in August 2004
25 Years of DSP
in Sound Reinforcement
By Joseph Bocchiaro III, PhD,
|Audio enhancement systems
are being used in arenas, concert halls and multi-use
spaces, such as the Tokyo Internatioanl Forum (left)
and the Hamamatsu (Japan) Arena (right).
Digital Signal Processing revolutionized the market.
inclusion of DSP (digital signal processor, or processing)
in an ever-widening array of audio products, it is easy
to overlook the fact that DSP is now only 25 years old.
This makes 2004 an appropriate year to review the state-of-the-art,
and to postulate on its future. According to Richard F.
Lyon, in his column “DSP 4 You,” (acm queue,
March 2004), “…1979 saw the design of the first
wave of user-programmable DSP chips: the Intel 2920, the
NEC uPD7720 and the Bell Labs (AT&T) DSP-1….The
conversion of traditionally analog media to digital began
slowly, with the invention of PCM (pulse code modulation)
by Alec H. Reeves in 1937. The 1948 paper, ‘The Philosophy
of PCM,’ by Barry Oliver, John Pierce and Claude Shannon
of Bell Labs, following on their use of PCM for secret wartime
communications between Churchill and Roosevelt, laid out
an amazing and far-thinking view of why and how digital
was going to take over the analog world. The rest, as they
say, is history….” Our industry has eagerly
adopted this technology, as we will explore here.
Figure 1. Simple
DSP audio system.
Evolution of DSP Audio
As with many technology revolutions,
there is a steady parallel development of both the applications
and the underlying, enabling equipment and techniques. In
the field of sound reinforcement, the requirements to convert
sound into a digital signal (A/D), to process it in some
fashion and to convert it back to sound (D/A) are very demanding.
As the ability to create more complex processors has become
possible, the types of applications have evolved from traditional
analog circuit replacements to entirely new possibilities.
At the same time, the audience
expectation of the quality of the sound has increased, necessitating
higher and higher sampling rates and resolutions. The vast
consumer market has fueled the demand for cost-effective
DSP chips, and for entirely new programming techniques.
As will be seen, DSP has not merely replaced the common
analog tools that it emulates, but in many cases makes new
Speaking at the Richard C.
Heyser Memorial Lecture Series of the Audio Engineering
Society in 2000, James A. Moorer speculated on some of the
trends in his presentation, “Audio In the New Millennium.”
He postulates that “extrapolating over the next 20
years, it is concluded that the main problem facing digital
audio engineers will not be how to perform a particular
manipulation on sound, but how the amount of power that
will be available at that time can possibly be controlled.”
This excellent presentation
goes on to explore the possibilities of thousands of channels
of digital audio that could be possible because of the processors
available, but saddled with the limitations of humans attempting
to utilize it. Moorer looks beyond the current surround
sound techniques to recording and auditioning spaces that
more closely emulate the original sound field.
What you are reading here
coincides with the introduction of the first mass-produced
active field control product (see “Active Field Control”),
which is a major step toward the multiple-channel, calculation-intensive
future of audio DSP that Moorer envisions. The currently
available individual DSP techniques are defined, with an
eye toward how they are combined in a fully integrated,
fully programmable DSP future.
It is important to remember
that these techniques have been developed to create a more
realistic sound space, with adequate perceived gain, high
intelligibility, low harmonic and temporal distortion, high
signal-to-noise ratio and a correspondingly low listener
fatigue level. When applied judiciously, each of the following
techniques alters the signal in a manner that more closely
matches the source with the acoustic space, or alters it
in a creative manner that the artist wishes to convey.
DSP Signal Processors
The use of the term “processors”
here refers to a wide variety of equipment packaging that
differs between manufacturers, applications and integrations.
Some processors are stand-alone, discreet rack-mounted pieces
of equipment that require integration within an audio system.
Some are multi-purpose devices that allow customized configurations
to suit the application (see Figure 1). Some are software
products that are companions to computer applications, often
referred to as “plug-ins.” Still others are
modules available to be installed into consoles, amplifiers,
Whatever the case, the concepts
remain the same, and the designer is left with the challenge
of determining the appropriate overall configuration of
the system. Many DSP audio product manufacturers offer their
equipment in several of these different formats, allowing
designers still more options.
Besides the audio processing
aspect of the devices, DSP has opened the door to new and
more versatile audio system control possibilities. Because
the signals are processed in the digital domain, it is possible
to change parameters by programming or “on-the-fly,”
remotely or in conjunction with other equipment. This possibility
has transformed the live stage show industry in ways equally
important to the sound manipulation possibilities.
In addition, the digitization
of the signals allows them to be transported and stored
in a pristine fashion, allowing sound engineers the ability
to automate performance settings while maintaining unprocessed
signals for subsequent use. Control protocols such as MIDI
(musical instrument digital interface) have evolved to exploit
the capabilities of the DSP products, not just for recording
purposes, but for live sound applications.
Delay, a relatively simple
concept, is now taken for granted because it is incorporated
into so many different audio products. One of the most important
audio tools, delay is used on a “micro” level
to acoustically align transducers in a crossover environment,
and on a “macro” level to compensate for distances
in large spaces with distributed transducers. The development
of easily adjusted delay components has coincided with an
understanding of, and the ability to measure, time alignment
in a sound reinforcement system.
It may be argued that this
most simplistic contribution of DSP technology has improved
the intelligibility of systems more than anything else.
In addition, the development of the integrated circuit (IC)
“chips” to provide delay has financed and led
to many of the more sophisticated DSP applications, such
as reverberation and echo.
Figure 2. Feedback suppressor
in audio system.
Feedback suppression products
often are thought of as “insurance policies”
built into sound reinforcement systems because they have
been developed to automatically detect and correct feedback
that is occurring in the live environment. These devices
have become increasingly more sophisticated in their ability
to discriminate when and what constitutes feedback, and
have become higher quality components as sampling rates
and resolutions have increased.
In addition, there are now
several completely different types of feedback suppression
components, allowing designers to select which is the most
applicable in each system design (see Figure 2). The devices
are useful in cases where the Potential Audio Gain (P.A.G.)
of the system is close to the Needed Audio Gain (N.A.G.)
of the acoustic environment but not quite attainable. In
these cases the systems often are adjusted just below the
onset of feedback.
The original feedback suppressors are based on the “parametric
equalizer” approach, i.e., they introduce variable
“notch” filters into the digital signal chain.
The circuitry constantly samples the signal, and in the
event that the digital “signature” of feedback
is detected, a corresponding filter is activated at the
exact frequency. The depth of this filter is adjusted constantly
until the feedback is controlled.
This type of component has
been highly refined by reputable manufacturers such as Sabine,
Shure, dbx, Peavey, Roland, Behringer, Samson and others.
Of course there are limitations to these products, most
notably the occasional inability to distinguish certain
signals (such as sustained organ notes) from feedback. Because
these devices essentially are equalizers, they may “color”
the sound as well, due to phase inaccuracies and frequency
contouring, often causing undesired results. In a well-adjusted
sound system, however, their use as “insurance”
against unexpected events is invaluable.
A second type of feedback
suppressor is the “frequency shifting” variety,
popularized by the manufacturer Polyfusion. This device
constantly shifts the audio frequency of all signals passing
through by as much as 6Hz. This shift is not apparent to
an audience unless the amplified signal is compared directly
to the original sound. Signal regeneration is not possible
because the feedback has been frequency shifted from the
original and does not build upon itself.
The newest and most sophisticated
feedback suppressor is the Feedback Canceller, manufactured
by Wide-Band Solutions. This advanced application of DSP
digitally subtracts the feedback it detects on a continuous
basis. An advantage of this type of device is that the entire
digital signature is being analyzed constantly, making possible
other processing, such as noise cancellation.
Figure 3. Simplified
electronic acoustic enhancement system.
Manufacturers of DSP-based
audio products increasingly are incorporating noise suppression
or cancellation capabilities. The bane of any sound reinforcement
system, particularly in indoor environments, is a steady-state
noise such as from air-handling equipment and ducts, fans,
preamplifier hiss, etc. This noise contributes to an overall
decrease in the system’s signal-to-noise ratio, decreasing
intelligibility and causing audience fatigue.
The acoustical engineering
solutions to these problems usually are quite expensive
because modifications to HVAC equipment and architecture
often are required. The use of a digital device that samples
the background noise and cancels it is extremely useful,
particularly in audio- or videoconferencing applications.
Another important, early application
of DSP is dynamics processing. This includes compression,
limiting, expansion, gating and combinations of these such
as companding. Although this is an area where analog circuitry
is very advanced, DSP products are catching up to the adjustability,
sound quality and nuances of these tools.
The ability of DSP products
to individually process multiple channels simultaneously
makes them cost effective, and makes it possible to process
individual signals as never before. Besides stand-alone
units, dynamics processing is being incorporated in a wide
variety of product types, most notably mixers, preamplifiers
and matrix routers. The low cost of DSP has also made it
feasible economically to utilize multiple-frequency band
processors, such that only the aspect of the signal that
requires compression is affected, for example. Intelligence
and “look-ahead” modes will continue to become
popular to further automate dynamics processing, particularly
when sound engineers are not in attendance.
Equalization is an extremely
important tool in sound reinforcement systems. Most people
in the audiovisual industry are familiar with this process
because equalization is nearly as fundamental as amplification
in a signal path. Equalization is used for many purposes,
but primarily for preferential sound shaping, signal “flattening”
to compensate for room variations, and for feedback control
due to standing waves or other acoustic anomalies.
DSP equalization has not merely
replaced the traditional analog circuits, however. It has
made it possible for sound engineers to select the type
of equalizer desired in a system, whether graphic (discreet
frequency control) or parametric (variable frequency control),
or a combination of the two. Most importantly, advanced
DSP equalization promises to overcome the phase shifts present
at filter overlap regions, a major cause of distortion in
Mixing, Switching,Signal Routing
An essential and foundational
component in any audio system is the mixer, switcher or
signal router. These components have been transformed gradually
from audio/mechanical devices to audio/logic devices to
fully digital devices over the past 25 years. The possibilities
of DSP application in this area are tremendous.
The groundbreaking TOA Saori
pioneered the concept of an integrated mixer/router/signal
processor, and has been followed by a dizzying array of
products from other manufacturers. Some of these products
include the capacity to incorporate nearly all of the other
DSP tools discussed here. The use of these devices has in
many cases transformed audio systems into programming-intensive
designs as opposed to wiring-intensive designs. This is
due to the ability to configure a single component into
a multiplicity of applications.
The power of this is to allow
designers to create systems that may be reconfig-ured at
will, such as with multi-purpose rooms, divisible ballrooms,
performance spaces and sports venues. This is accomplished
by programming the DSP components to become mix-minus, zoned,
bi-amplified, tri-amplified or clustered circuits. It is
anticipated that DSP developments will further the trend
toward audio systems consisting of a multi-purpose DSP device
and audio amplifiers!
Audio effects include a wide
variety of techniques to change the timbral, temporal and
dynamic nature of sound, often at the same time and often
in conjunction with other effects. In essence, these effects
are combinations of the dynamics and equalization effects
described earlier. They include echo, reverberation, chorusing,
phase shifting, flanging, filter sweeping and others. Although
most of these effects are not used for typical sound reinforcement,
particularly with speech, they represent some of the most
creative and interesting applications of DSP.
These effects often are used
to give a performing artist a signature sound, for example.
Reverberation, however, the most common effect, is used
routinely in live sound reinforcement, whether for speech
only or for music. The quality of reverberation techniques
varies greatly, as we will cover later, and is one of the
most significant areas of development for DSP designers.
Pitch Correction,Shifting, Harmonization
One of the newer and most
significant DSP developments is in the area of pitch correction
and harmonization. Industry leaders such as DigiTech, Antares,
TC-Helicon and Eventide offer the ability for vocalists
to have their off-pitch notes automatically adjusted back
into exact pitch in a real-time fashion. Available as both
software and as hardware devices, this technique already
has revolutionized audio recording sessions and is making
its way into sound reinforcement systems.
This tool typically is applied
in a transparent fashion, but is also used as a special
effect for signature sounds. Pitch shifting is applied as
a companion effect, often to transpose a singer’s
voice when required. Harmonization, a close cousin to pitch
shifting, allows layering of the original, corrected signal,
with one or several pitch shifted notes above or below it.
Harmonization usually is programmed to allow for particular
intervals creating the effect of several people singing,
often with the characteristics of unique vocal groups such
as the Beach Boys or CSNY.
Although echo cancellation
(EC) and echo suppression (ES) are not used widely in typical
sound reinforcement applications, they are important and
sophisticated DSP tools. Developed primarily for audioconferencing
and videocon-ferencing systems, EC is invaluable in controlling
intelligibility and feedback in conference rooms. Pioneers
from the audioconferencing field such as Gentner and ASPI
Digital (now ClearOne and Polycom), and videoconferencing
companies such as Coherent, VTEL, Picturetel and Polycom
have incorporated this essential feature into microphone
mixers, audioconference hybrids and videoconference codecs.
This circuitry analyzes the
audio signal from microphones and uses a reference microphone
to determine the echo signature of the room. It then subtracts
this echo, including early reflections and subsequent reflections,
from the output signal. As this technique has developed,
features such as real-time accommodation have been added,
along with individual microphone channel circuitry. Further
developments may include longer cancellation durations,
decreased cost, higher sampling resolutions and incorporation
into other types of audio devices.
The most sophisticated application
of audio DSP has been developed over the past 20 years,
and is utilized primarily in large multi-purpose performance
spaces. There are several purposes for this technology,
and several variations. The concept of “electronic
architecture” or “electronic acoustic enhancement”
is applied to spaces where the inherent acoustics of the
space are not suitable for the type of performance. For
example, opera houses, symphony halls and drama theaters
typically are designed with very different reverberant fields
and effective degrees of diffusion.
Electronic architecture systems,
utilizing DSP technology, allow the acoustics of the space
to be transformed to best suit the situation by emulating
natural reflections. This technology is also applied to
spaces deliberately designed with little acoustic coloration
of their own, with the intention of adding the desired effect
to simulate a particular hall, a jungle, the outdoors, etc.
Pioneering work in this field has been accomplished simultaneously
by several manufacturers. These techniques and products
include the Assisted Resonance System (AR), the Multiple-Channel
Reverberation System (MCR), the Multiple Channel Ambiophony
System (MCA), the Variable Room Acoustics System (VRA),
the Early Reflected Energy System (ERES), the Reverberation
on Demand System (RODS), the Acoustic Control System (ACS),
the Lexicon Acoustic Reverberation Enhancement System (LARES),
the System for Improved Acoustic Performance (SIAP) and
the Active Field Control (AFC) System.
Each of these represents a
technique utilizing reverberant field sampling microphones,
DSP processing and loudspeaker arrays within the space (see
Figure 3, page 50). Control over routing, reverberation,
gating and other parameters allows electronic acousticians
to “tune” the space as desired.
The use of Audio DSP over the last 25 years has expanded
into many aspects of our lives. Cell phones, computers,
home theater systems, voice recognition and other applications
have followed the development of the equipment used in recording
and live events. As manufacturers continue to evolve processing
power, we may anticipate that DSP will reach into other
aspects of our lives, and further enhance the live event
Active Field Control
The Piano Salon (left) at
the recently opened Yamaha Artists Services, Inc.,
facility in New York City contains the first AFC system
installation in the U.S.
The newest development
in electronic architecture is the arrival of the Active
Field Control (ACF) System from Yamaha, announced
last month. Sound & Communications was given an
early look. This technology has been developed over
the past 20 years, although in highly proprietary
and customized form. As an integration product, it
allows acoustics designers and installers new options
in a lower-cost package. This may make electronic
acoustic enhancement achievable in more spaces than
The technology differs
from some of its competitors in that it is based on
the Assistance of Sound Field (A-SF) technology, which
utilizes a feedback loop to modify a room’s
existing acoustic properties without physical modification.
ACF is capable of changing the auditory impressions
of architectural sound, such as reverberation, loudness
and the perception of spaciousness. In addition, it
may enhance early reflections to improve the sound
quality in undesirable acoustic conditions, such as
under-balcony or stage areas.
ACF can also add reflected
sounds by extending reverb time through time-variant
FIR (Finite Impulse Response) Filters and increasing
the gain of the FIR filters—uniformly distributing
enhanced reflected sounds—and can intensify
the feeling of spaciousness by adding lateral reflected
sounds. The system is adjustable and may store presets
for configuring the room for different desired effects.
As a programmable DSP product, it is capable of being
upgraded with additional features in the future.
This system is newly
installed in the United States in the Yamaha Artist
Services, Inc. facility located in the Elizabeth Arden
building in New York City.
Joseph Bocchiaro, a principal
consultant with Electro-Media Design, Ltd. and manager the
EMD Western New York office, is the Chair of ICIA’s
ICAT, member of AECT and IACC and participates in Sound
& Communications’ Technical Council.