A Shure Educational Publication

SELECTION
AND
OPERATION
AUDIO SIGNAL
PROCESSORS
By Gino Sigismondi
 Selection and Operation of

Ta b l e o f C o n t e n t s AUDIO SIGNAL
Processors

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4

What Are Audio Signal Processors? . . . . . . . . . . . . . . . . . . . . . . . . . 4

What Types of Problems Can Benefit from Audio Processing? . . . . . 5
Feedback . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6

Chapter 1
Types of Audio Processors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
1.1 Volume (Gain) Control . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
1.2 Filters and Equalization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
1.3 Dynamics Processors . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
1.4 Delay . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
1.5 Adaptive Audio Processors . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

Chapter 2
Practical Applications For Audio Signal Processors . . . . . . . . . . . . . . . 23
2.1 Maximizing Gain-Before-Feedback . . . . . . . . . . . . . . . . . . . . . . 23
2.2 Improving Speech Intelligibility . . . . . . . . . . . . . . . . . . . . . . . . . 24
2.3 Sound System Gain Structure . . . . . . . . . . . . . . . . . . . . . . . . . . 26
2.4 Digital Signal Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Reference Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Appendix A: Sound Waves . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Appendix B: Potential Acoustic Gain (PAG) and
Needed Acoustic Gain (NAG) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
Glossary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Shure Product Selection Charts . . . . . . . . . . . . . . . . . . . . . . . . . . . 36
Bibliography . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
About the Author . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

Audio Signal
Processors 3
 Selection and Operation of

AUDIO SIGNAL
Processors

Introduction

For any sound system, the primary goal is good if the three primary measures are not satisfied, any
sound. What, however, constitutes "good" sound? The subjective terms take on even less importance. Speech
three primary measures of good sound are audibility, that is "warm" but unintelligible does the listener little good.
intelligibility, and fidelity. Many factors contribute to the Audio signal processors offer a variety of tools to assist
quality of the sound, including the quality of the sound in optimizing a sound system for audibility, intelligibility,
sources, the sound system, and the room acoustics. and fidelity. While not usually essential for a sound system
The audibility of speech or music at the furthest to operate (i.e., provide high-level sound reinforcement
listener must be sufficient to achieve the desired effect: of low-level sources), audio signal processors can be
usually a comfortable listening level for speech, and more invaluable tools in sound system design. A basic sound
powerful levels for certain kinds of music. These levels system consists of four components:
should be attainable without distortion or feedback. • Input devices (microphones, CD players, etc)
Intelligibility is determined by the signal-to-noise ratio and • Mixers (to combine inputs, control levels,
direct-to- reverberant ratio at the listener’s ear. The "signal" and provide preamplification, if necessary)
is the desired sound source (speech, musical instruments, • Amplifiers
etc.), while the "noise" is ambient sound in the room as • Output devices (loudspeakers)
well as electrical noise produced by the sound system. Audio signal processors are typically employed
Maximum speech intelligibility requires a speech level of within or just after the mixer stage, but before amplification.
at least 20 dB above the noise floor at the listener’s ear. (See Figure 1-1.) A processor can be used at the input
The direct-to-reverberant ratio is determined by the stage, but since most processors are designed to operate
directivity of the loudspeakers and the reverberation with line level sources this is rare. Signal processors can
characteristics of the room. High levels of reverberation be analog or digital, single- or multi-function, stand-alone
can severely degrade intelligibility by making it difficult to devices or integrated with other components in the sound
distinguish the end of one word and the start of the next. system. Most signal processors originated as stand alone
Finally, fidelity of sound is primarily defined by the overall devices designed for a specific purpose. Over time,
frequency response of the sound arriving at the listener’s integration of similar processors into one device became
ear. The frequency range must be sufficiently wide and popular (e.g. compressor/limiters). The development of
relatively uniform in order to provide realistic and accurate audio processors that operate in the digital domain allowed
reinforcement of speech and music. Every component in for further integration, leading to multi-function digital signal
the signal chain contributes to this, and a limitation at any processors (DSP) that combine seemingly disparate
point will affect the fidelity of the entire system. functions into a single unit. Perhaps more importantly, DSP
Other more subjective terms may be applied to good devices offer these functions at a cost that is a fraction of
sound ("warmth", "punch", etc.), but these colloquialisms the purchase price of several individual processors.
are not measurable in any meaningful way. Additionally,

Introduction
4
 Selection and Operation of

AUDIO SIGNAL
Processors

INSTRUMENT
MICROPHONE AMPLIFIER

LOUDSPEAKER

VOCAL MICROPHONE

BOUNDARY MIXER PROCESSOR

MICROPHONE

Figure 1-1: basic sound system

What Types of Problems Can Benefit from effect on the sound. Reverberation can be reduced only by
Audio Processing? absorptive acoustic treatment or structural modification;
To understand the purpose of audio signal processing, electronics cannot remove it. If additional acoustic
it is necessary to examine the problems encountered in treatment is not an option, directional loudspeakers allow
a typical sound system. Note that an audio processor the sound to be "aimed" toward the listener and away from
cannot solve all the potential problems in a sound reflective surfaces. Simply raising the level of the sound
reinforcement system. The most common problems are system will only aggravate the problem by raising the
listed to the on the next page: reverberation level as well. Long reverberation times
The importance of good room acoustics cannot be severely reduce intelligibility. In audio teleconferencing
underestimated. In any room where sound reinforcement systems, this results in a hollow, or "bottom-of-the-barrel"
will be used, excess reverberation times introduce a sound received by the remote site.
myriad of problems that cannot be solved by any audio
processors. Reverberation time is the length of time that a
sound persists in a room after the sound source has Want to know more about proper microphone usage?
stopped. All attempts should be made to keep unwanted Shure offers the following educational guides free
sounds from entering the microphone in the first place. of charge:
The level of desired sound at the microphone should be
• Microphone Techniques for Studio Recording
at least 30 dB above any ambient sound picked up by
the microphone. Proper microphone placement (a full • Microphone Techniques for Live Sound
discussion of which is beyond the scope of this publication) Reinforcement
is also crucial. A good rule of thumb: always keep Visit shure.com or contact your local Shure
microphones as close as possible to the sound source. office (see back cover) to request your
Once sound energy is introduced into the acoustic complimentary copies.
space by the loudspeaker, processing no longer has any

5
 Selection and Operation of

AUDIO SIGNAL
Processors

Feedback
Problems: Remedies: Feedback is characterized by a sustained,
Feedback Parametric Equalizer/ ringing tone, which can vary from a low rumble to a
Automatic Mixer/ piercing screech. Echoes and reverberation caused
Feedback Reducer by room acoustics, as well as ground buzz and other
Poor tone quality (subjective) Graphic equalizer extraneous noises, are not the same thing as feed-
Sound source too loud Compressor/Limiter/AGC back, and cannot be cured in the same manner.
Sound source too quiet AGC Feedback occurs whenever the sound entering a
microphone is reproduced by a loudspeaker, picked
Varying signal levels Compressor/Limiter/AGC
up by the microphone, and re-amplified again and
from multiple sound sources again. The familiar howl of feedback is an oscillation
Unwanted noise Noisegate/Downward expander that is triggered by sound entering the microphone.
Unexpected transients Compressor/Limiter/No overshot The easiest way to (intentionally) create feedback is to
("Look-ahead") Peak Limiter point a microphone directly into a loudspeaker.
Comb filtering Automatic Microphone Mixer Placing the microphone too close to the loudspeaker,
too far from the sound source, or simply turning the
due to open microphones
microphone up too loud exacerbates feedback
Frequency response Delay problems. Other contributing factors are too many
anomalies due to open microphones, poor room acoustics, and uneven
misaligned loudspeakers frequency response in either the microphones or
Poor intelligibility Parametric Equalizer/ loudspeakers.
Automatic Microphone Mixer The single easiest way to reduce feedback is to
move the microphone closer to the desired sound
Acoustic echoes Acoustic Echo Canceller
source. Additionally, using a directional microphone
(in teleconferencing systems) (cardioid, supercardioid, etc.) will slightly increase the
Distortion Compressor/Limiter amount of gain-before-feedback. Reducing the
(due to wide dynamic range) number of open microphones with an automatic
mixer will also improve the situation. Try to keep
Problems that cannot be solved by audio processing: microphones and lou speakers as far away from each
other as possible. Lastly, acoustically treat the room
• Echoes because of poor room acoustics
to cover hard, reflective surfaces such as glass,
• Poor sound due to excess room reverberation times marble, and wood. Realize, though, that in certain
• Feedback caused by operating beyond the limits of PAG rooms long reverberation times may be desirable,
(see Appendix 2) such as a house of worship used for acoustic
music performance.
• Noise (amplifier hiss, ground buzz, etc.) due to improper If the system has been designed with careful
system setup consideration of these factors and feedback is still
• Distortion due to improper gain structure an issue, an automatic feedback reducer can be
used to flatten the response at problem frequencies.
These devices are discussed in Section 1-5.

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 Selection and Operation of

AUDIO SIGNAL
Processors

C H A P TE R ONE Filters and Equalization

Filters are signal processors that affect frequency
TYPES OF AUDIO PROCESSORS balance. At a basic level, filters are used to attenuate or
boost the level of specific frequencies or frequency ranges.
Designed originally to compensate for frequency-
Volume (Gain) Control dependent loss in telephone lines, some form of
Although often overlooked as an audio processor, a frequency- dependent filtering (or equalization) is found in
simple volume (or gain) control fits the definition. Volume all but the most basic of sound systems. The simplest form
adjustments can be made at several points within the of filter is the tone control, basically a filter that attenuates
sound system, from the microphone inputs on the mixer all high frequencies above a predetermined frequency.
the way to the inputs of the power amplifiers. Volume Equalizers are typically characterized by combining several
levels are typically manipulated in one of two ways: filter sets to offer more precise frequency response
continuously variable adjustments, such as those made by shaping. Historically, filters were passive devices capable of
rotary potentiometers or faders, or fixed attenuation such as attenuation only. The frequency range and amount of
that provided by a pad. attenuation were achieved with capacitors, inductors, or a
If adjusting a volume control adds amplification to the combination of both. Favorably, passive filters do not
audio signal, it is said to be providing gain. The volume require power and do not generate noise. The large size
control that adjusts the amount of amplification added at a and expense of discrete components, however, precludes
mixer’s microphone input is sometimes referred to as a the ability to develop equalizers with larger numbers
gain (or trim) control, since the volume potentiometer is of filters and more precise control of frequency and level.
controlling the gain of the microphone input’s preamplifier. Active filters allow for fast, easy tuning and the ability to
The function of this gain control is to match the input add gain, using smaller components at lower cost. Tone
sensitivity of the device to the level from the source. controls employing active filters can be found on even the
A second type of volume control acts as an most inexpensive home stereo systems. In this scenario
attenuator, basically a continuously variable resistor that there are typically two controls, treble and bass, which
adjusts the amount of signal allowed to pass through it. correspond to filters that affect low frequency and high
No additional gain is provided by the volume control. The frequency response. Since they are active, these tone
volume control on an electric guitar is an attenuator. These controls are capable of cut or boost.
devices are often referred to as passive volume controls, Simple filters that affect a broad range of frequencies
since they do not require any power. Occasionally, a are divided into four types: high pass, low pass, band pass,
volume control will combine attenuation with gain. Faders and band reject. High pass filters, as the name implies,
on a mixing console typically provide attenuation below the allow high frequencies to pass, and low pass filters do the
"0" indication, and gain above that point. same for low frequencies. It is often more convenient to
Pads allow input stages to accommodate a variety of think of these filters in terms of the frequencies that they
signal levels. Microphone inputs typically feature an input cut instead. High pass filters are also known as low cut
attenuation pad of some kind to reduce the sensitivity of the filters, and low pass filters are known as high cut filters, but
input beyond that of the preamplifier gain control,
typically by 20 dB. A 50 dB pad is required for microphone
inputs that are designed to accept either microphone or line
level. The output stage of various devices can also employ
pads, usually to prevent overloading of the input stage of the
next device in the signal path. Care should be taken to use
pads only when necessary. For example, using a 20 dB pad
on a microphone input that does not need additional
attenuation will require additional gain be added by the
preamplifier, which adds more noise to the audio signal.
Low Cut: -6dB/octave below 125 Hz
While volume controls are the simplest of all audio High Cut: -6dB/octave above 2 kHz
processors, they often the most misused. Correct
calibration of the various volume controls in a sound
system is known as proper gain structure. (See Section Figure 1-2: low cut and high cut filters
2-3: Gain Structure.)
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 Selection and Operation of

AUDIO SIGNAL
Processors

their function is the same and these terms can be used

interchangeably. (See Figure 1-2.) Low and high cut filters
have an associated slope that defines how rapidly output
declines below (or above) the filter frequency. Slope is
typically defined in dB/octave. The span of an octave re-
lates to a doubling (or halving) of frequency, for example,
50 to 100 Hz or 5 kHz to 2.5 kHz. A 6 dB/octave low cut
beginning at 100 Hz, therefore, translates into 6 dB less Figure 1-4: graphic equalizer
output at 50 Hz, and 12 dB less at 25 Hz. Typical slopes
are 6, 12, 18 and 24 dB/octave. The slope also has an
associated rolloff characteristic, most commonly Bessel, Graphic Equalizers
Butterworth, or Linkwitz-Riley. See "Crossovers" for more The most common equalization tool for sound
information on filter slope types. The frequency that de- reinforcement is the graphic equalizer. A typical graphic
fines a filter is usually stated at its 3 dB down point (A low equalizer consists of a bank of sliders (or faders),
cut filter set to 100 Hz is actually 3 dB down at 100 Hz). corresponding to specific frequencies, which can cut or
A band pass filter allows only a certain range of frequencies boost the chosen frequency. (See Figure 1-4.) The center
to pass (called the passband). The same effect can be frequencies of these filters are identical for all graphic
achieved by using a low cut and high cut filter together. equalizers, regardless of manufacturer, because they are
The result is the similar to boosting the frequencies that defined by ISO (International Standards Organization)
comprise the pass band. A band-reject filter reduces a documents. Since the position of the sliders roughly
range of frequencies. represents the amount of cut or boost, this type of
equalizer offers an approximate visual representation of
the frequency response alteration created by the equalizer,
hence the term "graphic." The actual width of the filters,
though, is wider than what is implied by the graphic
equalizer, and the combined response of multiple filters
will most likely be much more dramatic. Also, note that this
is only the response imposed on the audio signal by the
equalizer, not the actual frequency content of the audio
signal. For example, if the audio signal is already 2 dB up
at 2 kHz, using the EQ to add another 3 dB of boost at 2
Low Shelf: -10dB below 125 Hz
kHz results in a total increase of 5 dB. However, the
High Shelf: -10dB above 2 kHz
graphic equalizer only reflects the 3 dB boost. An analysis
of the total frequency response of the sound system
Figure 1-3: shelving equalizers requires a measurement device, such as a Real Time
Analyzer (RTA).
The number of filters available on a graphic equalizer
A further subdivision of high and low cut filters is can vary from as few as 5 (a 5-band graphic equalizer) to 31
the shelving equalizer. (See Figure 1-3.) Rather than (a 31-band graphic equalizer) or more. On a graphic
continuing to decline at a certain dB/octave rate, equalizer, there is a direct correlation between the number
attenuation flattens out at a certain fixed level, forming what of filters and the bandwidth of each filter. In general, more
appears as a "shelf" when observed on a frequency filters offer more precise control because the range of
response chart. Unlike low or high pass filters, most frequencies that each filter affects is smaller. The bandwidth
shelving equalizers allow for boost as well as cut. of each filter is also a defining characteristic of the equalizer.
Consumer bass and treble controls are typically shelving Typical classifications are one octave, 2/3-octave, or 1/3-
equalizers where increasing the amount of cut or boost octave. Higher bandwidth filters, such as 1/6-octave, exist
often changes the frequency at which the EQ begins to but are rarely encountered. 1/3-octave graphics are the
take effect. More advanced shelving equalizers allow the most common, since they offer a fairly precise level of
user to select the frequency, the amount of cut, and control with a manageable amount of sliders. The audible
occasionally the rate of cut (again in dB/octave). frequency range requires 30 or 31 1/3-octave filters; a

8
 Selection and Operation of

AUDIO SIGNAL
Processors

Parametric Equalizer
The parametric equalizer offers a much greater degree
of control than a graphic equalizer by giving the user more
parameters to adjust. In addition to cut or boost of specific
frequencies, a parametric equalizer also allows adjustment
of the center frequency and bandwidth of the filter.
(See Figure 1-6.)

combining filters

Low: 1/40 octave, -18dB @ 30Hz

Middle: 1/3 octave, -18dB @ 300Hz
High: 1 octave, -18dB @ kHz

Figure 1-6: parametric filters

non-combining filters

Figure 1-5

1/6-octave graphic requires at least 60 1/6-octave filters.

Lower bandwidth devices, like one octave or 2/3-octave, are
broadband in nature and typically used for overall tone
shaping rather than precise control. A one-octave graphic
equalizer usually has 7 or 8 filters, a 2/3-octave has 15. Note
frequency
that some older equalizers use a rotary potentiometer rather
than a vertical fader. This device is still termed a graphic
equalizer, though the visual representation of frequency Figure 1-7: sweepable filter
response created by pointers on knobs is far less intuitive.
Graphic equalizers can further be defined as combining The center frequency is defined as the point at
or non-combining. (See Figure 1-5.) If a frequency that which the maximum amount of boost or cut occurs. The
needs to be attenuated lies between two 1/3-octave band bandwidth, as stated above, indicates the actual range of
centers, those two filters can be cut to reach that frequencies affected by the filter. A semi-parametric (or
frequency. In a non-combining equalizer, the area of sweepable) equalizer allows selection of center frequency
overlap between two filters will be somewhat higher in level, and boost or cut, but the bandwidth is fixed. (See Figure
requiring excessive cut to adequately reduce the level of 1-7.) This is a common feature on more affordable mixing
the desired frequency. A combining equalizer, however, consoles. Most modern mixing consoles with an EQ section
has a smoother transition between adjacent bands, have at least one sweepable filter (usually for midrange).
requiring less overall cut to reach the same level of More advanced mixers include two or more bands of
attenuation at said frequency. Additionally, there is less fully-parametric EQ. Concentric potentiometers are often
"ripple" in the overall frequency response, whether boosting used to help save real estate on the console, typically to
or cutting. Due to this smoother response, graphic control frequency and bandwidth. Stand-alone, analog
equalizers with combining filters are preferred for sound parametrics also are limited by space requirements, since
reinforcement applications. Generally, graphic equalizers each band requires three controls. They are typically
use combining filters, unless otherwise specified. available with 5 to 7 filters. Digital parametric equalizers, on

9
 Selection and Operation of

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Processors

the other hand, are limited only by the processing

capabilities of the device. More processing power means
more filters. Many digital signal processing (DSP)
equalizers allow the user to deploy the filters using a display
that shows a graphical representation of the filters. This
type of EQ combines the visual advantage of a graphic
equalizer with the more precise control of a parametric.
The main advantage to adjustable bandwidth is less
effect on adjacent frequencies when applying corrective
equalization. (See Figure 1-8.) For example, in a sound
system where 500 Hz needs to be attenuated by 6 dB, using 1/3 octave graphic EQ:
a 1/3-octave graphic equalizer results in approximately 3 dB -3 dB @ 500 Hz, -6 dB @ 1000 Hz, -3 dB @ 2 kHz
of attenuation at 400 Hz and 630 Hz. By using a parametric
equalizer and reducing the bandwidth to 1/10 of octave, the
same frequencies are barely affected. Conversely, employing
a wider bandwidth allows several adjacent frequencies to
be intentionally cut (or boosted) with only a single filter.
Different devices express bandwidth using one of
three measures: fractions of an octave, Q, or number of
Hertz. (See Figure 1-9.) At 1 kHz, a filter with 3 dB of cut
and a 1-octave bandwidth corresponds to a Q of 1.41 and
covers approximately 709 Hz. For the purpose of defining
Q, bandwidth is measured from the 3 dB up or down 1/10 octave parametric EQ:
points (depending on whether there is boost or cut). -3 dB @ 500 Hz, -6 dB @ 1000 Hz, -3 dB @ 2 kHz
Dividing the center frequency by this bandwidth in Hz
gives the Q, which stands for "Quality Factor." Q gives
an indication of how tightly the filter is focused near the Figure 1-8
center frequency. In this example, the –3 dB points for a
one-octave filter are approximately 645 Hz and 1355 Hz,
a difference of 710 Hz, therefore: Applications Tip:
Graphic EQ vs. Parametric EQ
Q = 1000 Hz/710 Hz Many audio professionals differentiate the two main
Q = 1.41 types of equalizers in this way:

Note that when determining Q, the 3 dB points are defined Parametric EQ: The "problem solver." Use the
relative to the peak or trough, not the audio pass band. parametric equalizer to correct response peaks in
This sometimes leads to confusion, because the effective the sound system. Microphones and loudspeakers,
bandwidth of a filter is sometimes also defined as the in particular, introduce many irregularities into the
difference in frequencies at 3 dB points relative to unity overall frequency response. With the appropriate
gain, rather than the center frequency. Unfortunately, the audio measurement device, these irregularities are
meaning of the term bandwidth can change with context. easily identified and corrected by a parametric
While significantly more powerful than graphic equalizer.
equalizers, parametrics do require a greater level of
understanding on the part of the user, particularly when Graphic EQ: The "tone control." Use the graphic
adjusting bandwidth. A graphic equalizer provides simple equalizer to make broad changes to the sound
operation for general tone shaping and on-the-fly tweaks. system’s frequency response. Once the parametric
With proper application, the parametric equalizer is a equalizer has flattened the frequency response
powerful tool for surgical adjustment of frequency of the system, the graphic equalizer serves as a
response anomalies and problematic feedback frequencies. tool for subjective shaping to achieve "pleasing"
Also, note that a parametric filter can be adjusted to sound quality.
duplicate the function of an individual graphic EQ filter.
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 Selection and Operation of

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for high frequency transducers, such as compression

drivers, that have very little response below 1 kHz.
A crossover divides the audio signal into two or more
frequency bands. (See Figure 1-10.) The frequency at
which the division occurs is the crossover frequency.
A crossover can be either active or passive, and is
described using parameters similar to those found in low
pass and high pass filters, namely: frequency, slope, and
filter type. The most common filter types found in
crossovers are Bessel, Butterworth, and Linkwitz-Riley, with
Q=1.41 slopes of 6, 12, 18, or 24 dB per octave. While providing
••• a minimal amount of phase shift, a 6 dB per octave slope
results in significant overlap between the frequency ranges
fed to the loudspeaker components, and may not provide
enough protection for high frequency drivers. Historically,
645 Hz 1,000 Hz 1,355 Hz the 18-dB per octave Butterworth filter has been a sound
(-6 dB) (-6 dB) system standard, though the 24 dB/octave Linkwitz-Riley
crossover has eclipsed its popularity. Besides the
Figure 1-9: 1 octave filter, -9 dB @ 1kHz advantage of minimal overlap at the crossover point, the
Linkwitz-Riley filter provides an in-phase relationship at the
Crossovers crossover outputs.
To understand the purpose of a crossover, it is helpful
to understand the frequency response characteristics of a
typical loudspeaker. When measured with a pink noise
source, it becomes apparent that any given loudspeaker
can, at best, only reproduce a decade of frequency
response without compromise. Whereas an octave 20 kHz
20 Hz
represents a doubling of frequency, a decade (from the
Greek deca) represents a factor of ten. The range from 2-way crossover
100 Hz to 1 kHz is a decade. Therefore, to accurately
reproduce the entire audible range for humans requires at
least three loudspeakers, each theoretically optimized for
the following frequency ranges:
20 – 200 Hz
200 – 2 kHz
20 Hz 20 kHz
2 kHz – 20 kHz
In reality, most loudspeakers will not have exactly 3-way crossover
these specifications, due to compromises that must be
made in the design of loudspeaker systems. Very few Figure 1-10
sound systems actually produce much output below 40
Hz, especially for speech applications. Therefore, two-way A passive crossover, basically a combination low-pass
loudspeakers with acceptable fidelity are possible, and and high-pass filter, is typically the last processor encountered
quite popular. The frequency response of this type of before the loudspeaker; often integral to the design of the
loudspeaker typically extends only as low as 80 Hz. In this loudspeaker itself. Passive crossovers do not require power
case a subwoofer could be used to provide extended low to operate and are normally invisible to the user. The
frequency response, if necessary. crossover frequency is fixed, optimized by the designer for
Furthermore, since loudspeaker drivers are generally that particular loudspeaker. Passive crossovers are often
optimized to reproduce a particular band of frequencies, a referred to as high-level, since they operate with speaker-level
given loudspeaker may be subject to damage if driven with signals. Unfortunately, the full output of the amplifier may not
a high-level signal that contains a frequency range it was be delivered directly to the loudspeaker since some power if
not designed to handle. This situation is particularly true absorbed by the crossover. Also, the electrical components
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 Selection and Operation of

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Processors

pink noise - narrow dynamic range

Figure 1-11: 3-way active crossover

required for passive crossovers may dictate physically large

devices, and production tolerances in these devices can vary,
affecting the accuracy of the crossover.
The active crossover, also known as a low-level or
electronic crossover, provides several significant advantages
over the passive design. These advantages include uncompressed speech - wide dynamic range
increased amplifier headroom and more efficient use of
available amplifier power. Low frequencies tend to place the
greatest demands on amplifier power. If a single amplifier is
used to drive a multi-way loudspeaker, any distortion due to
overload at the amplifier input is reproduced by every
transducer in the system. This situation can result in audible
clipping, especially of high frequency material. By dividing
the audio signal with an active crossover, a separate
amplifier is used for each frequency band, (see Figure
1-11) thereby reducing the likelihood of audible distortion. compressed speech
If low frequency energy causes the woofer amplifier to clip,
the other amplifier, and the loudspeaker connected to it, will
not be affected. This is known as a bi-amplified sound Figure 1-12
system. Similarly, a three-way crossover feeding three power
amplifiers is called a tri-amplified system. If clipping occurs
in the low frequency amplifier, the higher frequency Dynamics Processors
harmonics created by the clipping are reproduced only by The term dynamics refers to the wide variations in
a woofer that has very low output at high frequencies, thus signal levels commonly encountered in sound systems.
reducing the audibility of the distortion. The use of active Every sound has a dynamic range, defined as the
components also offers smaller size and more repeatable difference between the loudest and quietest levels. A signal
production due to better tolerances. that varies greatly in level, such as speech, is described as
Quite often, a sound system combines elements of having a wide dynamic range. A noise source (such as
both passive and active crossover networks. These types pink noise) that is held to a consistent level has a narrow
of systems typically use an active crossover to provide a dynamic range. (See Figure 1-12.) Music sources typically
separate subwoofer output for low frequencies, while a fall somewhere in between speech and noise, although
passive crossover in a two-way loudspeaker divides some music sources can have a dynamic range much
mid- and high frequencies. This could be described as a greater than speech. Used properly, a dynamics processor
three-way, bi-amplified sound system. can manipulate the level variations in a signal to increase
Most active crossovers allow for control of crossover audibility and reduce undesired noise in a sound system.
frequency and level at each output. DSP-based crossovers Common dynamics processors include compressors,
typically offer greater adjustment, providing the user with limiters, expanders, noise gates, and speech levelers
selectable filter slope, filter type, and polarity reversal. (a.k.a. automatic gain controls.)
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 Selection and Operation of

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uncompressed vs. compressed hard knee vs. soft knee

Figure 1-13 Figure 1-14

Compressors allows the user to boost the overall level of the signal, yet
Perhaps the most commonly encountered dynamics keeps loud signals from getting "too loud" and causing
processor, a compressor reduces (or "compresses") the distortion further down the audio chain - or simply
dynamic range of an audio signal. A compressor functions annoying listeners. The compressor itself does not boost
by reducing the level of all signals above a user-defined point lower signal levels, but simply allows them to be perceived
(the threshold), by a specified amount. (See Figure 1-13.) closer in level to louder signals.
A ratio defines the amount of reduction that occurs above Other compressor settings include attack, release, and
the threshold. A ratio of 2:1, for example, will allow an audio decays. A compressor’s attack time relates to how quickly
signal to exceed the threshold by only half as much as what the compression takes effect once the signal exceeds the
it would have without compression. Assuming a threshold threshold. Shorter attack times offer greater transient control.
setting of 0 dB, a +10 dB signal is output at +5 dB. Similarly, Longer attack times generally sound more natural, and are
a 4:1 setting will reduce the output by one-quarter of the often employed in musical applications. Too long an attack
original signal level. This reduction limits variation between time can cause the compressor to miss signals that
the lowest and highest signal levels, resulting in a smaller otherwise should be compressed. Release refers to the time
dynamic range. A common myth concerning compressors it takes for the compressor to return the signal level to its
is that they make quiet signals louder. While this may be the original value after the level drops below the threshold. Too
perceived effect, reducing the dynamic range of a signal short a release time can result in "pumping" and "breathing"
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 Selection and Operation of

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with signals that have rapid level changes. Too long a release
time can render quieter passages inaudible since gain
reduction is still being applied to the audio signal.
A compressor’s knee is the point where the signal
crosses the threshold. Standard compression schemes
reduce the signal by the same amount once the signal
has passed the threshold. This is known as hard knee
compression. Some compressors allow the user to select
soft knee compression instead, where the onset of
compression near the threshold occurs more gradually than
the more aggressive hard knee compression. (See Figure
1-14.) The compression ratio near the threshold is actually
less than specified. Audibly, soft knee compression creates
a more gradual transition from uncompressed to com-
pressed signals, making the compression less noticeable.

Applications Tip: Compressor vs. Loudspeaker Figure 1-15: limiter

Here is a common complaint made by the owner of Limiters

a damaged loudspeaker: "How could I have blown a A limiter functions in much the same way as a
loudspeaker? I have a compressor!" Unfortunately, compressor, differentiated more by its application than its
while compressors and limiters help prevent audio operation. Similar to a compressor, a limiter also reduces
transients from causing clipping or possibly damaging signals that pass a threshold by a certain ratio. The ratios
a loudspeaker, high-level transients are not the used by limiters, though, tend to be much greater than
only cause of damaged loudspeakers. In fact, over- those used by compressors. Typical limiter ratios can range
compression of the audio signal can contribute to anywhere from 10:1 to ∞:1 (infinity:1, where the threshold
premature loudspeaker failure. setting dictates the maximum signal level). (See Figure
It is standard practice to use an amplifier with 1-15.) The goal of a limiter is usually system protection, by
a power rating at least twice the loudspeaker’s preventing transient audio peaks from causing distortion
continuous power rating (e.g. use a 200 watt amplifier further up the audio chain or, worst case, damaging
for a 100 watt loudspeaker). The extra headroom loudspeaker components. Typically, limiter threshold
afforded by the larger amplifier allows for peaks in the settings are also much higher than on compressors; low
program material to be delivered to the loudspeaker threshold settings on a limiter lead to excess compression.
without clipping. The majority of the amplifier power Limiters also share other parameters with compressors,
goes largely unused since the average level of an including attack and release.
uncompressed audio signal is considerably lower than To further illustrate the difference between compressors
the peaks. Highly compressed signals have an average and limiters, imagine someone jumping on a trampoline in
level much closer to the peak level. If the level of the a low-ceilinged room. The up and down motion of our
compressed signal is raised to take advantage of the trampoline artist represents the varying voltage of an audio
additional amplifier power (thereby making it louder signal; the ceiling represents the threshold of either the
for the audience), the average power delivered to the compressor or limiter. If the ceiling is made of thin rubber, it
loudspeaker may be more than the continuous will give when the trampoline artist hits it, allowing the person
power rating of the loudspeaker, overheating the to pan beyond the ceiling (or “threshold”). But not by as
loudspeaker’s voice coil and causing failure. much as he would if there were no ceiling there at all. A hard
As with all audio processors, using a compressor plaster ceiling, however, is analogous to a limiter. When the
does not eliminate the need for proper system artist hits the ceiling, no further travel beyond it is possible.
operation. Though a compressor or limiter is essential In practice, the operation of a limiter is not quite this
for reducing transient peaks, excessive compression absolute. A standard limiter cannot have an attack time of
is the enemy of the loudspeaker. zero. An unexpected, loud transient could pass through the

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 Selection and Operation of

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output before the limiter circuitry has a chance to act on it.

To provide maximum loudspeaker protection, a limiter needs
the ability to anticipate peaks. DSP-based limiters can
accomplish this by inserting a small amount of delay
(normally 1ms) into the signal path. By delaying the signal,
the limiter is able to act on transient peaks that would
otherwise escape before the limiting occurs. The attack time
of this limiter is effectively zero. These types of limiters are
commonly known as look-ahead or peak stop limiters. They
are often the last device in the signal path before the power
amplifier and are typically assigned a very high threshold.
Since the nature of the look-ahead limiter is last-resort
system protection, such a limiter may have less than
pleasing audio quality. A high threshold assures the limiter
low level unprocessed signal will not affect the audio signal unless absolutely necessary.

Expanders and Noise Gates

An expander, as the name implies, functions as the
reverse of a compressor by expanding the dynamic range
of an audio signal. An expander works by raising signals
that pass above the threshold and, in some cases, by also
attenuating signals that are below the threshold. As in a
compressor, the ratio dictates how much gain is added to
the signal. A downward expander, conversely, only
reduces signal levels below the threshold, again using a
ratio. The same set of adjustments (attack, decay) also
apply to expanders. The applications for true expanders
in sound systems are limited. They are often used in
conjunction with a compressor to form a compander, a
circuit commonly used in noise reduction systems and
expander processing – smoother transition wireless microphone systems. Compression is normally
employed in the transmitter of a wireless system to
prevent the radio frequency signal from deviating beyond
(usually government imposed) bandwidth limitations.
An upward expander in the receiver serves to restore the
original dynamic range of the audio signal. In a noise
reduction system, frequency-dependent companding is
used to reduce unwanted hiss and tape noise. For sound
system applications, the downward expander can be used
to reduce unwanted background noise when there is no
program material present. A system with multiple open
microphones benefits greatly from downward expansion.
A downward expander with a ratio setting of
∞ :1 becomes a noise gate. (See Figure 1-16.) When signal
level drops below the threshold, the output is essentially
turned off (or "gated"), also preventing build-up of
noise gate processing – more abrupt transition undesired noises. The audible effect of a noise gate can
be somewhat more disturbing than a downward expander,
since the transition to the "off" state is more abrupt, audibly
Figure 1-16: limiter similar to manually muting or un-muting an audio channel.

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 Selection and Operation of

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fulfilling a dynamics processor’s purpose of increasing

audibility. A typical AGC has a hinge point. Gain is added
to signals below the hinge, while signals above the hinge
are reduced in level. Another way to think of the hinge is
as the unity gain point, where no addition or subtraction of
gain occurs. The hinge point should be set at the desired
output, or target level, for the given sound source.
The threshold sets the level where the AGC begins to take
input

effect. Signals below the threshold are not processed. (See

Figure 1-17.) Similar to the compressor, the attack setting
adjusts the speed at which the AGC takes effect, and
decay sets how long the AGC takes to release. AGCs
typically use longer attack and decay times than other
dynamics processors, in part to emulate the reaction
output time it would take for a human to make similar gain
adjustments. The AGC is one of the only processors that
AGC compresses above hinge point (H) vs. can raise the volume of the sound system to compensate
AGC boosts below hinge point for soft talkers. To use an AGC, the sound system must
have high enough gain-before-feedback to accommodate
the maximum gain setting of the AGC.

Applications Tip:
Use AGC to compensate for different talkers.
input

Automatic gain controllers tend to work best when

used to compensate for differences in level from
various talkers, rather than from a single talker.
Attempting to level a single talker requires relatively
short attack, hold, and release times to create a
noticeable effect. These shorter times can lead to
output undesirable pumping and breathing in the audio
as the AGC continuously raises and lowers the
Figure 1-17 gain to keep up with the rapidly changing levels
characteristic of speech signals. When used to make
The downward expander sounds more like a rapidly gain adjustments for different talkers, an AGC with
raised (or lowered) fader – a much less jarring transition. longer attack, hold, and release times results in
The terms noise gate and expander are often used smoother transitions and less false triggering.
interchangeably, since many noise gates have an adjustable
ratio rather than solely infinite attenuation. The gate circuit
found in some automatic mixers allows the user to select an Delay
"off-attenuation" setting that uses a fixed amount of gain A third type of audio signal processor works in the time
reduction, such as –15 dB or ∞(off), rather than a ratio. domain, by delaying the incoming audio signal by some
user-defined amount. The primary function of a delay unit
Automatic Gain Control (Speech Leveler) in sound systems is loudspeaker alignment, either to align
A unique case of dynamics processor, the automatic drivers within the main loudspeaker array or align remote
gain control (AGC) either adds or reduces gain, depending speakers with the main PA. Within a given loudspeaker, the
on the strength of the incoming signal. The term speech individual drivers are often physically offset, causing phase
leveler more accurately describes the function of the anomalies due to the differences in time arrival from the
AGC. A properly adjusted AGC should compensate for drivers. In a sound system where every driver is mounted in
differences in level between loud and soft talkers, again its own cabinet, this problem can be corrected by moving
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 Selection and Operation of

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100 ft.
Delay = 1000 (100/1130) = 90 ms

Figure 1-18: under balcony loudspeaker delayed to arrive with main loudspeaker

the boxes until the drivers are aligned. In most two- or have an effect on the speed of sound in air. Delay times
three-way loudspeakers, the drivers cannot be moved. A few may need to be adjusted by a few milliseconds to
milliseconds of delay applied to the "forward-mounted" driver compensate. DSP-delays can usually calculate delay times
are usually sufficient to restore proper alignment. Note that if the required distance is known, and most algorithms are
this method of alignment requires a bi-amplified system with able to take air temperature into consideration. In general,
an active crossover, since the signal for each individual driver the speed of sound increases as the temperature rises.
must be delayed independently.
In larger sound systems, delayed loudspeakers are Adaptive audio processors
used to provide additional coverage to remote areas. (See Adaptive audio processors perform real-time,
Figure 1-18.) Larger houses of worship and theaters will automated functions to optimize sound systems, ideally
often have loudspeakers mounted above or under without the intervention of an operator. Three of the most
balconies. Outdoor concerts sometimes use delay towers. commonly employed adaptive processors are automatic
Since the distance between the main PA system (which is microphone mixers, feedback reducers, and acoustic echo
typically mounted on or near the stage) and the remote cancellers.
loudspeaker is significant, the signal sent to the remote
loudspeaker must be delayed. Without delay, the audience Automatic Microphone Mixers
will experience a degradation of sound quality that, Automatic microphone mixers, also known as
depending on the distances involved, could range from voice-activated or sound-activated microphone mixers,
comb filtering to an audible echo. Use the following formula have one fundamental function: to attenuate (or reduce in
to determine the proper amount of delay: level) any microphone that is not being addressed, and
Delay (milliseconds) = 1000(D (feet)/1130) conversely, to rapidly activate any microphone that is
The speed of sound varies with environmental addressed by a talker. The operation of a well-designed
conditions, but 1130 feet per second is commonly used automatic mixer should be transparent to the audience
in calculations. If D = 100 feet, the required delay is 90 of the sound system. In general, any speech sound
ms. Delaying the signal by an additional 10 ms or so may reinforcement system that uses four or more microphones
help increase the perception that the sound is originating should employ an automatic mixer. To fully understand the
from the stage and not the remote loudspeaker. This advantages of an automatic mixer, it is necessary to
approach takes advantage of the precedence effect, a examine in some detail the audio problems caused by
psychoacoustic phenomenon in which listeners perceive multiple open microphones. These problems are:
sound as coming from the direction of the first sound 1. Excessive pickup of background
arrival, even if it is somewhat lower in level than a sound noise and reverberation
that arrives a short time later. Keep in mind that air 2. Reduced gain-before-feedback
temperature, humidity, and elevation above sea level all 3. Comb filtering
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Shure SCM810 Automatic Microphone Mixer

The first problem of multiple open microphones is the NOMA circuit helps prevent feedback. Assuming all
excessive pickup of background noise, which adversely microphones are equidistant from loudspeakers, an
affects the audio quality of the sound system. Consider a automatic mixer ensures that if there is no feedback with
city council with eight members and eight microphones. one microphone open, then there will not be any feedback
For this example, only one member is talking at a time. even if all the microphones are open.
If all eight microphones are open when only one Comb filtering is phase cancellation that occurs when
microphone is needed, the audio output will contain a single sound source is picked up by more than one
the background noise and reverberation of all eight microphone at different distances from the source, and
microphones. This means the audio signal will contain those signals are combined at the mixer. Since sound
substantially more background noise and reverberation travels at a finite speed, the resultant frequency response
than if only the talker’s microphone was open. Speech of the combined microphone signals is considerably
clarity and intelligibility always suffer as background noise different from that of a single microphone. The frequency
and reverberation increase. In the city council example, response chart of the combined signals resembles the
the audio output of eight open microphones would contain teeth of a hair comb, thus the name. (See Figure 1-19.)
9 dB more background noise and reverberation than a The aural result sounds hollow, diffuse, and thin. An
single open microphone. To the human ear, the noise automatic mixer significantly reduces comb filtering by
would sound roughly twice as loud when all eight keeping the number of open microphones to an absolute
microphones were open. minimum. Certain models of automatic mixers further
In addition to only activating microphones that are reduce comb filtering by employing a circuit that will only
being addressed, an automatic mixer uses a NOMA allow one microphone to turn on for a given sound source.
(Number of Open Microphones Attenuator) circuit, or Most popular automatic mixers belong to one of two
equivalent, to help minimize the build-up of background categories, either some form of gated mixer or a gain-
noise and reverberation. This circuit proportionally reduces sharing automatic mixer.
the overall output of the mixer whenever the number of
open microphones increases. A well-designed automatic
mixer maintains a consistent level of background noise
and reverberation, regardless of how many or few
microphones are active.
The NOMA circuit also plays a major role in controlling
the second major problem with multiple open
microphones, reduced gain-before-feedback. Acoustic
feedback, characterized by an obnoxious howling
or screeching sound, can be a problem with any sound
system using microphones. Most sound systems are
operated below the point where feedback occurs. The
margin for stable (feedback-free) operation reduces every
time another microphone is opened. Each doubling of the
number of open microphones results in 3 dB less gain-
before-feedback. Open one open microphone too many,
and feedback occurs. By keeping the overall system gain
constant no matter how many microphones are open, the Scott Air Force Base
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 Selection and Operation of

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changes over several seconds, based on variations in the

Summary of automatic mixer benefits microphone input signal. Sound that is constant in
frequency and amplitude, like a ventilation fan, will not
• The primary function of an automatic mixer is activate an input but will add to the noise-adaptive threshold.
to keep unused microphones turned off and to Sound that is rapidly changing in frequency and amplitude,
instantaneously activate microphones when needed. like speech, will activate an input. The mixer activates an
• Using an automatic mixer will: input based on two criteria:
- Improve gain before feedback 1. The instantaneous input signal level from
- Reduce audio degradation caused by the talker is greater than the channel’s
superfluous open microphones noise-adaptive threshold.
- Control the build-up of background noise 2. The input channel has the highest signal
• Keeping the number of open microphones to a level for that talker.
minimum always improves overall audio and quality This second criterion ensures that a very loud talker
• The additional control circuitry on automatic only activates one channel at a time.
mixers provide a variety of additional functions like: Mixers of this type usually require a "last mic on"
- Audio privacy switches feature that keeps at least one microphone activated at all
- Chairperson control of all microphones times to maintain a consistent level of background noise.
- Illuminated indicators of microphone status A NOMA circuit is essential to keep the overall mixer
- Automatic video camera selection based on output below the feedback threshold. An automatic mixer
microphone activation without NOMA is really nothing more than a multi-channel
noise gate.
It should be noted that automatic mixers do not "mix"
Gated Automatic Mixers in the traditional sense. The gain adjustments made to
The most basic form of automatic mixer functions as individual channels are not continuously variable, but
essentially a multi-channel noise gate. When the input simply a transition from an "on" to an "off" state. Any
signal surpasses a fixed threshold of some level, the balancing of signal levels must be accomplished by either
channel activates. The input is attenuated when the level a human operator or, to a limited extent, a dynamics
drops below the threshold. These mixers tend to either clip processor such as an AGC. Consequently, automatic
desired speech if the threshold is set too high, or trigger on microphone mixers are not recommended for musical
undesired sounds if the threshold is set too low. Some applications. Mixing for music is as much an art as a
designs only allow one talker at a time to prevent multiple science, and artistic decisions are best left to a human
microphones from gating on for a single source. being. Additionally, automatic mixers that use noise-
A variable-threshold automatic mixer attempts to rectify adaptive threshold technology may be confused by
these problems by using the signal from a remote musical program material, causing erratic gating.
microphone to provide a reference signal for setting the
threshold. The desired talker must exceed this level by
some preset amount to activate the channel. The remote
microphone must be located such that it will not detect the
program material, but only variations of room noise and re-
verberation. These levels, however, may not be identical to
those at the talker’s location. If the background noise levels
are louder than those at the talker, the talker may not be
speaking loudly enough to activate the channel. Some
models use the sum of the outputs all the microphones to
derive a threshold, rather than a remote microphone. This
approach can work very well, because background noise
is measured at the talker’s location, and the talker will have
a natural tendency to talk above the ambient level.
The noise-adaptive threshold automatic mixer employs
a dynamic threshold unique to each input channel. Each
input channel sets its own minimal threshold that continually Figure 1-19: multi-mic comb filtering
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 Selection and Operation of

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Most automatic mixers share many of the same interruptions and overlaps in speech. Again, mixers of
controls as manual mixers, including individual level this type are not appropriate for music applications,
adjustments, phantom power, basic equalization, etc. where microphone signal levels should be balanced
Several functions unique to automatic mixers are equally. Finally, microphones in this system are never
detailed below: turned "off", negating the need for last microphone hold
Input Channel Threshold: Determines the signal level or one-mic-per-talker circuits.
where the mixer will pass the incoming microphone signal
to the mixer’s output. Feedback Reducers
Last Microphone Lock-On: Keeps the most recently As discussed previously, equalizers can be powerful
activated microphone on until another channel is activated, tools for minimizing feedback problems in a sound system.
maintaining room ambience when the automatic mixer The proper use of an equalizer for feedback control,
is used to provide a feed for broadcast, recording, or to an however, requires a skilled operator with either a
assistive listening system. well-trained ear for identifying feedback frequencies or
Hold Time: Specifies the amount of time a channel stays analysis tools to identify the problems. A feedback reducer
activated after speech has ceased. The feature prevents (eliminator, suppressor, destroyer) accomplishes the
the channel from gating off during the natural gaps that same function automatically. These devices are basically
occur in speech patterns. adaptive equalizers. The equalizer employs a digital
Off-Attenuation: Determines how much gain reduction is algorithm that can identify the characteristic build-up of a
applied to an input channel when the channel is not active. particular frequency that is feeding back, and places an
The range of adjustment can vary from 3 dB to 70 dB, but extremely narrow filter at that frequency. The bandwidth
15 dB is a common value. Some mixers allow for a setting of a feedback reducer filter typically ranges from 1/10 to
of infinity, or a true "off" setting. 1/70 of an octave. The depth of the filter is usually dependent
Decay Time: Establishes the time required for an input on the level of the offending frequency. Most feedback
to be lowered from the activated state to the attenuated reducers will only cut the frequency as much as necessary.
state. As in a dynamics processor, decay time is always in It is usually desirable that the filter width narrow as the
addition to hold time. depth increases, to prevent unwanted attenuation of
adjacent frequencies. An effective feedback reducer
Gain Sharing Automatic Mixers should react quickly, with negligible effect on the overall
A gain-sharing automatic microphone mixer works sound quality of the audio system. The net effect of the
from the premise that the sum of all the signal inputs from feedback reducer should be to flatten the overall system
all microphones in the system must be below some response by using adaptive filters to reduce peaks.
maximum gain value that avoids feedback. The mixer (See Figure 1-20.)
maintains this level by distributing a constant total gain
among the inputs, based on their relative levels. If nobody
is speaking, the total available gain in the mixer is distrib-
uted equally to each input. When one person speaks, that
channel has more signal than the others. Consequently,
the mixer allocates more gain to that channel, and less
gain to the others, roughly in proportion to the relative
increase in signal level. The total gain of the system is the
same as when no one is speaking.
For example, a 3 dB level increase at one microphone
causes that channel gain to rise by 3 dB, while the gain of
the other channels decreases by a total of 3 dB. When two
talkers speak into separate microphones with levels that
differ by 3 dB, they appear at the output of the system with
a 6 dB difference. The microphone with the highest signal
is given the most gain, while the microphone with the low-
est signal is given the least. Since a gain-sharing automatic
mixer increases the level difference between microphones,
the key to transparent operation is fast action to prevent DFRs in rack
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 Selection and Operation of

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Assume that two frequencies begin to ring, one at 500 Hz

and one 5000 Hz. A 500 Hz wave completes a full cycle
in 1/500th of a second, or 2ms. The 5000 Hz wave
will complete a cycle in 1/5000th of a second, or .2 ms.
A feedback reducer should be able to identify the 5000 Hz
feedback tone 10 times faster than the 500 Hz tone.
More importantly, feedback reducers are subject to
the same limitations as manual equalizers. Foremost
among these limitations, a feedback reducer cannot cause
the sound system to achieve more gain-before-feedback
than the levels dictated by the PAG equation (Appendix 2).
Remember that adaptive equalization attempts to flatten
frequency response anomalies in the sound system.
frequency response peak
Once this has occurred, no further benefit is possible from
equalization. A feedback reducer can provide a maximum
of 6 to 10 dB of additional gain. A feedback reducer is not
a substitute for poor system design. Proper choice and
placement of loudspeakers and microphones must be
the first priority.

feedback reduction filter Applications Tip:

Not Enough Feedback Filters?

Feedback reducers cannot deploy an unlimited

number of filters. The number of possible filters
is limited by the DSP capabilities of the device.
Increasing DSP power makes it possible to deploy
more filters, but if more than 10 filters are required,
other problems with the sound system may need to
be addressed. Instead of getting a feedback reducer
that has more filters, investigate other alternatives to
reducing feedback. (See Section 2.1.)

Acoustic Echo Cancellers

Figure 1-20: corrected frequency response Echo cancellers reduce residual echo return in audio
conferencing (teleconferencing) installations. Possible
Audible feedback must occur before the reducer can sources of echo in a teleconferencing system include:
perform its task, hence, these devices are not "pre- improperly balanced hybrids, signal coupling within the
emptive." A feedback reducer does not anticipate telephone lines, and satellite transmission links with long
feedback, but reacts accordingly once feedback is propagation delays. These types of echoes are electronic in
detected. The faster the frequency detector algorithm nature and can be reduced by a line echo canceller.
works in a feedback reducer, the less chance that the Acoustic echo occurs when audio received from the
audience will be annoyed by feedback. The speed of remote site reaches active microphones at the near site,
feedback detection is frequency-dependent, as well. and is transmitted back to the remote site with sound from
For the detector to properly identify the frequency, the both the near site talkers and acoustic echoes of the sound
sound wave must complete several cycles. The longer that originated at the remote site. This type of echo requires
wavelengths associated with lower frequencies take more an acoustic echo canceller (AEC).
time to complete a cycle. Therefore, lower frequency An AEC monitors the incoming audio signal from
feedback takes longer for the detector to properly identify. remote sites, and compares it to the signal that is about to
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 Selection and Operation of

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be transmitted. If the echo canceller detects the presence

of the incoming audio in the outgoing signal, it attempts to
remove it electronically from the outgoing signal. This
reduces the amount of echo, but does not completely
"cancel" it. Notice that the echo canceller attempts to
prevent the incoming audio from other sites from being
sent back to them, but it does not do anything about
echoes that other sites may be sending to your site. The
AEC only improves audio for the remote site, not the one
where the unit is installed. (See Figure 1-21.) Therefore, if
one site on a network requires an echo canceller, all of the
sites will probably need one.
More powerful processors and advanced cancellation
algorithms have resulted in acoustic echo cancellers that
are better and less expensive. It is now possible to have a
separate echo canceller for each microphone input
channel, which provides optimum echo reduction.
Acoustic echo cancellers are commonly believed to
be capable of removing the hollow sound associated with Teleconference system without AGC
a room that is too reflective. In fact, excess reflective sound
makes it difficult for the echo canceller to work properly,
and reduces its effectiveness. As with all audio processing, Near Site
a room outfitted with proper acoustic treatment should
come before attempts to fix a problem electronically.

Far Site

Teleconference system with AGC at far site

Figure 1-21

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 Selection and Operation of

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C H A P TE R TW O 1. Move the microphone closer to the talker.

This is the easiest, most effective, and sometimes most
PRACTICAL APPLICATIONS FOR economical way of improving gain-before-feedback. Expect
AUDIO SIGNAL PROCESSORS an increase in gain-before-feedback from 6 to 25 dB.
Moving the microphone from a distance of 8 inches to 4
inches away from the talker provides a 6 dB increase in
Maximizing Gain-before-feedback gain. Switching from lavalier microphones to headset
If a sound reinforcement system cannot produce microphones changes the distance to the talker from
enough sound level at the audience position before it starts approximately 12 inches to less than one inch, which
to feed back, intelligible audio and balanced frequency provides a 24 dB increase in potential acoustic gain.
response are next to impossible. The first and most basic 2. Move the loudspeaker closer to the listener (i.e.
function of a sound system is to provide enough sound level away from the talker) or add a second loudspeaker for the
at the audience position so that they can hear the performers rear part of the room. Expect an increase in gain-before-
at a comfortable level above the room’s ambient noise. feedback from 3 to 15 dB. Installing a second loudspeaker
Feedback occurs when the amplified sound from any set or satellite loudspeakers to provide sound coverage for
loudspeaker reenters the sound system through any open the rear of the room allows the front of house (FOH)
microphone and is amplified again and again and again loudspeakers to be turned down, as they no longer have to
This is a physical phenomenon and audio processors project sound all the way to the back of the room. The
cannot help a sound system obtain gain-before-feedback second set of loudspeakers in the rear of the room effectively
beyond the limits of physics. Most sound systems do not brings the loudspeaker much closer to the listener providing
operate near their physical limit, yet they still experience more gain-before-feedback. Every time the distance
feedback problems. Potential Acoustic Gain is the amount between the loudspeaker and the listener is cut in half, the
of level (in dB) that a sound system can produce before the potential acoustic gain of the system increases by 6 dB.
onset of feedback. 3. Reduce the number of open microphones. Expect
The following list highlights the only possible solutions an increase in gain-before-feedback from 3 to 12 dB. Every
to feedback problems and what kind of improvement to time the number of open microphones in a sound system
expect. Note that some audio processors, like automatic doubles, the potential acoustic gain of the system is
mixers and feedback reducers, can help a system achieve reduced by 3 dB. If 2 new microphones are turned on in
the maximum amount of gain before feedback by a system that previously had 2 open microphones, the
optimizing some variables in the equation. Realize that system will have to be turned down by 3 dB to maintain the
the most helpful, yet least expensive options do not even same feedback stability margin. Adding more microphones
involve an audio processor. can be a solution to feedback problems only if the
See Appendix 2 for a complete mathematical microphones are being placed much closer to the talker
discussion of the Potential Acoustic Gain equation. than they were previously. For example, a few overhead,
hanging microphones can be replaced with many lavalier
microphones. In general, double the number of
microphones only if the distance from the microphone to
the talker is reduced by at least half. This should result in
a minimum of a 3 dB increase in PAG with better coverage
Listener D1 Talker of the desired pick up area. Automatic microphone
D2 (source)
mixers greatly help sound systems with 4 or more open
Microphone microphones by keeping microphones that are not being
addressed turned down. This effectively reduces the
number of open microphones and increases the potential
acoustic gain of the system.
Ds
4. Use unidirectional microphones and loudspeakers.
D0 Expect an increase in gain-before-feedback from 3 to 8
dB. The proper use of unidirectional microphones, such as
those with cardioid or supercardioid polar patterns, can
help pick up more of the desired sources (the talkers) and
Figure 2-1: Minimize DS and D2 less of the undesired sources (in this case, loudspeakers).
23
 Selection and Operation of

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Processors

They also help reject other undesired sources such as

ambient noise and the room reverberation.
Loudspeakers with high directionality or narrow
dispersion patterns are also available and can improve
gain-before-feedback as well as intelligibility. They
accomplish this by directing most of the sound energy
to the audience. In doing so, less energy is sent to the
reverberant field or back to the stage to cause feedback
problems. The latter approach usually requires a
complete redesign of the sound system.
5. Move the loudspeaker further from the
microphones. Expect an increase in gain-before-feedback Figure 2-2: overall system response with peaks
from 2 to 9 dB. Doubling the distance between the
microphone and the closest loudspeaker results in a 6 dB room has been built with inappropriate geometry, it is very
increase in gain-before-feedback. However, moving the loud- difficult to fix acoustical problems. Covering the walls, floor,
speaker to twice the distance from the microphones is a less and ceiling with sound absorbent materials is at best a fair
realistic option than the ones previously discussed. This solution. While more expensive than all the other options
approach usually results in inappropriate coverage for the discussed above, it can help reduce problems like long
audience in the front of the room, and there may be a space reverberation or decay times that affect intelligibility, and
limitation that does not permit moving the loudspeakers. standing waves and reflections that affect gain-before-
If stage monitors are being employed, they should be feedback and system frequency response. Keep in mind
used only for the monitoring of music, effects, cues, and that, as a rule of thumb, to make a noticeable change you
playback. The signal from lavalier, boundary or overhead must treat 50% of the room’s surfaces with sound
microphones on stage intended to pick up the performer’s absorbent materials. In some cases, a single wall or
voice should never be routed to these monitors as it will surface, such as the back wall, could be causing most of
severely handicap the amount of level the system can the feedback problems. Treating this surface alone could
provide for the audience. If performers on stage need to produce a good, noticeable improvement in gain-before-
hear each other or themselves, they must wear in ear feedback, even though it will not dramatically improve
monitors or consider using handheld, headset, or other intelligibility or reduce reverberation time.
microphone designs where the distance between the There are no other solutions! These guidelines
talker’s mouth and the microphone capsule is extremely provide you with the ONLY options available to increase
small (less than 1 inch.) the potential acoustic gain of a sound system.
6. Reduce gain at feedback frequencies using notch
filters (narrow equalizer filters). Expect an increase in Improving Speech Intelligibility
gain-before-feedback from 3 to 9 dB. Narrow peaks in the Speech intelligibility is among the most difficult goals
overall frequency response of the sound system are the to achieve for any medium-to-large indoor sound
first to cause feedback. These peaks rise 2 to 10 dB above reinforcement system. Some of the factors that play major
the overall system response and prevent the system from roles in obtaining good intelligibility are beyond the control
reaching its maximum potential acoustic gain. (See Figure of the sound system or signal processing. These factors
2-2.) This technique can be done with a manual equalizer include the acoustic characteristics of the space (in
(and the appropriate measurement tools) or a feedback particular its geometrical shape), its background noise
reducer. A digital feedback reducer can detect feedback level, and its reverberation time.
and insert a notch filter at the exact offending frequency, There are two basic ways that audio processing can
which effectively flattens the sound system’s frequency improve the speech intelligibility of a sound system. The
response and allows it to reach its maximum possible first, and most effective, is by reducing the number of open
gain-before-feedback. microphones. This approach involves using an automatic
7. Improve room acoustics with acoustic treatment. mixer to keep the microphones of participants who are not
Expect an increase in gain-before-feedback from 3 to 6 talking turned down. The second method employs an
dB. An acoustical consultant, working in conjunction with equalizer to limit the frequency response of speech
the architect, should design a venue with good acoustics microphones to the speech frequency range only. A
before construction of the building even begins. Once the bandpass equalizer is typically the appropriate tool for this job.
24
 Selection and Operation of

AUDIO SIGNAL
Processors

Reducing the Number of Open Microphones: the preferred way of employing automatic microphone mixers
Automatic microphone mixers are typically the easiest is in the form of speech gates. In this scenario, the automatic
audio processor to implement, since most designs require mixer is connected to the mixing console on a per-channel
very little setup on the part of the user. In the majority of basis via the insert jacks for each input channel. The operator
applications, microphones are attached directly to has full control of each microphone’s level when it is in use
the mixer. Common applications include boardrooms, and retains all the functionality of the mixing console. The
courtrooms, houses of worship, and theater. automatic mixer keeps only the microphones of performers
Boardrooms/Meeting Rooms/Council Chambers: that are talking turned up.
Any meeting facility that uses more than three
microphones should consider an automatic microphone Equalizing for Speech Intelligibility:
mixer. Even if the talkers are using push-to-talk Using equalization in sound reinforcement takes on
microphones to keep the number of open microphones to two forms: the objective and the subjective. Objective
a minimum, they often forget to activate the microphone, equalization entails the use of corrective equalization to
leading to missed speech. Or, in the case of push-to-mute compensate for frequency response anomalies in the sound
microphones, the delegate forgets to turn the microphone system components and room resonances that cannot (for
off. Momentary mute (or cough) switches are usually financial or logistical reasons) be cured by acoustical
desirable, since the automatic mixer cannot distinguish means. Proper objective equalization requires the use of
between a cough and speech. A mixer with microphone measurement devices to obtain a theoretically flat frequency
logic capabilities can provide additional functionality for response. Flat frequency response, while desirable as a good
chairman microphone override, remote LED indication, starting point, may not produce the most audibly pleasing
and automatic camera-switching. result. Here is where subjective EQ enters the picture.
Houses of Worship: As above, use an automatic mixer Subjective equalization is more art than science, and
if there are more than three microphones. Additionally, for requires a skilled operator with a trained ear to obtain optimal
worship leaders who use a lavalier microphone as well as results. "Sounds good" cannot necessarily be quantified in
a gooseneck microphone at the lectern, the automatic measurable terms. However, some general guidelines can
mixer will only activate one of the microphones, preventing help with regard to enhancing intelligibility.
comb filtering. The same applies to lecterns with two Reproducing intelligible speech demands a minimal
microphones. While logic dictates that two microphones frequency response from a sound system equal to that of
provide better coverage for roaming talkers, the trade-off in a telephone system - about 300 Hz to 3 kHz. A wider
comb filtering often creates more problems than it solves. frequency response can enhance the tonal quality of the
Using an automatic mixer prevents comb filtering while reproduction but can also degrade intelligibility by
providing a wider coverage area. emphasizing pops, rumble, hiss, room acoustics, and other
As mentioned previously, automatic microphone noises that are extraneous to speech and would not be
mixers are not recommended for music sources. Since present in a normal conversation. Wider frequency
most house of worship applications combine music and response also permits more sound energy to unnecessarily
speech, both a manual and an automatic mixer should be contribute to the reverberant field of the room. This makes
used. The simplest setup could use the automatic mixer to the system more prone to feedback and less intelligible.
submix the speech microphones into one channel of the Equalization can noticeably, but not dramatically,
manual mixer. Alternately, if using a sound system improve the naturalness or intelligibility of a sound
processor that has a matrix mixer, the outputs of the reinforcement system by emphasizing the frequency
automatic mixer and manual mixer can be combined and ranges most critical for speech.
routed by the processor. Either way, speech and music Equalization cannot make a poorly designed sound
sources are handled independently. If the application only system work satisfactorily or improve intelligibility problems
has an automatic mixer, use the logic functions to "force" caused by reflections, mechanical vibration, and high
the music microphones on so they will not mute. Note that background noise levels. It cannot improve intelligibility prob-
for mixers with a NOMA circuit, this approach will reduce lems caused by the talker being too far from the microphone,
the output of the mixer, and any additional noise picked improve the performance of substandard audio components,
up by the music microphones will always be present unless or eliminate distortion and noise problems caused by
muted by a human operator or traditional noise gate. mismatched audio levels between system components.
Theater: In theater applications, where the sound system A hi-cut/low-cut (or band pass) equalizer is the most
operator requires complete control over the performer’s audio, basic tool needed to equalize speech microphones for
25
 Selection and Operation of

AUDIO SIGNAL
Processors

optimum intelligibility. Perception research and studies of Sound System Gain Structure
human hearing suggest the following EQ curve as a good Setting gain structure in a sound system concerns the
starting point. It maintains good, natural voice tonality while proper calibration of signal levels between devices in the
attenuating all unnecessary frequencies. audio chain to achieve good signal-to-noise ratio and
• Low Cut Filter (LC) set to 125 Hz, adequate headroom. Poor signal-to-noise ratio results
6 dB per octave. in a high level of background noise (hiss) that, at best,
• High Cut Filter (HC) set to 4000 Hz, is annoying for the listener, and, at worst, obscures
6 dB per octave. (See Figure 2-3.) intelligibility. Objectionable background noise usually results
Increasing the response bandwidth, for example from in a system with excessive headroom, where the desired
80 Hz to 8000 Hz, would provide a slight improvement in audio signal level is close to the noise floor. In contrast, low
tonal quality. Decreasing the bandwidth slightly from the headroom, where system noise is quiet but the audio signal
low end should improve intelligibility. The minimum is close to clipping, can lead to overload conditions that
response should never be narrower than 400 Hz to 2.5 could cause distortion or loudspeaker failure. If every piece
kHz and the filter slopes should not exceed 12 dB per of audio equipment clipped (started to audibly distort) at the
octave. Note that the human voice contains very little same level and had a similar dynamic range, then audio
energy below 100 Hz. While adding response below this systems would be "plug-and-play." Unfortunately, this is not
point may sound impressive, the effect on intelligibility is the case. (See Figure 2-4.)
more detrimental than helpful. Novice sound technicians commonly mistake the input
In addition to bandpass filters, a parametric equalizer sensitivity control on a power amplifier for a "volume" knob,
can be used to boost a selective frequency range. Using often rotating the control to maximum in an attempt to get the
a parametric filter to help intelligibility is mostly an highest possible level out of the sound system. Unfortunately,
experimental exercise and the exact frequency, band- the end result is usually additional noise. The input sensitivity
width, and boost will vary from system to system. The knob should be set just high enough to ensure maximum
idea is to boost a set of frequencies that are most output from the amplifier. This point is determined by the
essential to speech to overcome interference from the setting at which the amplifier input sensitivity indicators begin
acoustical environment. This frequency is typically to show clipping. Any additional boost beyond this point only
between 1 and 4kHz. The typical boost is 3 to 5 dB. The adds noise. Maintaining the highest possible signal levels
width of the filter can vary from 1/6 octave to 1 octave. throughout the various components of the sound system in
In general, approach equalization slowly. After every the easiest way to realize maximum output with minimal
adjustment, listen carefully to the resulting sound. Most noise. If the power amplifier controls are indiscriminately
changes are not perceived as good sounding immediately. placed at maximum, the sound technician must operate the
Listen for at least 3 minutes to each change to allow your mixer and other audio components in the signal chain at
ear to adapt. If the equalizer has a bypass button, use it lower levels. Consequently, the program material is close in
often to provide a reference point. When the system is clear level to the noise floor of the mixer. Using the amplifier’s input
enough, stop equalizing. sensitivity control to compensate for low levels from the
When listening to live microphones, have someone mixer only exacerbates the noise problem by raising the noise
else talk, never try to equalize to your own voice. When floor of the mixer as well as the program material. If sound
using recorded material to equalize, choose a recording levels in the room are too loud, the input sensitivity of
that you are familiar with and have listened to many times the amplifier, rather than the level control on the mixer, should
in different sound systems. be reduced to maintain good signal-to-noise. In any case,
amplifiers should be turned down,
or off, until good gain structure is
achieved in all components prior to
the amplifiers.
This section introduces two
methods of setting system gain
structure, the unity method and the
optimized method. Both methods
rely on strong signal levels through-
out the sound system, but differ in
Figure 2-3: speech EQ curve approach.
26
 Selection and Operation of

AUDIO SIGNAL
Processors

Clip +24 dBu +20 dBu +18 dBu

Headroom 20 dB 16 dB 14 dB
Output level +4 dBu +4 dBu +4 dBu

LOUDSPEAKER

Figure 2-5: gain structure, unity method

The Unity Method system are consistent, headroom is not. The likelihood
The historically conventional way to set sound system of clipping components post-mixer is the single biggest
gain structure, this method relies on unity amplification, drawback. Consider a mixer with an output clipping level
that is, every component after the mixer should produce an
output voltage equal to the voltage at its input. If we assume
Maximum Output Level
typical line level, +4 dBu, each device in the system should +20 dBV
be calibrated to produce this level at its output, ultimately
resulting in +4 dBu at the amplifier input. The amplifier’s
input sensitivity control is used to set the desired sound +4 dBV
level in the room. Advantages to this approach include:
1. Easy calibration
2. Easy to substitute components
3. Fast implementation -80 dBV
However, there are several significant disadvantages to Noise Floor
the unity method. While operating levels throughout the too close to noise floor (excess headroom)

Peaks Maximum Output Level

+20 dBV

+4 dBV

-80 dBV
Quiet Passage
Noise Floor

typical audio signal too close to ceiling (distortion)

each device has... Maximum Output Level

+20 dBV
Maximum Output Level (dBm or dBV)

+4 dBV

-80 dBV
Noise Floor
Noise Floor (dBm or dBV)
just right

Figure 2-4
27
 Selection and Operation of

AUDIO SIGNAL
Processors

Clip +24 dBu +20 dBu +18 dBu

Headroom 20 dB 20 dB 20 dB
Output level +4 dBu 0 dBu -2 dBu

LOUDSPEAKER

Figure 2-6: gain structure, optimized method

of +24 dBu. (See Figure 2-5.) Assuming that mixing at A pad may be required before the input of the power
meter "0" produces +4 dBu output level, the mixer has 20 amplifier if clipping occurs at a low gain setting.
dB of headroom. If the output of the mixer is connected to Otherwise, raise the input level control of the power
an equalizer with a clipping level of +20 dBu, the equalizer amplifier until either the desired sound level is achieved
only has 16 dB of headroom. Therefore, a waveform that for the audience, or the amplifier begins to indicate
contains transients well within the headroom of the mixer clipping. Realize that if clipping does occur before the
could potentially cause distortion at the equalizer. Mixing desired sound level is achieved, a larger power amplifier
below meter "0" results in lower output voltage, which (and consequently, loudspeakers that can handle the
could help maintain 20 dB of headroom, but most likely power) may be required.
will prove confusing for system operators unfamiliar with
this sound system. Optimally, all components in a system Digital Signal Processing
should clip at the same point. A digital signal processor (DSP) uses complex digital
software algorithms to emulate the operation of analog signal
The Optimized Method processors in digital hardware. A DSP is nothing less than a
Establishing gain structure using the optimized specialized audio computer with its own operating system
method results in inconsistent operating level, but and software. Some models can be configured with front
consistent headroom. With this approach, each device can panel controls, but others need to be connected to a PC for
output its maximum voltage, yet not overdrive the next
component. This technique typically requires a resistive
pad between components. Using the above example, the
equalizer’s clipping level is 4 dBu lower than the mixer.
Therefore, the output signal from the mixer needs to be
reduced by 4 dB before the input of the equalizer. (See
Figure 2-6.) Occasionally, the attenuation can be achieved
by lowering the input sensitivity control of the device. If not,
a 4 dB attenuator should be placed between the mixer and
the equalizer. The output signal from the mixer will be
lowered to 0 dBu at the input of the equalizer, maintaining
20 dB of headroom. Advantages to the optimized
method include:
1. Optimized signal-to-noise ratio
throughout the system.
2. All components clip simultaneously.
Mixing at meter zero results in the same
headroom throughout the system.
Of course, this method requires more time and
expertise on the part of the installer, and component
substitution is more difficult since a replacement device Figure 2-7:
may have a different clipping level. real time monitoring of Automatic Gain Control (AGC) functions

28
 Selection and Operation of

AUDIO SIGNAL
Processors

setup. The latter requires a program called a Graphical User Work anywhere: The software for most processors
Interface (GUI) to control the DSP. (See Figure 2-7.) does not require that the user by connected to the
While single-function DSP devices are available, the processor itself for design purposes. This functionality
real advantage lies with multi-function devices. The allows the installer to design the system anywhere there is
majority of these products provide every type of processing access to a computer with the software, anytime it is
required between the outputs of the mixer and the inputs convenient, and then load the design into the processor
of the power amplifiers and, in some cases, they can later. While certain parameters require on-site adjustment
eliminate the need for a stand-alone mixer. Depending on (such as equalization), signal flow, at the very least, can be
the feature set, these devices can be classified as either planned in advance.
loudspeaker processors or sound system processors. Control Options: Many digital signal processors offer
A loudspeaker processor tends to emphasize tools for control options for remote adjustment of certain processor
protecting and aligning loudspeakers, such as crossovers, parameters. These features are particularly useful for
limiters, and delay. A sound system processor adds more situations where the end-user needs some sound system
front-end functionality, such as feedback reduction, echo control, but leaving behind a PC with the software could
cancellation, and more advanced matrix-mixing capability. prove disastrous. Typical control options include preset
Some processors even provide microphone inputs and au- selection, remote volume control, and remote muting of
tomatic mixing. A key benefit of many DSPs is the ability to inputs or outputs.
lock settings with password protection for installations in Low noise and easy system connectivity: Gain
which a tamper-proof sound system is desired. Without a structure is greatly simplified due to fewer physical
PC, the appropriate software, and the system password, components in the signal chain. Signal levels between
access to parameters that could jeopardize the functionality functions within the processor do not need to be calibrated.
of the sound system is eliminated. Other significant Additionally, the noise floor of a single processor is
advantages to digital signal processors include: significantly lower than that created by multiple devices.
Flexibility: While certain guidelines often dictate Cost: A single multi-function digital signal processor
the order of components in the signal path, different typically costs far less than the equivalent amount
situations may require a more flexible architecture. Some of processing in several stand-alone devices. Also, if
processors only provide a fixed signal path (for example, additional processing is required after the design phase, it
input-EQ-compressor-crossover-output). At the other is just a matter of reprogramming the software rather than
extreme, some processors use a completely open re-laying out the equipment rack and purchasing another
architecture, where the designer is essentially given a blank hardware device.
page to design the sound system using a GUI that works Time: It takes much less time to install a single DSP
like CAD (Computer Aided Design) drafting software. Lastly, device compared to the time required to install, wire, and
a hybrid of the two methods offers a fixed number of place connect multiple processing components. The ease with
holders for processor modules, but gives the designer the which these processors can be programmed and imple-
ability to place the desired processing in any available mented saves cost in installation and design time, as well.
place-holder, and route the signal as required. The power and flexibility provided by digital signal
Ease of programming: Using a computer for system processors gives sound system operators and installers
setup should be intuitive and easy to learn. Hardware- all the necessary tools to provide an optimal auditory
based interfaces are typically more difficult to learn due to experience for the intended audience. As listener
limited display area and multi-purpose controls. Adjusting expectations continually get more and more sophisticated,
a single parameter often requires searching through a complete set of tools is required to meet those
multiple layers of menus. Most digital processors that are expectations: equalizers for tone shaping and feedback
programmed by computer present the user with GUI control, dynamics processors for increased audibility, and
software that can make programming as simple as drawing adaptive audio processors to automate control when
lines or entering parameter values directly into the proper possible. The combination of skilled design and proper
fields. The entire system layout and signal flow can be application of the various audio processors results in
displayed on a single screen. superior sound quality for any venue.

29
 Selection and Operation of Reference Information

AUDIO SIGNAL A p p e n d i x A : Sound

Processors

Appendix A
Sound is produced by vibrating objects. These include
musical instruments, loudspeakers, and, of course,
human vocal cords. The mechanical vibrations of these
objects move the air which is immediately adjacent to
them, alternately “pushing” and “pulling” the air from its
resting state. Each back-and-forth vibration produces a
corresponding pressure increase (compression) and
pressure decrease (rarefaction) in the air. A complete
pressure change, or cycle, occurs when the air pressure
goes from rest, to maximum, to minimum, and back to rest
again. These cyclic pressure changes travel outward from
the vibrating object, forming a pattern called a sound wave.
A sound wave is a series of pressure changes (cycles)
moving through the air.
A simple sound wave can be described by its Instrument Frequency Ranges
frequency and by its amplitude. The frequency of a sound
wave is the rate at Another characteristic of a sound wave related to
1 CYCLE
▲
▲

/ CYCLE
1
which the pressure frequency is wavelength. The wavelength of a sound wave
▲
▲

changes occur. It is the physical distance from the start of one cycle to the
is measured in start of the next cycle, as the wave moves through the air.
+ ▲ Hertz (Hz), where Since each cycle is the same, the distance from any point
PRESSURE

0 AMPLITUDE 1 Hz is equal to 1 in one cycle to the same point in the next cycle is also one
_ ▲
cycle-per-second. wavelength: for example, the distance from one maximum
DISTANCE WAVELENGTH
▲
▲

The range of pressure point to the next maximum pressure point.

Schematic of Sound Wave frequencies audi- Wavelength is related to frequency by the speed of
ble to the human sound. The speed of sound is the velocity at which a
ear extends from a low of about 20 Hz to a high of about sound wave travels. The speed of sound is constant and
20,000 Hz. In practice, a sound source such as a voice is equal to about 1130 feet-per-second in air. It does not
usually produces many frequencies simultaneously. In any change with frequency or wavelength, but it is related to
such complex sound, the lowest frequency is called the them in the following way: the frequency of a sound,
fundamental and is responsible for the pitch of the sound. multiplied by its wavelength always equals the speed
The higher frequencies are called harmonics and are re- of sound. Thus, the higher the frequency of sound, the
sponsible for the timbre or tone of the sound. Harmonics shorter the wavelength, and the lower the frequency,
allow us to distinguish one source from another, such as a the longer the wavelength. The wavelength of sound
piano from a guitar, even when they are playing the same is responsible for many
fundamental note. In the following chart, the solid section acoustic effects.
of each line indicates the range of fundamental frequen- After it is produced,
cies, and the shaded section represents the range of the sound is transmitted
highest harmonics or overtones of the instrument. through a “medium”. Air is
The amplitude of a sound wave refers to the magnitude the typical medium, but
(strength) of the pressure changes and determines the sound can also be
“loudness” of the sound. Amplitude is measured in decibels transmitted through solid or
(dB) of sound pressure level (SPL) and ranges from 0 dB liquid materials. Generally,
SPL (the threshold of hearing), to above 120 dB SPL (the a sound wave will move in a
threshold of pain). The level of conversational speech is straight line unless it is
about 70 dB SPL. A change of 1 dB is about the smallest absorbed or reflected by
SPL difference that the human ear can detect, while 3 dB physical surfaces or objects
is a generally noticeable step, and an increase of 10 dB is in its path. However, the Sound Pressure Level
perceived as a “doubling” of loudness. transmission of the sound of Typical Sources
30
Reference Information Selection and Operation of

A p p e n d i x A : Sound AUDIO SIGNAL

Processors

wave will be affected only if the size of the surface or

object is large compared to the wavelength of the sound.
Indirect
Indirect
If the surface is small (compared to the wavelength) the Sound Path
Path
Sound
sound will proceed as if the object were not there. High
frequencies (short wavelengths) can be reflected or
absorbed by small surfaces, but low frequencies (long
wavelengths) can be reflected or absorbed only by very Direct
Direct
Sound Bar
Bar
large surfaces or objects. For this reason it is easier to Sound
Sound
control high frequencies by acoustic means, while low Source
Source
frequency control requires massive (and expensive)
techniques. Direct vs. Indirect Sound
Once a sound has been produced and transmitted, it
is received by the ear and, of course, by microphones. In corresponding to that wavelength will be amplified, by
the ear, the arriving pressure changes “push” and “pull” addition of the incoming and outgoing waves, resulting in
on the eardrum. The resulting motion of the eardrum is a strong, stationary wave pattern between the two
converted (by the inner ear) to nerve signals that are surfaces. This happens primarily with low frequencies,
ultimately perceived as “sound”. In a microphone, the which have long wavelengths and are not easily absorbed.
pressure changes act on a diaphragm. The resulting A very important property of direct sound is that it
diaphragm motion is converted (by one of several becomes weaker as it travels away from the sound
mechanisms) into electrical signals which are sent to the source, at a rate governed by the inverse-square law. For
sound system. For both “receivers”, the sound picked up example, when the distance increases by a factor of two
is a combination of all pressure changes occurring just at (doubles), the sound level decreases by a factor of four
the surface of the eardrum or diaphragm. (the square of two). This results in a drop of 6 dB in
Sound can be classified by its acoustic behavior; for sound pressure level (SPL), a substantial decrease.
example, direct sound vs. indirect sound. Direct sound Likewise, when the distance to the direct sound source is
travels from the sound source to the listener in a straight divided by two (cut in half), the sound level increases by
line (the shortest path). Indirect sound is reflected by one 6 dB. In contrast, ambient sound, such as reverberation,
or more surfaces before reaching the listener (a longer has a relatively constant level. Therefore, at a given
path). Since sound travels at a constant speed, it takes a distance from a sound source, a listener (or a
longer time for the indirect sound to arrive, and it is said to microphone) will pick up a certain proportion of direct
be “delayed” relative to the direct sound. There are several sound vs. ambient sound. As the distance increases, the
kinds of indirect sound, depending on the “acoustic direct sound level decreases while the ambient sound
space” (room acoustics). level stays the same. A properly designed sound system
Echo occurs when an indirect sound is delayed long should increase the amount of direct sound reaching
enough (by a distant reflecting surface) to be heard by the the listener without increasing the ambient sound
listener as a distinct repetition of the direct sound. If significantly.
indirect sound is reflected many times from different
surfaces it becomes “diffuse” or non-directional. This is
called reverberation, and it is responsible for our auditory 8M
perception of the size of a room. Reverberant sound is a
major component of ambient sound, which may include 4M

other non-directional sounds, such as wind noise or 2M

building vibrations. A certain amount of reverberant
sound is desirable to add a sense of “space” to the sound, 1M
but an excess tends to make the sound muddy and 1
/2 M
unintelligible.
One additional form of indirect sound is known as a
standing wave. This may occur when the wavelength of a
sound is the same distance as some major dimension of 52 58 64 70 76db
a room, such as the distance between two opposite walls.
If both surfaces are acoustically reflective, the frequency Inverse Square Law
31
 Selection and Operation of Reference Information

AUDIO SIGNAL Appendix B:

Processors Potential Acoustic Gain and
Needed Acoustic Gain

Appendix B
Listener
D1 Talker
As previously discussed, there is a physical limitation D2 (source)
to how much level a sound reinforcement system can
achieve before uncontrollable feedback occurs. The Microphone

available level, known as Potential Acoustic Gain (PAG),

can be determined by a relatively simple equation. Before
calculating acoustic gain, though, it is helpful to know how Ds
much gain is required to provide an adequate listening D0
level for all members of the audience. The Needed
Acoustic Gain (NAG) equation calculates the amplification
necessary for the furthest listener to hear as well as nearest Potential Acoustic Gain
listener. This equation assumes the nearest listener is close
enough to hear the sound source directly (without Where: PAG = Potential Acoustic Gain (in dB)
amplification).
DS= distance from sound source to microphone
NAG = 20 x log (Df/Dn)
D0 = distance from sound source
Where: Df = distance from sound to furthest listener
source to furthest listener
Dn = distance from sound source D1 = distance from microphone
to nearest listener to nearest loudspeaker
Log = logarithm to base 10
D2 = distance from loudspeaker
The NAG equation is based on the inverse-square law, to furthest listener
which states that sound level decreases by 6 dB for each
doubling of distance from the sound source. For example, NOM = number of open microphones
the front row of an audience (10 feet from the stage) may
experience a comfortable level (without a sound system) of -6 = a 6 dB feedback stability margin
85 dB. The last row, which is 80 feet from the stage, will
only experience 67 dB; 18 dB less than the front row. log = logarithm to base 10
Therefore, the sound system needs to provide 18 dB of
gain to the last row of the audience, so it will experience the The 6 dB feedback stability margin is required to
same listening level as the front row. Using the equation: provide a small amount of "headroom" below the feedback
threshold, even when NAG and PAG are equal. The NOM
NAG = 20 x log (80/10) term reflects the fact that gain-before-feedback reduces
NAG = 20 x log 8 by 3 dB every time the number of open microphones
NAG = 20 x 0.9 doubles. For example, if a system has a PAG of 20 dB with
NAG = 18 one open microphone, adding a second microphone will
cause a 3 dB decrease to 17 dB. Doubling the number of
Potential acoustic gain (PAG) is calculated from the open microphones again, to four, drops PAG to 14 dB.
distances between various components in the sound Consequently, the number of open microphones should
system, the number of open microphones, and other always be kept to a minimum. Unused microphones
variables. The sound system is sufficient if PAG is equal to should be turned off or attenuated, either manually (by a
or greater than the Needed Acoustic Gain (NAG). While it human operator) or electronically (by an automatic mixer).
appears somewhat complex, the equation is easily solved In fact, using an automatic microphone mixer with a
with a scientific calculator: NOMA (Number of Open Microphones Attenuator) circuit
removes the NOM component from the equation, since
PAG = 20 (log D1 – log D2 + log D0 – log Ds) NOMA ensures that the overall output of mixer will always
– 10 log NOM – 6 be equivalent to one open microphone.

32
Reference Information Selection and Operation of

Appendix B: AUDIO SIGNAL

Potential Acoustic Gain and Processors
Needed Acoustic Gain

System will work: PAG>NAG System will not work: PAG<NAG

To provide maximum gain-before-feedback , the following away will decrease gain-before-feedback by 6 dB. Moving
rules should be observed: it to 4 ft. away will cause a 12 dB decrease. Conversely,
1. Place the microphone as close to the sound moving it to 6 inches away increase gain-before-feedback
source as practical. by 6 dB, and moving it to 3 inches away will increase it by
2. Keep the microphone as far away from the 12 dB. The single most significant (and inexpensive) way
loudspeaker as practical. to maximize gain-before-feedback is to place the
3. Place the loudspeaker as close to the microphone as close as possible to the sound source.
audience as practical. The PAG equation allows the performance of a sound
4. Keep the number of open microphones system to be evaluated solely on the basis of the relative
to a minimum. location of sources, microphones, loudspeakers, and
audience, as well as the number of microphones, but
Achieving noticeable results when making changes to without regard to the actual type of component. Note that the
a sound system requires a level difference of at least equation also assumes omnidirectional components. As
6 dB. Due to the logarithmic nature of the PAG equation, discussed previously, using directional microphones and
a 6 dB change requires a doubling or halving of the loudspeakers may increase PAG. Component characteristics
corresponding distances. For example, if a microphone is notwithstanding, the results provided by this relatively simple
placed 1 ft. from a sound source, moving it back to 2 ft. equation still provide a useful, best-case estimate.

33
 Selection and Operation of Reference Information

AUDIO SIGNAL Glossary

Processors

Active – A device that requires power to operate. Decade – The distance between two frequencies that are
multiples or divisions of ten (e.g. 200 Hz – 2000 Hz).
Acoustic Echo Canceller (AEC) – A processor that
attempts to remove acoustic echoes in a Decibel – A number used to express relative output
teleconferencing system. sensitivity. It is a logarithmic ratio.

Ambience – Room acoustics or natural reverberation. Dynamic Range – The range of amplitude of a sound
source. Also, the range of level between the noise floor
Amplitude – Magnitude of strength of signal or wave. and clipping level of a device.

Audio Chain – The series of interconnected audio Echo – Reflection of sound that is delayed long enough
equipment used for recording or reinforcement. (more than about 50 msec.) to be heard as a
distinct repetition of the original sound.
Automatic Gain Control (AGC) – A signal processor
that attempts to compensate for the differences in Equalizer – A signal processor that allows the user
level between different sound sources. to boost or cut selected frequencies. Used for tone
shaping and limited feedback control. Variations
Band Pass Filter – A filter that only allows a certain include graphic or parametric.
range of frequencies to pass.
Expander – A signal processor that expands the
Band Reject Filter – A filter that reduces a range dynamic range of an audio signal.
of frequencies.
Feedback – In a PA system consisting of a microphone,
Bandwidth – The range of frequencies that a filter affects. amplifier, and loudspeaker, feedback is the ringing
or howling sound caused by the amplified sound from
Cardioid Microphone – A unidirectional microphone with the loudspeaker entering the microphone
moderately wide front pickup (131 deg.). Angle of best and being re-amplified.
rejection is 180 deg. from the front of the microphone,
that is, directly at the rear. Fidelity – A subjective term that refers to perceived
sound quality.
Clipping Level – The maximum electrical signal level that
a device can produce or accept before distortion occurs. Filter – A processor that cuts or boosts a specific
frequency or frequency range.
Comb Filtering – The variations in frequency response
caused when a single sound source travels multiple Frequency – The rate of repetition of a cyclic
paths to the listener’s ear, causing a "hollow" sound phenomenon such as a sound source.
quality. The resultant frequency response graph
resembles a comb. Can also occur electronically with Frequency Response – Variations in amplitude of a
multiple microphones picking up the same sound source. signal over a range of frequencies. A frequency
response graph is a plot of electrical output (in decibels)
Compressor – A signal processor that reduces the level of vs. frequency (in Hertz).
incoming audio signals as they exceed a given threshold.
The amount of reduction is usually defined by the user. Gain – Amplification of sound level or voltage.

Crossover – A processor that divides the audio signal into Gain-Before-Feedback – The amount of gain that can
two or more frequency bands. be achieved in a sound system before feedback or
ringing occurs.

34
Reference Information Selection and Operation of

Glossary AUDIO SIGNAL

Processors

Gate (Noise Gate) – A signal processor that mutes the PAG – Potential Acoustic Gain is the calculated gain
audio signal when it drops below a given threshold. that a sound system can achieve at or just below
the point of feedback.
Headroom – The difference between the nominal operat-
ing level of a device and the point at which the device Passive – A device that does not require power to operate.
clips.
Phantom Power – A method of providing power to the
High Pass (Low Cut) Filter – A filter that attenuates low electronics of a condenser microphone through the
frequencies below a certain frequency. microphone cable.

Inverse Square Law – States that direct sound levels in- Q – Quality Factor. Indicates how tightly a filter is
crease (or decrease) by an amount proportional to the focused near the center frequency.
square of the change in distance.
Reverberation – The reflection of sound a sufficient
Limiter – A signal processor that prevents signals levels number of times that it becomes non-directional and
from exceeding a certain threshold. persists for some time after the source has stopped.
The amount of reverberation depends on the relative
Low Pass (High Cut) Filter – A filter that attenuates high amount of sound reflection and absorption in the room.
frequencies above a certain frequency.
Shelving Equalizer – Reduces (or raises) the frequencies
Mixer – A device which allows the combination, below (or above) a certain frequency to a fixed level.
manipulation, and routing of various audio input signals. The response when viewed on a frequency response
graph resembles a shelf.
NAG – Needed Acoustic Gain is the amount of gain that
a sound system must provide for a distant listener to hear Signal to Noise Ratio – A measurement of the noise of
as if he or she was close to the unamplified sound source. device expressed as a ratio between the desired signal
level (dBV) and the noise floor.
Noise – Unwanted electrical or acoustic energy.
Sound Reinforcement – Amplification of live sound
Noise Gate – A signal processor that mutes the audio sources.
when the signal level drops below a certain threshold.
Speed of Sound – The speed of sound waves, about
NOM – Number of Open Microphones in a sound 1130 feet per second in air.
system. Decreases gain-before-feedback by 3 dB every
time the number of open microphones doubles. Supercardioid Microphone – A unidirectional microphone
with tighter front pickup angle (115 deg.) than a cardioid,
Octave – The distance between two frequencies that is but with some rear pickup. Angle of best rejection is 126
either double or half the first frequency (e.g. 500 Hz to deg. from the front of the microphone.
1000Hz).
Voltage – The potential difference in an electrical circuit.
Omnidirectional Microphone – A microphone that picks Analogous to the pressure on fluid flowing in a pipe.
up sound equally well from all directions.

35
 Selection and Operation of Reference Information

AUDIO SIGNAL Shure Product Selection Charts

Processors

Mixers+Amplifiers

Model >> SCM262 SCM268 SCM410 SCM800 SCM810

Features:
Transformer-balanced input ●
Active-balanced input ● ● ● ●
Transformer-balanced output ●
Active-balanced output ● ● ● ●
Low-Z mic-level input ● ● ● ● ●
Line level input ● 1 ● ●
Aux level input ● ● ● ●
Mic level output ● ● ● ● ●
Line level output ● ● ● ● ●
Phono jack aux level output ● ● ●
Headphone output ● ●
Phantom power ● ● ● ● ●
48 V phantom power ● ●
VU meter
Peak meter ● ● ● ●
EQ ● ● ● ●
Tone oscillator
Linkable ● ● ●
Slate mic + tone
Limiter ● ● ●
Stereo operation ●
AC operation ● ● ● ● ●
Battery operation

1 Internal modification or optional accessory.

36
Reference Information Selection and Operation of

Shure Product Selection Charts AUDIO SIGNAL

Processors

Integrated Signal Processors

DSPs
Model >> DFR22 P4800
Features:
Inputs x outputs 2x2 4x8
Connectors XLR & Phoenix Phoenix
Rack space 1 rack 1 rack
Audio specs Dynamic range > 110 dBA Dynamic range > 100 dBA
Matrix Mixer Full matrix mixer Full matrix mixer
Front panel controls Preset selector for 16 presets. No front panel
Controls for DFR parameters controls
Front panel audio metering Mute, 20 dB, 0 dB, Clip LEDs for Full string metering for each input
each input and output and output
Automatic feedback reduction Drag and drop blocks for Drag and drop blocks for
5-, 10-, and 16-band single 5-, and 10-band
channel and stereo DFR single channel DFR
DFR filter removal Auto clear Hold mode
Additional processing Drag and drop blocks for GEQ, PEQ, cut/shelf, delay,
single channel and stereo compressors and limiters,
peak stop limiter, AGC, gate, downward expander,
ducker, crossover
External control options DRS-10 & serial commands (AMX or Crestron); contact
closures and potentiometers for preset, volume and mute.
Control pin inputs 4 8
Logic outputs None 8
Security Front panel lockout with password protected
password protected multi-level security
multi-level security
Shure link Yes Yes

37
 Selection and Operation of Reference Information

AUDIO SIGNAL
Processors

BIBLIOGRAPHY & ADDITIONAL REFERENCES

Bohn, Dennis "Linkwitz-Riley Crossovers," Pro Audio Reference.

Rane Corporation, Mukitelo, WA

Bohn, Dennis, and "Constant-Q Graphic Equalizers," Pro Audio Reference.

Pennington, Terry Rane Corporation, Mukitelo, WA

Brown, Pat "System Gain Structure," Handbook for Sound Engineers, 3rd Edition.
Focal Press, Boston, MA

Davis, Gary D., and Sound Reinforcement Handbook.

Jones, Ralph Hal Leonard Publishing, Co., Milwaukee, WI

Lyons, Christopher Audio for Distance Learning.

Shure Incorporated, Niles, IL

McMannus, Steven "Filters and Equalizers," Handbook for Sound Engineers, 3rd Edition.
Focal Press, Boston, MA

Vear, Tim Audio Systems Guide for Houses of Worship.

Shure Incorporated, Niles, IL

Whitlock, Bill, and "Preamplifiers and Mixers," Handbook for Sound Engineers, 3rd Edition.
Pettersen, Michael Focal Press, Boston, MA

ACKNOWLEDGEMENTS
The following individuals contributed to this publication, either with their words or their editing skills:

Luis Guerra
Tim Vear
Michael Pettersen
Cris Tapia

38
Reference Information Selection and Operation of

AUDIO SIGNAL
Processors

ABOUT THE AUTHOR

Gino Sigismondi
Gino Sigismondi is the Manager of Training in the U.S. Business Unit. A Chicago

native, Gino has been with Shure Incorporated since 1997. Gino earned his BS degree in

Music Business from Elmhurst College, where he was a member of the Jazz Band, as both

guitar player and sound technician. Gino was an Applications Specialist in Shure’s

Applications Engineering Department for 10 years. In this role he conducted product-

training seminars for Shure customers, dealers, international distribution centers and

company staff. He has also authored several Shure educational publications. In addition to

his work as a live sound and recording engineer, Gino’s experience includes performing and

composing, and sound design for modern dance and church sound.

39
 Additional Shure Publications Available:

Printed or electronic versions of the following guides are available free of charge.

To obtain your complimentary copies, call one of the phone numbers listed below

or visit www.shure.com.

• Selection and Operation of Personal Monitor Systems

• Audio Systems Guide for Video Production

• Audio Systems Guide for Houses of Worship

• Audio Systems Guide for Meeting Facilities

• Microphone Techniques for Studio Recording

• Microphone Techniques for Live Sound Reinforcement

Our Dedication to Quality Products

Shure offers a complete line of microphones and wireless microphone systems for everyone

from first-time users to professionals in the music industry–for nearly every possible application.

For over eight decades, the Shure name has been synonymous with quality audio.

All Shure products are designed to provide consistent, high-quality performance under the

most extreme real-life operating conditions.

United States: Europe, Middle East, Africa: Asia, Pacific: Canada, Latin America,
Shure Incorporated Shure Europe GmbH Shure Asia Limited Caribbean:
5800 West Touhy Avenue Wannenäckerstr. 28, 3/F, Citicorp Centre Shure Incorporated
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Phone: 847-600-2000 Phone: 49-7131-72140
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Email: [email protected] Email: [email protected] Fax: 852-2893-4055 Fax: 847-600-6446
Email: [email protected] Email: [email protected]

www.shure.com ©2008
©2007Shure
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