn mics.
Microphone Pickup Patterns
Another way to classify a mic is by its directionality or pickup pattern. This pickup pattern is defined by how
sensitive a microphone is to sounds coming from different directions.
Mics can be classified as:
● Omni-directional – 360-degree coverage
● Bi-directional – Coverage from the front and back
● Uni-directional – Coverage from the front
● Cardioid – 180-degree coverage
● Super-cardioid – 160-degree coverage. Slight pickup from rear.
Even more narrow directional mics, such as a shotgun mic, are available.
The pickup pattern of a mic is an important consideration when selecting a microphone for an AV application.
Microphone Output Signal Characteristics
Audio signal levels for microphones range from -60 dBu to -20 dBu.
These are very low signal levels that often require a preamplifier with at least 60 dB of gain before the signal
can be used with line level equipment.
Use a Mic to Line Preamplifier such as a mic preamp for applications where a microphone must be connected
to a line level audio input on a device such as a switcher or line level mixer. Mic level signals are boosted up to
balanced or unbalanced line level signals, thus allowing a microphone to be directly connected to the Aux/Mix
input on a switcher or mixer.
The front panel of a preamp will usually have a continuous gain control for precise adjustment of input
microphone level. Toggle switches may be provided to engage 48-volt phantom power for compatibility with
condenser microphones, and a low cut filter at 75 Hz to reduce any rumble and wind noise picked up by the
microphone.
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Mixers
The primary role of the mixer is to adjust all mic and line level program source inputs to a desired level and
route the output signal via line level to the signal processor or amplifier.
Program source signals are input to the mixer where they are blended or “mixed.” The output of the mixer is
sent to the signal processing and amplifying stage(s) of the audio signal path.
Mixers versus Switchers
AV installations commonly use a switcher when audio signal selection is to follow the video signal selection,
also known as audio follow.
Switchers can also allow for independent switching of audio signals from their video component. This is
referred to as breakaway. Mixers are chosen over switchers for audio when:
● There are many independent audio sources to be selected independently of video signals.
● The audio will require frequent adjustments by hand or by an operator during a presentation
● There is a great deal of signal processing that must be done to the individual audio signals in
order for them to blend properly
Types of Mixers
Common types of mixers found in AV installations include:
● Rack-mounted automatic mixers – Designed for installations where mics must be
automatically opened or closed and for remote control, not designed for active control. These
can also feature digital signal processing to enhance the audio signal and prevent feedback.
● Console Mixers – Designed for live sound production where many audio elements must be
actively controlled by a sound engineer. These mixers often feature a large number of controls
and faders for signal enhancement, routing and gain adjustments.
● Mixer Amplifiers – Designed for installations where source mixing and amplification must be
housed in a single enclosure to save space or simplify integration.
Mixers and Gain Staging
Gain staging is critical in audio. Gain, or signal strength, must be maintained throughout the audio system to
achieve audio design goals.
If the signal is too loud, or “hot,” in any of the links of the audio signal path, distortion (any undesired change
in an audio signal between the input and the output) will be heard and intelligibility will be lost. Sending a
distorted signal to the amplifier will not only cause it to amplify the distortion with unpleasant results, but
there can be damage to the speakers as well.
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Mixer Levels and Connections
Mixers accept a wide variety of source signal levels and connector types based on the application and the type
of mixer used in the installation. Mixers may accept XLR, RCA, ¼”, and Captive Screw input connections using
mic level or line levels over balanced or unbalanced connections.
Mixer outputs are generally line level and may be on any of the connectors listed above. Professional
installations tend to use balanced captive screw or XLR connections to maintain signal integrity and reduce
occurrences of RFI or EMI.
For example, a Three Channel Audio Mixer cab combine three line level mono audio sources (such as a drum
kit) into a single line level mono output. By accepting balanced or unbalanced input signals, a mixer
accommodates input sources with various impedances. Ideally, you will have:
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Balanced and unbalanced operation for universal compatibility
Gain control for each channel with a wide adjustment range
Compact size for easy installation in a rack or in limited space
Audio Signal Processors
Signal Processors
Many times in AV installations the audio signal must be changed or adapted to better suit the needs of the
installation or application. For example, to make the resulting output sound different we may need to change
the shape of the signal or changes to the electrical characteristics of the signal itself may be required.
To achieve such changes, we need to process the signal using a device which is external to the mixer or
switcher. Devices that manipulate the audio signal in this manner are collectively known as signal processors.
Signal Processing Types
A very simple way of looking at signal processors is to break them into two groups.
Signal processors that change the electrical characteristics include:
● Microphone Preamplifiers
● Summing Amplifiers
● Balanced to Unbalanced Converters
● Crossovers
Signal processors that change the sound characteristics include:
● Graphic Equalizers and Parametric Equalizers
● Compressors, Limiters, and Gates
● Feedback Eliminators
● Effects processors such as Delay, Reverb, Echo
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This is by no means a full listing of signal processors and we will discuss only a few of these.
Signal Processing Application - Equalization
Let’s look at an example of how a signal processor is used.
Integrators often use a real time analyzer, or RTA, to measure the frequencies in a room. If the analyzer
reveals the presence of too many high or too many low frequencies, an equalizer (EQ) can be used to boost or
reduce those frequency groups. The EQ processes the signal to maintain a “flat” audio program where all the
frequencies present are at the same approximate level.
Signal Processing Application - Summing
Here’s another signal processing example.
In this application, a stereo CD player is being connected to a mono 70/100 v speaker system. Now, we could
simply connect the CD player’s left or right channel to the 70/100 v amplifier, but that could cause half of the
stereo program to disappear!
A better solution would be to use a signal processor called a summing amplifier to convert the stereo signal to
a mono signal in which both the left and right signal are “summed” (combined together) without changing
their relative levels. This signal could then be routed to all speakers.
An Audio Summing Amplifier that converts two-channel stereo audio signals to balanced/unbalanced mono
audio signals is ideal for mono multi-speaker PA systems in large churches, halls, arenas, stadiums, and other
environments where stereo sound reinforcement would provide an uneven reproduction to the listener. The
capability output balanced audio would enable longer cable runs and immunity from interference and noise.
Side Chain
Another way to employ a signal process is the side chain. In a side chain, the output signal of a device is
processed by the signal processor and then looped back to the original device as input.
This varies from the more common “in-line” approach of connecting devices in sequence.
The side chain allows for certain signal processors such as reverb units to process “effects loops.” Feedback
eliminators use notch filters to remove frequencies that are ringing out of control can also be employed in this
manner.
Amplifiers
Form Factor
Power amplifiers are generally simple in form. Typically, they feature one or more of the following:
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line level inputs
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speaker level outputs
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gain control
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An integrated power amplifier features multiple inputs with volume and tone controls as well as internal
amplification in a space-saving size and shape, called form factor. A smaller form factor is generally beneficial
in terms of air circulation and heat removal.
Channel size
Power amplifiers can be single channel - mono, dual channel - stereo or multi-channel devices. The design
and selection of an amplifier is based on the needs of the application.
Examples:
● In a classroom with two speakers, a stereo amplifier is a likely choice.
● To drive several dozen speakers to concert level volumes probably requires a very high
powered, multi-channel amplifier.
Output Impedance
In AV, it is not uncommon to encounter a sound system described as an 8-ohm system or a 70/100 v system.
This has quite a bit to do with how the audio signal is carried to the speakers. We will discuss speaker
distribution in the next topic.
For now, it is important to understand that there are amplifiers designed to be used in an 8 ohm or low
impedance sound system (such as a classroom) and others designed for use in high impedance, 70/100 v
sound systems (like those found in department stores).
In some cases, you can find a product that will accommodate both.
Output Wattage
Often amplifiers are described by their output wattage. Such a description might indicate, “20 watts per
channel (rms) into an 8-ohm load.” Generally, this refers to how many watts per channel can be delivered into
an 8-ohm speaker. An acronym for “root mean square,” rms is used in audio to help rate the continuous
power output of an amplifier or input capability of speakers.
The requirements of the room dictate the SPL and speaker coverage needed. The requirements dictate the
size or wattage output of the amplifier required for proper operation of the system.
Amplifier Class
The amplifier class refers to the amplifier output circuit, also called the output stage. An amplifier’s
classification is based on the amount of time the output devices operate during the voltage swing.
The expression “Class X” is used in specifications where X could be A, AB, D, G, or H among others.
Knowledge of an amplifier’s class is useful as each class has inherent advantages and tradeoffs, some of which
can be found in an article by Steve Somers from Extron (see footnote). 1
1 http://eionline.extron.com/forceten/NewLinx/RunThruCluster.php
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Amplifier Connections
Amplifiers feature a variety of input connectors; XLR, ¼”, RCA, captive screw, etc.
Due to the voltage present, output connectors on amplifiers are larger to accommodate the size of speaker
wires necessary to carry speaker level signals.
Captive screw output connectors are larger than their counterparts used as input connectors.
Other connectors can be found on amplifiers including barrier strips, binding posts, and connectors, which
have the ability to carry multiple speaker connections in a single connector.
Amplifier Operation
As we discussed in the section on mixers, it’s critical to maintain proper gain staging of the audio system.
Proper amplifier setup is important to the success of the audio portion of an AV system.
Here are a couple of steps for setting the proper gain on an amplifier:
1.
Start by rotating the amplifier’s gain control to the full attenuation position; all the way down.
2. Then set the other audio signal path element’s output controls to their nominal position.
3. Bring up the gain of the amplifier to the desired sound pressure level in the room using an SPL meter or
to the maximum setting on the amp before distortion occurs.
Note: The amplifier and speaker combination should be chosen based on the required SPL for the room. This
prevents you from “running out of amplifier or speaker” for your application.
Loudspeakers
The last link of the audio signal path is the loudspeaker (speaker). The speaker is a transducer that changes
electrical energy into acoustic energy. The speaker’s driver unit must push air to create sound waves that
replicate the original sound waves captured by the microphone at the start of the audio path. This makes
speaker design and placement critical factors in recreating sound.
The speaker works much like a dynamic microphone in reverse. The electrical signal from the amplifier passes
through a coil of wire called a “voice coil,” suspended within a magnetic field created by a large magnet. The
result is a force which varies with the current and causes the voice coil, with the attached speaker cone, to
oscillate forward and backward to create sound waves.
Other speaker technologies exist but the cone style described here is the most prevalent in AV.
Speaker Components
● Drivers, or the combination of the voice coil and speaker cone, are designed to reproduce
different portions of the audible frequency range (high, middle, low). Tweeters carry high
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Audio Fundamentals
frequencies, mid-range drivers are used for middle frequencies, and woofers carry low
frequencies.
● Crossovers are used to divide the incoming signal into the frequency bands that will be
handled by different drivers in the speakers. These crossovers can either be an internal speaker
component or an external signal processor depending upon the needs of the AV system and
speaker configuration. Crossovers are usually identified as passive (no power required) and
active (power required).
● Enclosures refer to the housing of the speaker and are very important to the sound of the
speaker. Speaker designers take into account the amount of cubic volume inside the enclosure
as well as the ports in the enclosure to help reinforce low frequencies.
Speaker Types
There are many types of speakers, but in general, there are four types of speakers that are commonly used:
● Ceiling speakers install into the ceiling with a flush-mounted grille.
● Surface mount speakers typically feature two or more drivers, a box-like enclosure and mount
on a wall or vertical surface such as a column.
● In-Wall speakers recess into the wall like a ceiling speaker. These rectangular speakers use two
or more drivers and may feature a back box.
● Large format speakers are used in larger venues needing extended audio coverage. These box-
or trapezoidal-shaped speakers can be very large and have many drivers. Groups of these
speakers are used in arrays for greater coverage.
Speaker Configurations
Speakers in AV systems can be made up of different configurations of drivers, crossovers and amplification
needs. These include:
● Single range: These speakers carry only a portion of the overall frequency range. Examples:
Subwoofers carry frequencies below 100 Hz; Paging speakers are designed for speech
reinforcement with a frequency response from approximately 80 Hz to 15 kHz. Frequency
response is the range over which signals are reproduced within a stated amplitude range.
● Two-way: This type of speaker employs a woofer and tweeter to share the workload of a wider
frequency response. This speaker uses a passive crossover network to split the incoming signal
to the two drivers.
● Three-way: Here, a mid-range driver is added to the woofer and tweeter rounding out the
vocal range capabilities of the speaker. This features a more complex passive crossover.
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Audio Fundamentals
Speaker Amplification Types
Two-way and three-way speakers are also called “full-range” speakers. These speakers carry as close as
possible the audible range of 20 Hz to 20 kHz from a single amplifier channel.
Other speaker amplification types include:
●
●
Bi-amplified or “bi-amped” speakers which have two inputs; one for high frequencies and one for low
frequencies
Tri-amplified or “tri-amped” speakers which have three inputs, one for each driver
These speaker configurations require a separate amplifier channel for each driver as well as an active
crossover signal processor between the mixer and the amplifiers in the component chain.
Speaker Selection - Program Speakers
Speaker selection is based on the needs of the project and the role the audio will play.
The goal with Program Speakers is to preserve the audience’s focus on the presentation. In addition, program
speakers are used to maintain a stereo or multi-channel “image” of the original program material.
Program speakers are placed in front of the presentation area and project sound towards the back of the
seating area. These are used for stereo or mono signals; however, in a large room, the “stereo effect” is often
lost.
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Audio Fundamentals
Program Speaker Amplification
Conventional Program Speakers are customarily 8 ohm or 4 ohm speakers. This low impedance transmission
method allows for one or two pairs of speakers to be connected to an amplifier.
These speakers receive a signal from the amplifier that is anywhere from 9 to 29 volts for a 10 to 100-watt
amplifier respectively.
Speaker Selection - Distributed Speakers
Distributed speakers are utilized for even audio coverage throughout the installation as with background
music or voice reinforcement such as those found in department stores.
These systems can provide a more even distribution of sound at a lower volume at each speaker as each
speaker is placed in the ceiling to direct sound downwards.
Distributed speakers almost always carry mono signals for music and paging applications or for speech
reinforcement and mix minus applications. The benefit is a reduction of undesirable reverberation and
increased intelligibility.
Distributed Speaker Amplification
In a distributed speaker application, a “70 volt” (or, more precisely, 70.7 volts) scheme is used (100 volt
systems are commonly used in Europe). In simple terms, this constant voltage method of amplification allows
for more speakers to be placed at a greater distance from the amplifier using a single speaker cable run.
This is because higher voltage with lower current allows for smaller cables and better distribution. This
scheme is comparable to the power company where the voltage is stepped up at the generator, or amplifier in
our case, to run down a single line where each user, the speaker, takes a piece of that voltage with a step
down transformer to convert the signal back to a low impedance signal for use at the speaker.
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