Chapter

10.7
Audio Processing and Control

Jerry C. Whitaker, Editor-in-Chief

10.7.1 Introduction
For more than 50 years, audio systems have used some form of equalizer technology. From the
earliest days of the tube-based Lang and Pultec filters to today's digital multiband graphic and
parametric systems, the equalizer has been relied upon to correct and enhance sound. It has also
formed the basis for many of the sophisticated automatic audio-processing systems in use today.
Perhaps because of this popularity, equalizers are some of the most overused and misunderstood
devices in the field of sound.

10.7.2 Types of Equalization

The frequency response curves of three simple equalizer circuits are shown in Figures 10.7.1
through 10.7.3. These curves represent the gain of each filter with respect to frequency. Figure
10.7.1 shows the response of a boost circuit, with the important characteristics labeled. The cen-
ter frequency is the frequency of maximum boost, so called because it marks the center of the
response peak. On each side of the center frequency is a point at which the amplitude is 3 dB
lower than the maximum level. The range in frequency from the first point to the second is the
bandwidth, an indicator of the sharpness of the filter. The smaller the bandwidth, the smaller the
range of frequencies that will be affected by the actions of the filter. Because the ear hears
changes in frequency on a percentage or octave scale, a filter of a given bandwidth will have sub-
stantially greater effect with a low center frequency than with a high center frequency. For exam-
ple, a 50-Hz-bandwidth filter centered at 10 kHz affects less than 1 percent of an octave and will
be virtually inaudible. A 50-Hz-bandwidth filter at 100 Hz, however, will affect almost an entire
octave, radically altering the sound.
The filter sharpness can be defined by dividing the center frequency by the bandwidth. The
resulting number is the Q of the filter. For a 50-Hz-bandwidth filter at 100 Hz, the Q is 2. For a
50-Hz-bandwidth filter with a center frequency of 10 kHz, the Q is 200. Typical boost filters
have Qs between 1 and 10.
Virtually all equalizers give the user control over the amount of boost or cut. The maximum
boost on most equalizers ranges from 10 to 15 dB. At small values of boost, the concept of band-

10-101
10-102 Production Standards, Equipment, and Facility Design

Figure 10.7.1 Basic boost filter response. (After [1].)

Figure 10.7.2 Constant-Q boost filter response. (After [1].)

width or Q becomes hard to define. If the maximum boost is 2 dB, how do you define the 3-dB
(down) frequencies? Most manufacturers, therefore, only specify the Q at full boost.
The filter response curve shown in Figure 10.7.2 maintains constant Q as the gain varies. This
causes the frequency range that the filter affects to decrease as the maximum boost is reduced.
The Q of the filter shown in Figure 10.7.3 decreases as the maximum boost is reduced, although
the shape of the filter remains the same. The filter affects all frequencies around the center fre-
quency by the same relative amount as the degree of boost is varied. Both characteristics are used
in commercially available equipment, although the manufacturer may not specify which type of
circuit is used. In practice, there are situations in which one has advantages over the other, but it
is difficult to say that one or the other is superior.
So far, we have considered boost circuits. The situation changes, however, when the filter is
set to produce a cut or reduction in gain over some frequency band. Figure 10.7.4 shows the
response of a simple filter circuit when the boost cut control is adjusted for a 10-dB cut (–10 dB).
The center frequency of the filter is defined as the frequency of minimum gain, the opposite of a
boost filter. The bandwidth, and therefore the Q, is defined by the frequencies at which the gain
is 3 dB less than maximum.
Figure 10.7.5 shows the response curves of a constant-Q filter when adjusted for varying val-
ues of boost and cut. The boost and cut curves are not mirror images of each other. This type of
filter produces sharp nulls in the response curve at the center frequency. There are applications in
which this characteristic is not desirable, and the inverse of the boost curve would be more
appropriate. Such a response characteristic is shown in Figure 10.7.6. The Q of the filter is not
 Audio Processing and Control 10-103

Figure 10.7.3 Constant-shape boost filter response. (After [1].)

Figure 10.7.4 Basic cut filter response. (After [1].)

Figure 10.7.5 Constant-Q boost/cut filter response. (After [1].)

10-104 Production Standards, Equipment, and Facility Design

Figure 10.7.6 Reciprocal peaking filter response. (After [1].)

Figure 10.7.7 Low-frequency shelving equalizer response. (After [1].)

constant in the cut mode. It is much lower than the corresponding boost position. This character-
istic is called a reciprocal peaking filter Most graphic equalizers use reciprocal peaking filters,
while most parametrics offer constant-Q filters.

10.7.2a Shelving Equalizer

The need often arises in professional applications to boost or cut all frequencies above or below
some selected frequency. A unit that performs this function is called a shelving filter. The fre-
quency-response curve for a low-frequency shelving filter is shown in Figure 10.7.7, and the
curve for a high-frequency shelving filter is shown in Figure 10.7.8. These filters are effective in
eliminating or producing frequency rolloff at the extremes of the audio band.
The boost or cut amplitude of a shelving equalizer is defined as the maximum deviation from
the nominal (flat) gain of the filter. For a high-frequency shelving equalizer, it is the high-fre-
quency gain minus the low-frequency gain. For a low-frequency shelving equalizer, it is the
 Audio Processing and Control 10-105

Figure 10.7.8 High-frequency shelving equalizer response. (After [1].)

Figure 10.7.9 A straight-line approximation of the shelving equalizer response. (After [1].)

reverse. The frequency characteristics of a shelving filter are described by the turnover fre-
quency, the stop frequency, and the transition ratio:
• The turnover frequency is the point at which gain changes from the nominal value by 3 dB. In
a shelving equalizer adjusted for a boost, the turnover frequency is the frequency at which the
gain is 3 dB above the midband value.
• The stop frequency is the point at which gain stops increasing or decreasing. This is taken as
the frequency at which the gain is within 3 dB of maximum or minimum, for boost and cut
settings, respectively. When small boost or cut values are selected, these definitions can
become unclear. It is common to approximate the shelving equalizer curve with straight lines
and use the points where they intersect as the appropriate frequencies, as shown in Figure
10.7.9.
• The transition ratio is the ratio of the stop frequency to the turnover frequency. It is analogous
to the Q of the peaking filter.
10-106 Production Standards, Equipment, and Facility Design

10.7.2b Parametric Equalizer

The most flexible equalizers on the market today provide independent control over all parameters
of the basic filter sections. These parametric equalizers offer three to five filter sections in one
package. Each section is usually independently adjustable. Often, the frequency ranges of these
sections are not the same, and there is considerable overlap between sections. For example, the
first filter might be adjustable from 20 Hz to 2 kHz, and the second adjustable from 50 Hz to 5
kHz. By staggering the sections this way, the entire audio band may be covered without requiring
excessive operating range from any one filter stage.
Most of the parametrics on the market divide the frequency adjustment ranges on each filter
into two or three bands. The filter is made tunable within these bands via a multiturn potentiom-
eter. To simplify tracking requirements, the range is generally limited to about 10:1. Selecting
between bands (almost always in factors of 10) is done with a switch that changes capacitors in
the filter. This approach provides fine resolution on frequency setting and allows approximate
calibrations to be written on the equipment front panel.
Some parametrics provide switch-selected modes, such as constant-Q/reciprocal response or
peaking/shelving. These features can significantly enhance the flexibility of the unit, eliminating
the need to select between response types in advance of the purchase. Constant-Q cut capability
is rarely needed except for sound system feedback reduction.
Flexibility and freedom from predefined frequencies and Qs make the parametric equalizer a
powerful tool. This flexibility also makes the parametric equalizer a complicated tool. However,
with experience, the operator can achieve precise equalization that can closely match the desired
characteristics. Because the unit is really several simple filter sections in cascade, the effect of
using two sections simultaneously is equal to the sum of their individual responses. This lack of
interaction between controls makes the adjustment task easier. By using the sections one at a
time, aberrations in system response can be removed in order of their significance. Complex
response adjustments can be made by connecting two equalizers in series.

10.7.2c Graphic Equalizer

Graphic equalizers are so named because they contain a bank of filters on octave or fractional-
octave center frequencies whose gain controls are arranged to create a graph of the resulting fre-
quency response on the front panel of the unit. The row of linear sliders provides a simple opera-
tor interface, enabling instant recognition of the gain in each frequency band. Graphic equalizers
are commonly available with frequency resolutions from one octave (9 or 10 sliders covering the
audio band) to 1/3 octave (27 to 31 sliders). Graphic equalizers are almost always fixed-fre-
quency, fixed-Q devices. These limitations are imposed primarily by cost and panel space con-
siderations.
Unlike the filters used in a parametric equalizer, filters in a graphic equalizer are wired in
parallel. As a result, the response with two sliders boosted or cut is not the same as the sum of the
responses with each slider advanced individually. The Q of each filter is selected based on the
spacing between center frequencies. There is, however, some latitude allowed. The actual Q, and
some subtleties concerning the way in which the filters are connected, affects how well the filter
responses combine. Graphic equalizers are generally reciprocal filter devices. The response
obtained with any control setting may be undone with a complementary setting of the controls.
 Audio Processing and Control 10-107

10.7.2d Hybrid Equalizer

Several hybrid approaches to equalization are available, as are products based on computer tech-
nology. In the hybrid category, there are several commercial units that seek to provide the advan-
tages of parametric and graphic equalizers in one package. One approach offers a graphic
equalizer that has a fine frequency-adjustment control under each linear fader. This allows the
user to trim the center frequency of the filter to exactly match the frequency of the desired
response peak or dip. Another approach offers a graphic equalizer combined with several tunable
notch filters. Still another approach helps the user visualize the resulting response of a paramet-
ric equalizer by configuring the boost/cut controls as linear faders arranged in a manner similar
to a graphic unit. The positions of the boost/ cut controls help convey the resulting response
curve.
A number of programmable equalizers that use analog filters controlled by a microprocessor
are also available. With a programmable equalizer, the user can set the desired frequency, Q, and
boost or cut on a digital display. Accuracy and repeatability are assured. The resulting curve is
displayed on a suitable device. By using a mouse or some other simple controls, the response
may be adjusted to any desired shape within the capabilities of the system. The primary advan-
tage that almost all programmable units offer is the capability of storing any number of equalizer
settings in memory and recalling them when needed.
At the other end of the technology spectrum, equalizers of various forms can also be pro-
duced entirely in software. Such systems are beyond the scope of this chapter.

10.7.3 Audio Processing Equipment

From the earliest days of recorded sound, people have been trying to funnel more signal into the
recording and broadcast media. Because the dynamic range of the human voice or common
musical instruments is much greater than that of conventional recording systems, devices were
developed to compress the dynamic range of signals, making the loud sounds softer and the soft
sounds louder. Unfortunately, some audio processors actually create problems when dealing with
the dynamic range of the human voice. These problems include amplifier overload, pickup of
room noise, and excessive sibilance in speech.

10.7.3a Gain Control

Dynamic gain-control devices provide an output signal that ideally differs from the original sig-
nal only in level. The shape of the waveform theoretically remains the same, but its size (voltage)
is made larger or smaller as necessary. System gain is an important characteristic for these types
of devices. Therefore, their steady-state operation can be described by plotting input level versus
output level on a graph. On a log-log scale (decibel output versus decibel input), the result would
be a graph similar to that shown in Figure 10.7.10, commonly referred to as a transfer curve. For
a conventional amplifier, the graph is a straight line at a 45° angle. The gain of the amplifier
determines where the line is positioned on the graph, but the slope is always the same.
10-108 Production Standards, Equipment, and Facility Design

Figure 10.7.10 Basic amplifier gain transfer curve, showing output in decibels referred to 1 mW
versus input in decibels referred to 1 mW (log plot). (After [2].)

Figure 10.7.11 Typical compression graph, showing how the output level decreases or increases
depending upon input level. (After [2].)

Compressor
A compressor is a device that increases the level of soft sounds and decrease the level of loud
sounds in a somewhat predictable manner. An example of this action is graphed in Figure
10.7.11. As the input signal amplitude increases, the output signal amplitude increases by a
smaller amount. When the input signal amplitude decreases, the output signal amplitude
decreases by a smaller amount. There is always a point at which the input level equals the output
level, called the unity-gain point. The slope of the curve is called the compression ratio. A com-
pressor whose output level increases by 1 dB for every 3 dB of input level increase is said to have
a 3:1 compression ratio.
Compressors come in two basic types: feedback and feed forward. The classic block diagrams
of each approach are given in Figure 10.7.12. The feedback-type compressor is the older and
 Audio Processing and Control 10-109

(a)

(b)

Figure 10.7.12 Simplified block diagrams of two common compressors: (a) feedforward compres-
sor, (b) feedback compressor. (After [2].)

more common of the two. The output signal level is sensed and fed back to the gain-control ele-
ment, which precedes it. As the input level is increased, the output level tries to increase. This is
sensed by the level sensor circuit, which drives the gain-control element in an effort to reduce the
amplitude of the output. Changing the gain after the level sensor changes the slope of the com-
pression characteristic. These circuits are easy to build and are self-correcting for errors in the
gain element or level sensor. However, the approach essentially guarantees that the output will
overshoot its final value when the input level is suddenly increased.
The feedforward configuration senses the input level and generates the necessary control
voltage for the gain element to make the output level change as desired. This avoids the over-
shoot problem, but places more stringent requirements on the accuracy of the level sensing and
control circuitry.
The graphs shown in Figures 10.7.11 and 10.7.12 are all straight lines on linear decibel
scales. This characteristic is obtained with level sensors that output a voltage proportional to the
decibel signal level. Voltage-controlled amplifiers (VCAs) also exhibit similar characteristics,
with the gain in decibels proportional to the voltage at the control input.
Some broadcast audio processors are composed of both a compressor and a limiter. The com-
pressor is used to reduce the dynamic range of the input signal, and the limiter prevents over-
modulation of the station transmitter. Limiters intended for AM radio applications sometimes
treat the positive and negative signal peaks differently. The processor can allow slight overmodu-
lation of the carrier in the positive direction, but not in the negative direction. This process keeps
the modulated signal linear, eliminating the distortion that would occur if the carrier were
allowed to disappear (the result of excessive negative modulation).
10-110 Production Standards, Equipment, and Facility Design

Figure 10.7.13 Transfer curve for a typical limiter, showing gain reduction of the output signal
above the threshold or turnover point. (After [2].)

Limiter
Broadcasters are often faced with a signal that is generally fairly constant in level, but occasion-
ally increases suddenly, causing the system to clip or to distort. To correct such signal fluctuation
requires a limiter, a device that operates as a standard amplifier for signals below some input
level, but becomes a compressor for signals above this level.
A transfer curve for a typical limiter is shown in Figure 10.7.13. The level at which the limiter
changes from unity gain to compression is called the threshold or turn-over point. This point is
usually variable, so that the threshold can be adjusted to match the requirements of the station
and the program material. Above the threshold, the compression function is characterized by the
slope of the transfer curve, as with a conventional compressor. The knee in the transfer curve
may be sharp, as shown in Figure 10.7.13, or it may be rounded. Some limiter manufacturers
suggest that the side effects from a rounded knee characteristic are less audible.
A compressor can be converted into a limiter by the addition of a diode before the gain-con-
trol element, as shown in Figure 10.7.14. The dc voltage from the threshold pot is applied to the
output side of the diode. This forces the signal level to exceed the threshold before compression
can occur. As with compressors, limiters can be designed as either feedback or feedforward sys-
tems. The feed forward-type limiter requires predictable characteristics in the level sensor and
voltage-controlled element. The feedback-type limiter does not require closely controlled ele-
ments if the exact compression slope is not a major concern. The limiting threshold is set by the
diode bias voltage or its equivalent. The limiting function may be performed by a variety of
devices, including an FET or a light-dependent resistor-LED combination.

Expander
Expanders are the functional inverse of compressors; they make soft signals softer and loud sig-
nals louder. The technique involved is graphed in Figure 10.7.15. The slope of the lines is always
greater than the 45° slope of an amplifier. If an expander produces an increase of 3 dB in output
level for a 1-dB increase in input level, it is said to have an expansion ratio of 3:1. This will
 Audio Processing and Control 10-111

Figure 10.7.14 Basic block diagram of a limiter. A typical compressor may be essentially con-
verted to a limiter through the addition of the diode shown. (After [2].)

Figure 10.7.15 Typical transfer function for an expander circuit. (After [2].)

exactly cancel the dynamic range compression of a 3:1 compressor. An expander is shown in
block diagram form in Figure 10.7.16. The only change from a compressor is the addition of an
inversion stage to make the gain increase with increased signal level.

Noise Gate
Broadcast, recording, and sound reinforcement pickups must sometimes rely on multiple-micro-
phone setups, such as stage performances or panel discussions. Unfortunately, when there is no
sound from the desired source, each microphone continues to pick up ambient noise. The noise
gate offers a method of turning down the gain of a microphone or other audio source when the
signal level drops below some preset value.
Noise gates are to expanders what limiters are to compressors. Above the threshold level, a
noise gate operates as a normal amplifier. Below the threshold, the gain decreases with decreas-
ing signal level, making soft sounds much softer. This effectively gates out or removes the noise,
but does not affect the desired signal. This characteristic (graphed in Figure 10.7.17) is similar to
10-112 Production Standards, Equipment, and Facility Design

Figure 10.7.16 Simplified block diagram of an expander. (After [2].)

Figure 10.7.17 The transfer function for a typical noise gate. (After [2].)

a limiter transfer curve that has been flipped diagonally. As with a limiter, there are two impor-
tant parameters: the threshold level and the expansion ratio. With proper adjustment of the
threshold level, the unit can discriminate between desired signal and unwanted background
noise. If there is insufficient level difference between the two, erratic changes in gain will occur
as the noise gate switches in and out of expansion.
The noise gate is shown in block diagram form in Figure 10.7.18. It is similar to both an
expander and a limiter. The inverter is used, as in an expander, to make gain increase with
increasing signal level. The diode prevents the level sensor output from exceeding the desired
threshold. When this occurs, the gain is clamped. Below the threshold, the unit functions as a
conventional expander.

10.7.3b Dynamic Processing

So far, only the steady-state behavior of gain-control devices has been examined. When the sig-
nal amplitude changes with time, such signal analysis is not so easy. Audio signals are, by their
nature, ac waveforms that go positive and negative many times per second. However, the signal
amplitude must be controlled without affecting the waveshape of these ac voltages. For example,
 Audio Processing and Control 10-113

Figure 10.7.18 Simplified block diagram of a noise gate. (After [2].)

Figure 10.7.19 Representative waveforms showing typical tone-burst response of a compressor.

The waveforms illustrate the problems caused by inherent response delay of a compressor. (After
[2].)

if the signal amplitude is adjusted too quickly, the waveshape will be changed, causing audible
distortion. If it is adjusted too slowly, the compressor or limiter will not be able to control the
peaks.
Figure 10.7.19 illustrates the response time of a typical limiter. A tone burst changing 20 dB
in amplitude is applied to the input. When the signal amplitude increases, the limiter requires a
given time to respond, resulting in an overshoot at the output. As the limiter adjusts to the new
gain level, the output amplitude decays to the desired value. When the signal amplitude drops,
the output level also drops by the same amount. As the limiter readjusts to the new signal level,
the output gradually increases.
10-114 Production Standards, Equipment, and Facility Design

Many audible problems are related to the time required to adjust the gain. These are the
pumping and breathing sounds sometimes heard as medium-level background sound is modu-
lated in amplitude by large-level sounds. If a limiter is designed to respond slowly to avoid these
problems, it will not be able to prevent peak amplitudes from exceeding the desired level. With
any design, there is a trade-off between audible side effects and incomplete processing.

Processor Features
There are many features available on compressors and limiters that may be important for a par-
ticular application. Most professional compressors provide a visual indication of the gain or gain
reduction taking place. Many units allow this display to be switched to also monitor the input or
output signal levels. This visual indicator can be helpful when adjusting drive levels in a system.
Some compressors and limiters allow the control voltages in the level-sense path to be tied to
other similar units for use in multichannel systems. If separate units are used on the two channels
of a stereo sound mix or broadcast, tying together these points will prevent the image from shift-
ing between the two channels because of unequal channel compression. A few units allow the
level-sensing circuitry to be patched for special effects. The sense input is usually inserted into
an EQ path for removing rumble or other noise that would disrupt the level-sensing action.
Compressors can sometimes be used as remote-controlled attenuators. Inputs for remote gain-
control setting can be handy in special situations. For instance, a simple variable de voltage could
control the speaker or headset volume in a studio. This approach eliminates the problems associ-
ated with routing audio through remote volume controls.

Performance Specifications
Standard audio distortion and signal-to-noise performance specifications are difficult to apply to
dynamic range-modifying devices. Because the gain, as well as the selected ratios and threshold
voltages, changes with input signal level, the performance measures also change. Noise gener-
ally becomes worse at high values of gain (low signal levels for expanders and compressors).
Distortion will sometimes peak at intermediate values of gain and sometimes at the extremes of
gain, depending on the type of gain-control element used. Specifications such as frequency
response, common-mode rejection, and maximum input level typically are comparable to other
types of signal-processing devices.
It is difficult to quantify the specifications unique to limiters and compressors in a way that
allows meaningful comparison of the audible performance of different units. Attack and release
times are only two aspects of dynamic behavior.
Distortion performance of the limiter during attack will significantly alter the perceived dis-
tortion with actual program material. Some compressors and limiters have marginal headroom
and hard clip on large inputs until the level sensor responds and reduces the gain. Other devices
are designed with more headroom or a soft clip circuit, which greatly reduces the level of high-
order distortion products during overdrive.

10.7.4 Audio Control Equipment

Audio quality has never been more important than it is today. Some of the most significant devel-
opment in professional audio can be found in mixing consoles and processing equipment.
 Audio Processing and Control 10-115

10.7.4a Audio Mixing Consoles

The mixing console is the hub of any audio facility. Audio boards, as they are often called, come
in basically two varieties: on-air and production boards. A console designed for use as an on-air
board for a radio or TV station is characterized by relatively simple, straightforward design. The
board will have limited equalization capabilities and few, if any, special-effects features.
Machine control, status feedback, and monitoring provisions are paramount. The function of an
on-air console is simply to get program material on the air, not to do anything fancy with it.
Production boards for broadcast or other applications are another breed. In the production
room, the emphasis is on versatility and features. Production consoles usually have provisions
for equalization on some or all input channels, special-effects send/return, and multiple-track
recorder operation. A production board will take longer than an on-air board to set up, but it
gives the user extensive capabilities to meet the creative needs of the facility. As the creative
needs grow, so do the requirements for a complex and versatile audio production board.
An important aspect of any audio console design is the human element. A board that can per-
form every imaginable function is of limited value if it cannot be easily understood by operators.
Many manufacturers offer both production and on-air versions of their audio consoles. Some
use common modules in each design and add extra features to tailor the console to the produc-
tion environment. These options include equalizer, filter, compressor, studio monitor, effects
send/return, and slate/talkback modules. The facility planner simply selects the types of module
required for the installation.
Expandable configurations have become standard among many console manufacturers,
allowing installation of a custom-designed system that meets a facility's exact requirements.
Ultimately, the expansion limits are set by the mainframe size. Careful consideration must be
given to selection of a mainframe that will meet future needs and lend itself to facility expansion.
Mainframes are commonly available for 10 to 34 input positions, in a variety of steps. Large pro-
duction consoles used for a major postproduction facility or network television production center
are commonly available in 48 to 64 input positions.
The functions of the console are dictated by the individual input, output, monitor, and special-
purpose modules. The channel elements vary in function and form from one manufacturer to
another, but some generalizations apply to most units.

Microphone/Line Input Module

The input channel module for an on-air console is relatively simple compared to its production
facility counterpart. Figure 10.7.20 shows the layout of a typical microphone/line input module.
The module includes on/off pushbutton switches that control audio signal flow and operate a
machine interface circuit. The interface may consist of a contact closure, low-voltage logic sig-
nal, or open-collector transistor output. Lamps in the buttons can be used to indicate signal flow
status or external machine status in the case of a line amplifier module.
A cue button allows the audio source to be monitored without disturbing the attenuator set-
ting. A number of options are available in the selection of an attenuator. User preference and
field experience are the best guides. A pan pot provides the means to balance a stereo signal
source or to position a mono signal in the desired stereo image, a feature typically used for
announce or interview microphones.
Output routing switches on the input module allow selection of the program, audition, or util-
ity buses. Channel input selectors permit two or more inputs to be used. Modules that have provi-
10-116 Production Standards, Equipment, and Facility Design

sions for external machine control usually include logic that allows output commands to follow
the input selector switches. Front panel or internal gain trim controls are sometimes provided to
allow convenient adjustment of input levels.
Figure 10.7.21 shows a typical input module for production room applications. Although the
configuration is different from that shown in Figure 10.7.20, the module functions basically the
same, with a couple of additional features. A multi-band equalizer has been added to allow the
operator to tailor the sound to the production requirements. In this example, a seven-band equal-
izer is utilized. Other designs offer two- or three-band equalizers and separate shelving or notch
filters. An equalization in/out switch allows the production setup to be bypassed when not in use.
An input-mode control enables the operator to select stereo, mono, left, or right signals for mix-
ing. Modules can also be equipped with provisions for cue, solo, effects send/return, multitrack
assignment, and external machine control.

Output Module
The output module for an on-air or production console is fairly straightforward. A master gain
control may be provided for each of the three common buses: program, audition, and utility. Indi-
vidual gain-trim posts may also be available on the modules for precise adjustment of channel
balance and output level.
Because the output of a mixing console usually must be distributed to several pieces of equip-
ment external to the board, an internal distribution amplifier is sometimes provided. Complex
distribution requirements, however, dictate the use of an external DA.

Monitor Module
The ability to monitor signals from various sources is of prime importance in both on-air and
production situations. Figure 10.7.22 shows a control-room monitor module that offers the fea-
tures desired in an on-air operating environment. Volume controls are provided for the head-
phones and the control-room monitor, cue, and talkback speakers. A bank of selector switches
allows the user to hear program, audition, or utility channels as well as any of several external
signal sources.
The headphone feed may follow the monitor speaker signal or be independently selectable.
Provisions for switching the headphone input from stereo to left, right, or mono mix may also be
useful to the operator.

Meter Display Options

A number of methods are available to monitor program levels on an audio console. These range
from the classic analog VU meter to solid-state designs of various types. Selection of the meter-
ing is usually determined by the personal preferences of the users. Regardless of the type of
metering chosen, two basic display characteristics are used: VU ballistics and peak program
meter (PPM) ballistics.
The VU meter provides an easy-to-read display of average program energy. It does not, how-
ever, provide accurate data on the presence of short-duration program peaks. The PPM is an
increasingly popular method of monitoring audio levels for professional applications. The
dynamic characteristics of a PPM are radically different from those of a VU meter. The PPM is
designed to follow and display the peak energy of the audio waveform. Probably the best method
of monitoring the output level of an audio console is to use both PPM and VU meters. With this
 Audio Processing and Control 10-117

Figure 10.7.20 The layout of a typical Figure 10.7.21 The layout of a microphone
microphone or line-level input module for or line-level input console for a broadcast
an on-air audio console. production console.
10-118 Production Standards, Equipment, and Facility Design

arrangement, the operator can regularly observe the VU meter

for program level setting and use the PPM to check for high-
level peaks that are not displayed by the VU meter. The
increased use of solid-state bar graph displays for program level
metering permits combined or switchable characteristics. A
simpler solution involves the use of a standard VU meter and a
peak indicator lamp or LED built into the meter that is set to trip
at a given reference point to display short-duration program
peaks.

10.7.4b Automating Audio Consoles

Through digital control techniques, consoles can provide more
features in a smaller amount of space. Even more important is
the enhanced operational control available through assignable
controls. To better understand how assignable controls work,
you must first understand how a large console is typically used.
Console operations can be broken down into several basic
tasks: input selection, equalization adjustments, processing
adjustments, output selection, level setting, and monitoring.
Historically, each of these tasks has required the use of separate
knobs, switches, and faders. However, digital technology has
opened the door to a different approach.
A console with an assignable work surface allows one or
more sets of controls to be assigned to many functions. With
digital storage of commands and perhaps some digital process-
ing, a fewer number of knobs can perform a wide variety of
tasks. The advantages arc reduced console size and easier opera-
tion.
Although large consoles are impressive-looking, realistically
an engineer can accurately set the controls of only one channel
or module at a time. A recent study showed that on a conven-
tional console equipped with 40 modules, an engineer uses only
2.5 percent of the controls at one time. The philosophy of
assignable controls takes advantage of this fact. Figure 10.7.23
shows an equalization module that can be assigned to any input
module on its parent console.
The disadvantage of assignability is that it makes visual
feedback a sequential accessing routine through the use of
shared displays, robbing the audio engineer of much vital,
immediately available information.
The degree to which a console design implements assignabil- Figure 10.7.22 The layout
ity is a matter of balance. A successful design incorporates the of a typical monitor speaker
correct trade-offs for all parameters, including which functions and headphone control
are assignable and which have dedicated controls and displays. module for either on-air or
production-room use.
 Audio Processing and Control 10-119

Figure 10.7.23 A single assignable equalization module for an audio console. This design fea-
tures four 16-frequency parametric EQ sections, plus sweep high-and low-pass filters.

As assignability functions are incorporated into a console design, it becomes easier to take
the next step and provide programmable features. Automated or programmed actions can be per-
formed by a console operating under the control of a computer. Such an approach allows the
operator to set up the equalization controls, presets, input channel switching, and even level set-
tings; store them; and recall the entire setup at any time.

10.7.5 References
1. Cabot, Richard: “Dimensions in Equalization,” Broadcast Engineering, Intertec Publish-
ing, Overland Park, Kan, August 1985.
2. Cabot, Richard: “Limiters, Compressors, and Expanders,” Broadcast Engineering, Intertec
Publishing, Overland Park, Kan., August 1986.
10-120 Production Standards, Equipment, and Facility Design

10.7.6 Bibliography
Bartlett, Bruce: Introduction to Professional Recording Techniques, Howard W. Sams, Indianap-
olis, IN.
Benson, K. Blair, and Jerry C. Whitaker (eds.): Television and Audio Handbook for Technicians
and Engineers, McGraw-Hill, New York, N.Y., 1990.
Beranek, Leo L.: Acoustics, American Institute of Physics for the Acoustical Society of America.
Everest, F. Alton: Successful Sound Operation, Tab Books, Fender Application Manual, pp.
2224–2244, Fender Musical Instruments.
Mapp, Peter: Audio System Design and Engineering, Klark Teknik.