Applications Tip:

 Compressor vs. Loudspeaker
Here is a common complaint made by the owner of a
damaged loudspeaker: "How could I have blown a
loudspeaker? I have a compressor!" Unfortunately, while
compressors and limiters help prevent audio transients
from causing clipping or possibly damaging a loudspeaker,
high-level transients are not the only cause of damaged
loudspeakers. In fact, over-compression of the audio signal
can contribute to premature loudspeaker failure.
It is standard practice to use an amplifier with a power
rating at least twice the loudspeaker’s continuous power
rating (e.g. use a 200 watt amplifier for a 100 watt
loudspeaker). The extra headroom afforded by the larger
amplifier allows for peaks in the program material to be
delivered to the loudspeaker without clipping. The majority
of the amplifier power goes largely unused since the
average level of an uncompressed audio signal is
considerably lower than the peaks. Highly compressed
signals have an average level much closer to the peak
level. If the level of the compressed signal is raised to take
advantage of the additional amplifier power (thereby
making it louder for the audience), the average power
delivered to the loudspeaker may be more than the
continuous power rating of the loudspeaker, overheating
the loudspeaker’s voice coil and causing failure.
As with all audio processors, using a compressor
does not eliminate the need for proper system operation.
Though a compressor or limiter is essential for reducing
transient peaks, excessive compression is the enemy of
the loudspeaker.

Compressors

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

Types of Audio Processors

crossover feeding three power amplifiers is called a triamplified
system. If clipping occurs in the low frequency
amplifier, the higher frequency harmonics created by the
clipping are reproduced only by a woofer that has very low
output at high frequencies, thus reducing the
audibility of the distortion. The use of active components
also offers smaller size and more repeatable production
due to better tolerances.
Quite often, a sound system combines elements of
both passive and active crossover networks. These types
of systems typically use an active crossover to provide a
separate subwoofer output for low frequencies, while a
passive crossover in a two-way loudspeaker divides
mid- and high frequencies. This could be described as a
three-way, bi-amplified sound system.
Most active crossovers allow for control of crossover
frequency and level at each output. DSP-based crossovers
typically offer greater adjustment, providing the user with
selectable filter slope, filter type, and polarity reversal.

FILTERS AND EQUALIZATION

Filters are signal processors that affect frequency
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 frequencydependent
loss in telephone lines, some form of frequencydependent
filtering (or equalization) is found in all but the
most basic of sound systems. The simplest form of filter is
the tone control, basically a filter that attenuates high frequencies
above a predetermined frequency. Equalizers are
typically characterized by combining several filter sets to
offer more precise frequency response shaping.
Historically, filters were passive devices capable of
attenuation only. The frequency range and amount of
attenuation were achieved with capacitors, inductors, or a
combination of both. Favorably, passive filters do not
require power and do not generate noise. The large size
and expense of discrete components, however, precludes
the ability to develop equalizers with larger numbers of
filters and more precise control of frequency and level.
Active filters allow for fast, easy tuning and the ability to
add gain, using smaller components at lower cost. Tone
controls employing active filters can be found on even the
most inexpensive home stereo systems. In this scenario
there are typically two controls, treble and bass, which
correspond to filters that affect low frequency and high
frequency response. Since they are active, these tone
controls are capable of cut or boost.
Simple filters that affect a broad range of frequencies
are divided into four types: high pass, low pass, band
pass, and band reject. High pass filters, as the name
implies, allow high frequencies to pass, and low pass filters
do the same for low frequencies. It is often more convenient
to think of these filters in terms of the frequencies that they
cut instead. High pass filters are also known as low cut
filters, and low pass filters are known as high cut filters, but
their function is the same and these terms can be used
interchangeably. 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
relates 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

VOLUME (GAIN) CONTROL

Although often overlooked as an audio processor, a
simple volume (or gain) control fits the definition. Volume
adjustments can be made at several points within the
sound system, from the microphone inputs on the mixer all
the way to the inputs of the power amplifiers. Volume
levels are typically manipulated in one of two ways:
continuously variable adjustments, such as those made by
rotary potentiometers or faders, or fixed attenuation such
as that provided by a pad.
If adjusting a volume control adds amplification to the
audio signal, it is said to be providing gain. The volume
control that adjusts the amount of amplification added at a
mixer’s microphone input is sometimes referred to as a
gain (or trim) control, since the volume potentiometer is
controlling the gain of the microphone input’s preamplifier.
The function of this gain control is to match the input
sensitivity of the device to the level from the source.
A second type of volume control acts as an
attenuator, basically a continuously variable resistor that
adjusts the amount of signal allowed to pass through it. No
additional gain is provided by the volume control. The
volume control on an electric guitar is an attenuator. These
devices are often referred to as passive volume controls,
since they do not require any power. Occasionally, a
volume control will combine attenuation with gain. Faders
on a mixing console typically provide attenuation below
the "0" indication, and gain above that point.
Pads allow input stages to accommodate a variety of
signal levels. Microphone inputs typically feature an input
attenuation pad of some kind to reduce the sensitivity of
the 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.
While volume controls are the simplest of all audio
processors, they often the most misused. Correct
calibration of the various volume controls in a sound
system is known as proper gain structure.