Four dimensional acoustical audio system

A four-dimensional acoustical audio system utilizes both spatial and temporal signal processing to maximize the depth, width, and perceived directionality of the acoustic field with respect to the binaural auditory system. Transducers are critically placed within the confines of a reverberant enclosure, and each transducer is assigned a specific bandwidth of sound either in mono or stereo, depending upon the spatial location of the transducer to take advantage of the binaural auditory system of the intended listener. The system minimizes the number of transducers and electronics required to achieve the effect of a live performance, regardless of the dynamics of the enclosure in which it is placed. The loudspeaker system can be used in small enclosed volumes such as an automobile or extended to volumes the size of a motion picture theater or concert hall and even outdoors.

FIELD OF THE INVENTION 
The invention relates to audio systems, and is more particularly concerned 
with spatial and temporal signal processing techniques for loudspeaker 
design to achieve optimal psychoacoustic impact. 
BACKGROUND OF THE INVENTION 
The reproduction of music, identical to that which would be perceived in a 
concert hall or a live performance, has been the objective of many in the 
audio industry for years. In more recent years, digital signal processing 
has often been considered for the reconstruction of a sound field by 
concurrently measuring the acoustic response of the field and then 
modifying the input to an array of loudspeakers to produce the appropriate 
velocity and pressure within the fluid medium. As detailed in a recent 
publication by Nelson, P. A., 1994, "Active control of acoustic fields and 
the reproduction of sound," Journal of Sound and Vibration, 177(4), pp. 
447-477, this approach is somewhat ludicrous in terms of practical 
implementation. One need look only to the Kirchhoff-Helmholtz integral 
equation to verify this statement. In theory, it is possible to 
identically recreate a sound field within a volume by placing monopole and 
dipole sources about that volume and reproducing the pressure and velocity 
field. As detailed by Nelson, the linear separation between discrete 
monopole/dipole source elements used to generate a planar continuous 
source array should not exceed a half wavelength (.lambda./2) at the 
frequency of interest. Thus, to reproduce a sound field identical to the 
original within a spherical volume would require approximately 
4.pi.D.sup.2 /.lambda..sup.2 individual source elements. In words as 
opposed to mathematics, to identically reproduce a sound field with an 
array of transducers over a frequency range extending from 20 Hz to 10 kHz 
and for a sphere of 10 m diameter would require over 1 million individual 
sources| Even if the frequency range were limited to 1 kHz and the 
diameter of the sphere were reduced to 1 m, approximately 100 transducers 
would be required. 
This example emphasizes the need for a new approach and philosophy to the 
reconstruction of sound fields that minimizes the number of sources 
required, the complexity of the electronics, and yet takes advantage of 
the physiological processes by which humans hear, the binaural auditory 
system, to maximize the depth, width, and perceived directionality 
associated with the sonic image. It is desirable to provide a system that 
generates a sound field from the perspective of the physical or acoustical 
range as oppose to the artificially electronically induced realm. 
It has been determined that a typical stereophonic sound reproduction 
system designed for realistically recreating a sound stage according to 
AES standards for hi-fi, predominantly contains at least two distinct 
signals, one containing the information pertaining to that which should be 
heard by the right ear of the listener and one containing the information 
pertaining to that which should be heard by the left ear of the listener. 
Contemporary sound reproduction systems rely on three subsystems of 
transducers as illustrated in FIG. 1. One subsystem is a left enclosure 2 
directed at a listener or listening area 4 from the left. Another 
subsystem is a right enclosure 6 directed at a listener or listening area 
4 from the right. Each subsystem produces the appropriate sound for the 
side of the listener 4 at which it is directed. The third subsystem 
contains the information for the sound at the lower frequency limit of 
human hearing, which in typical systems is 500 Hz and below. The third 
subsystem transducer 8 may be placed centrally in front of or behind the 
listener 4 because it is generally accepted that these low frequencies are 
somewhat omni-directional as a result of the characteristic distance 
between the ears, approximately 0.2 m for a typical person. 
In the subsystems devoted to the left and right channels, there are often 
multiple transduction devices mounted within each respective speaker 
cabinet, each transducer capable of producing a limited frequency range 
with appropriate crossover networks to "match" the sensitivity of each 
transduction device over the intended bandwidth. These "satellite" 
loudspeaker systems incorporate only one aspect of psychoacoustic 
perception into the design. 
According to one aspect of the invention, to achieve greater depth and 
width of field acoustically, one must separate the transduction devices 
both spatially and temporally to match what is known about the 
directionality of sound in the azimuthal plane (referring to azimuthal in 
FIG. 1 and median or sagittal plane perpendicular to the azimuthal and 
bisecting the human symmetrically), yet known audio design, with a focus 
on stereophonic experience, does not account for this separation of 
auditory responsibility. 
Stereophonic sound can best be described as the science of three 
dimensional sound. It is an ever evolving, inexact science involving 
physics, psycho-acoustics and audio electronics. In its simplest form, the 
most common stereophonic standard consists of two channels of signal 
information. 
Some of the criteria used for stereophonic realism involve spatiality--the 
ability for a given sound to be captured in a hall, where the various 
reflections are recorded and later played back; timbre--the color of the 
sound; and phase linearity--all frequencies arriving in time, in phase 
without distortions. These and other issues, such as dynamics, 
intermodulation distortions, mechanical distortions and a host of other 
objective and subjective concerns make up a glimpse into the world of 
stereophonic sound. 
Yet, audio design using a stereophonic model is driven by a restricting 
standard. This standard requires that the way in which an artist is 
recorded in the recording studio, or on stage, is the same way in which 
the listener will hear the recording played back. This standard has 
essentially been centered around a two channel, two speaker model. 
In a dual loudspeaker set up, two speakers are set apart from each other 
(in front of the listener), at a distance optimum to produce a 
realistically proportioned illusionary sound stage. This sound stage is 
the result of cross talk, arrival time to the ear-brain relationship in 
time and space relative to the original recording. The illusion, however, 
is just that--an illusion, an effect, and this effect has become the 
industry standard. 
With this standard, manufacturers, inventors and marketeers compete to 
perfect reality within the confines of the standard, through such things 
as improved transparency, blossom, space, clarity, timbre, imaging, sound 
stage and a host of other objective and subjective goals which define the 
criteria. 
The prior art demonstrates an objective to improve upon one or more of the 
goals of this industry standard. Some manufacturers and inventors have 
recognized that all frequencies do not need to be contained in two 
loudspeaker boxes, but that one can separate the low frequencies from the 
high frequencies and produce a desirable and even improved sound stage 
using "three" loudspeaker enclosures. 
In an automobile, which provides an acoustically sealed environment, the 
illusion can be enhanced when two more "rear fill" speakers are employed 
at the rear deck of a sedan to aid in the reflective ambience otherwise 
lost through carpet and seats. 
Yet, the advances strive for improvement in a two speaker stereophonic 
model. The separation of transducers dedicated to different frequency 
ranges, commonly referred to as satellites, is largely driven by the need 
to separate the tasks of amplifiers so that the increased power needed for 
very low frequency signals does not undermine the signal quality in the 
mid and upper frequency range. 
Beyond the two channel "hi fi" stereophonic playing field, there is motion 
picture theater sound and its advances, for example, Holman THX as 
described in U.S. Pat. No. 4,569,076. Yet, again, the objective is not to 
dissimilar from that of high fidelity stereophonic systems. Here, the 
criteria for excellent theater sound requires that the audience hears what 
the director heard in the screening room. 
Thus, whether a given demand calls for stereophonic two channel sound, or a 
multi-channel movie theater matrix, the loudspeakers, amplifiers, signal 
processors and wires are all designed to perform within the confines of 
the established standards. 
As to the standards themselves, there are recording processes which enable 
these standards to exist, and, within the processes are standards, such as 
Dolby. 
Additionally, it is important to recognize that all systems and all 
standards have thus far utilized and specified the need for full band 
width audio in nearly all cases. As used throughout the application, full 
band width is defined in ASO as 20 Hz to 20 kHz and in ISO as 16 Hz to 16 
kHz. 
In the cases in which frequency fragmentation groups have been employed, 
such as the types used for large concerts, and other prosound and high end 
applications, the fragmentations are phase arrayed and are meant to be 
constant-directivity-based solutions. These bi-amped systems are no 
different from a traditional loudspeaker in terms of their end goal, and 
as such are not discrete self-contained systems designed to perform for a 
specific discrete processing function. 
In the past ten years, signal processing, and in particular, digital signal 
processing has become the most significant breakthrough to the science of 
stereophonic, or three dimensional sound. These digital signal processors 
(DSP) are programmed to perform tricks to fool the ear into believing that 
the sonic image is bigger than it really is, or more life-like, or more 
three dimensional. Yet, the focus of the processing of a signal has been 
on the input side and not specifically from the acoustical side. 
The ear-brain relationship may be tricked into believing something is 
larger, or more reverberant, through illusionary psychoacoustic DSP, but 
the ear is an amazing instrument. With all the advances in audio 
electronics, an average person can usually discern the difference between 
a recorded sound and the real thing. 
In a movie theater environment, we are suspended in our willingness to 
believe something is real, when in fact we know it isn't. We marvel at the 
technological wonder on the screen of a jet fly-by, or the soft splash of 
the whale, or the screeching tires of the gangsters' car. How realistic| 
Yet, perhaps it is only when we are in a theater projecting sound through 
a state of the art sound system and a real thunder clap strikes outside 
the theater that we fully come to understand reality versus the theater 
sound. 
We are a society approaching a paradigm shift in our culture. Perception 
itself is being questioned throughout the arts and the sciences. Virtual 
Reality, Multi-Media, and MIDI-based music synthesis are examples of the 
strives in technology to meet the thirst for reality in the reproduction. 
These converging technologies, combined with advanced simulation 
technologies, previously limited to military training, are now being made 
available to the average person at special venue amusement parks and 
attractions. Soon, these new formats will enter our homes. 
The ability for us to "enter the experience" cannot occur using 
conventional audio technology and loudspeaker systems, regardless of how 
many channels are employed. Our philosophical approach to acoustical 
applications must be revised before this can occur. The very process of 
sound distribution here entails a uniquely different approach. One cannot 
simply place a device designed to produce a given result and put in into 
an environment it was never intended for and expect ideal results. 
Today, musicians and composers are no longer limited to having to perform 
on stage to a traditional audience the way it has been done for centuries. 
Through MIDI and multi channel recording, one composer, alone, can bring 
to life the sounds previously requiring an entire symphony orchestra. No 
longer are there boundaries or restrictions. Yet, we play back through the 
same loudspeaker. loudspeaker used to provide stereophonic standards. 
Within the stereophonic standard, there is a constant drive to achieve 
linearity of response. This goal is so overwhelming, that compromises of 
other aspects of sound are made to achieve it. The focus shackles the 
development of systems that more accurately and efficiently emulate 
reality. 
SUMMARY OF THE INVENTION 
It is an object of the invention to provide a four dimensional acoustical 
audio system that combines the selection of transducers, the placement of 
those transducers and the spectral separation of frequency to the 
transducers to optimize the psychoacoustic effect to the observer. 
It is another object of the invention to provide the psychoacoustic 
experience to the observer with a focus on the binaural auditory system of 
the observer and not the audio source. 
The achievement of these objects according to the invention requires a 
merger of different aspects, namely, transducer type, spatial placement 
and frequency fragmentation of audio design, without limitation to the 
stereophonic models of the existing technology. The invention manifests 
itself in a variety of embodiments set forth more fully below, but each 
premised on a discovery of the merger of distinct aspects of the audio 
system design. 
According to the invention, however, these individual aspects should 
preferably not be used alone. For example, while an excessively large 
quantity of transducers can be utilized to achieve a desired auditory 
effect, the optimization of transducer type with placement and appropriate 
frequency separation can reduce the number of transducers needed to 
produce the effect and yet produce a more realistic effect. 
Spatial placement according to the invention has the function of 
establishing the acoustic framing of the auditory experience being 
created. According to the invention, the placement varies according to the 
application and is coordinated with the transducer selection and frequency 
fragmentation to optimize the experience of the application. 
The acoustic frame established can be varied as to what frequency groups 
are chosen for a particular job. In a theater environment or other setting 
in which the audience is oriented toward a screen or stage, a 360.degree. 
tweeter placed behind the listener will cause the pinna to recognize a 
slight "spatial" increase in the room. When power balanced together with 
the spectrum (4 kHz and up) the psycho-acoustic "illusion" begins to place 
the listener "IN" the experience. Thus, it is the merger of transducer 
type, placement and frequency separation that optimizes the experience. 
A four dimensional acoustical audio system has been designed which takes 
advantage of both spatial and temporal signal processing in accordance 
with the process by which the binaural auditory system processes sound to 
increase the width and depth of the "sonic image" and increase the "sweet 
spot" typically associated with stereophonic sound reproduction. 
In an embodiment according to the invention, the effect is achieved, for 
example, by placing one or two sub-woofers with a preferably 
summed-to-mono input ranging in frequency from 0 Hz to 250 Hz in one or 
two of the front corners of the enclosure and a mid-bass driver with a 
preferably summed to mono input ranging in frequency from 150 Hz to 3 kHz 
at the "center stage" of the audience. A stereophonic image is created 
with a left and right audio loudspeaker having inputs ranging in frequency 
from 900 Hz to 12-16 kHz, each preferably placed midway between the front 
and back of the enclosure on the left and right walls of the enclosure, 
respectively. By placing the drivers centrally on the side walls of the 
enclosure, maximum directionality of the sound source is achieved by 
interaural intensive difference processing, used by the ear to determine 
the direction from which a sound emanates at high frequencies. A fourth 
driver, a high frequency device having a preferably summed-to-mono input 
with a frequency range of 4-6 kHz to greater 20 kHz, is placed at the rear 
of the enclosure to create the effect of a "live" room. By placing this 
driver at the rear of the audience, the pinna naturally filters the sound 
radiation and thus delivers an attenuated sound to the ear which is 
perceived as a reflection and thus generates the effect of a more 
reverberant sound field. 
The resulting acoustical field not only creates an auditory environment for 
the observer in the enclosure that places the observer "in the experience" 
but also emulates the reality such that an observer outside the enclosure 
senses a realistic acoustical image is occurring within the enclosure. 
The invention in its various embodiments provides a new approach to sound 
design by synergistically combining transducer selection, placement and 
frequency fragmentation to provide realistic sound experiences beyond the 
limits of conventional stereophonic models.

DESCRIPTION OF PREFERRED EMBODIMENTS 
The invention relates to the reproduction of sound from recordings made on 
various media to imitate the initial sound produced at the time of 
recording. The invention is suitable for use within enclosures with 
volumes ranging from that of a typical automobile to a theater with a 
volume of over 400,000 cubic feet. The invention even has application in 
outdoor environments. This disclosure is directed to embodiments of the 
invention relating to the creation of a sound stage for listeners oriented 
in a particular direction, such as toward a motion picture or video screen 
or performing stage. The experience created, not only realistically places 
the listener in the room or enclosure in the experience, but also projects 
a realistic image to an observer outside the enclosure or room that the 
performance is occurring inside the room. The invention can have other 
applications in commercial environments to create a homogeneous sound 
field along a horizontal plane of listening, such as the ear level of 
seated diners in a restaurant. These commercial applications of the 
invention are explored in a copending application. 
The objective of the four dimensional acoustical audio system, through 
certain embodiments, is to increase the width and depth of the sonic image 
presented to the audience and thereby create a widened "sweet spot" so 
that the sound reproduction has greater uniformity and can be enjoyed by a 
variety of listeners, independent of their specific position within the 
enclosure. To achieve this effect, both spatial and temporal signal 
processing are used to shape the acoustic field. Spatial signal processing 
relates to the specific location of the transducer (driver) within the 
reverberant enclosure and has been applied to the control of reverberant 
structures in recent years as outlined by Clark, R. L., R. A. Burdisso and 
C. R. Fuller, 1992. "Design approaches for shaping polyvinylidene fluoride 
sensors in active structural acoustic control," The Journal of Intelligent 
Material Systems and Structures, 4, pp. 354-365; Bailey, T., and J. E. 
Hubbard, 1985. "Distributed piezoelectric-polymer active vibration control 
of a cantilevered beam," AIAA Journal of Guidance and Control, 6 (5), pp. 
605-611; Burke, S. E., and J. Hubbard, 1987. "Active vibration control of 
a simply supported beam using a spatially distributed actuator," IEEE 
Control System Magazine, pp. 25-30; Crawley, E. F., and J. de Luis, 1987. 
"Use of piezoelectric actuators as elements of intelligent structures," 
AIAA Journal, 25(10), pp. 1373-1385; Lee, C. K., and F. C. Moon, 1990. 
"Modal sensors/actuators," ASME Journal of Applied Mechanics, 57, pp. 
434-441. Temporal signal processing relates to the use of active filters 
to selectively achieve desired bandwidths of operation for specific 
transduction devices. 
As used herein, four dimensional refers to the use of the three spatial 
dimensions and time as a fourth dimension to create the acoustical sound 
field desired. 
Combining both spatial and temporal signal processing affords the 
loudspeaker designer with a degree of freedom and flexibility not 
previously explored to its full potential. Spatial and temporal signal 
processing can be combined for optimal performance with respect to the 
binaural auditory system, namely human ears (the transducers for which 
this system is intended) as opposed to a microphone placed at some fixed 
distance in an anechoic environment as in conventional loudspeaker design 
performance assessment. The loudspeaker systems of this invention are not 
designed to meet some specified frequency response characteristics in an 
anechoic environment as the transducers are spatially separated within the 
enclosure, independently filtered (actively), and amplified to recreate 
the desired acoustic response. In contrast to traditional design 
implementations, the acoustical systems envisioned by the invention are 
spatially and temporally optimized within the enclosure to take advantage 
of the binaural auditory system and maximize the perceived width, depth, 
and directionality of the sound field. 
Because the loudspeaker systems are designed for the binaural auditory 
system, it is appropriate to review this biological system here. 
Stereophonic loudspeaker systems take advantage of the human ability to 
resolve the direction from which sound emanates. Binaural hearing is 
required to physically locate stimuli in the real world, and there are two 
basic methods by which the location of a sound source is determined. Each 
is distinctly different and has an effective bandwidth of operation. 
Firstly, the interaural time difference (ITD) in the arrival of a sound 
wave at each respective ear can be used to determine the direction from 
which the sound emanated. At relatively low frequencies, below 1500 Hz, 
the wavelength of the sound wave is greater than the characteristic 
dimension between the ears (approximately 0.2 m for a typical person). 
Thus, a distinct time delay in the propagation of the sound wave can be 
resolved. While this method of resolving the direction can be effective up 
to 3000 Hz, it has limited accuracy between 1000 Hz and 3000 Hz as the 
acoustic wavelength decreases. At frequencies greater than 3000 Hz, the 
primary method of resolving the direction of a sound source is based upon 
the interaural intensive difference (IID). At higher frequencies and 
decreasing acoustic wavelength, sound waves are partially blocked by the 
effective "baffle" created by the head if the source is not positioned 
directly in front of the listener. Thus, variations in sound intensity 
presented at each ear help in discerning the location of a source at 
relatively high frequencies. 
In reverberant, enclosed, sound fields, the sound originating from a source 
will bounce off the walls several times in various directions until it 
decays sufficiently to be inaudible. However, for transient acoustic 
waves, extensive testing has shown that the direction from which a sound 
first arrives is perceived to be the location of the source even if the 
reflected (delayed arriving signal) is larger than the first arriving 
signal (Moore, 1989). 
Oddly enough, the frequency range in which directional information is 
difficult to discern by either ITD or IID is in a range of 1 kHz to 3 kHz 
where the sensitivity of the ear to sound is quite high. Accordingly, a 
single mono sound source placed in front of an audience with an upper 
frequency limit of approximately 3 kHz and will not have a dramatic effect 
on the perceived direction of the sound over the audible range, but can be 
effectively used to create the center stage. 
At higher frequencies, it is imperative to have both left and right stereo 
signals if stereophonic imaging is desired. In fact, based upon the IID 
method of detecting the position of a sound source, the optimal location 
of the stereophonic transducers producing sound in the approximately 900 
Hz to 16 kHz bandwidth are at opposite sides of the listener to maximize 
the IID. At low frequencies, the acoustic wavelength is so long that a 
listener cannot accurately resolve the direction of the source (because 
the sound heard at either ear is nearly in phase), so a sub-woofer (0 to 
250 Hz bandwidth) can be placed in the corner of the enclosure (at the 
front) to maximize the coupling to the room dynamics. Finally, a single 
mono high frequency device (approximately 4-6 kHz to &gt;20 kHz bandwidth) 
can be located near the rear of the audience or centrally overhead to 
achieve the effect of greater reverberation. The pinna (outer ear) serves 
to diminish the sound by virtue of reflection and diffraction at high 
frequencies when the sound wave is presented from behind. Acoustic waves 
reflected in a reverberant field also impinge the ear at reduced 
intensities than that of the original wave. Thus, placing a higher 
frequency driver at the rear of the audience can achieve the 
psychoacoustic impact of a more "live" acoustic field as opposed to the 
more complex use of full-bandwidth transducers and signal processing to 
achieve the same desired effect. 
All of the prior considerations have been taken into account by the design 
of one embodiment of the four dimensional acoustical audio system set 
forth herein. Conventional performance specifications in terms of the 
system sensitivity lose meaning here because the sound system provided by 
this invention is designed for the transduction devices used in the 
binaural auditory system, not a microphone positioned at a fixed distance 
from a speaker mounted in a baffle. Quality transduction devices are used 
in this system since the timbre that each device is capable of reproducing 
is critical to the overall performance of the system. In addition, the 
relative sensitivity of each transducer is not as important as is the 
location of each device in the enclosure, coupled of course with the 
associated temporal filtering which is unique to the position of the 
device within the enclosure. The loudspeaker systems of the invention are 
not limited to home audio systems, but by virtue of design can be applied 
within any reverberant enclosure, regardless of dimensions, to achieve the 
same desired effect: 1) an increase in the sonic depth and width of the 
enclosure, 2) the impact of a live performance, and 3) an increase in the 
perceived "liveness" of the room acoustics. 
The present invention provides unique methods of utilizing spatial and 
temporal signal processing with conventional loudspeaker transduction 
devices to maximize the width and depth of the sonic image in a 
four-dimensional (time being the fourth dimension), reverberant sound 
field, regardless of the spatial dimensions of the sound field. Referring 
to FIG. 2, an embodiment of the invention for immersive observation by a 
binaural auditory system, such as human ears, is provided for use in an 
enclosure. As used throughout, observation refers to the facts that the 
observer may not only listen to the sound but may also feel vibrations 
from the system as part of the complete experience. 
An enclosure 10 can be a room of a residential dwelling, a theater, a 
conference room or any other enclosed environment for presenting sound to 
an audience facing in a predetermined direction. The enclosure 10 includes 
a front wall 12 adjoining, at a first corner 14, a left wall 16 and, at a 
second corner 18, a right wall 20, the left wall 16 and the right wall 20 
extending rearwardly from the front wall. The enclosure can further 
preferably includes a rear wall 22, a floor and a ceiling (not 
illustrated). The enclosure can further include doors, windows and other 
openings (not shown). 
An embodiment of the invention directed to the audio experience for an 
audience facing a predetermined forward direction includes at least one 
central audio loudspeaker 24 placed substantially centrally between the 
left wall 16 and the right wall 20. The central audio loudspeaker 24 has 
an input filtered to range in frequency from substantially 150 Hz to no 
more than 10 kHz. The input to the central audio loudspeaker 24 should be 
limited in frequency to 6 kHz, or even preferably to 3-4 kHz. The central 
audio loudspeaker can be any of a variety of loudspeakers capable of 
performing in the frequency range specified but is preferably selected to 
have an optimal sensitivity and performance in the input range. 
The embodiment for immersive observation further includes a left audio 
loudspeaker 26 placed adjacent the left wall 16 and a right audio 
loudspeaker 28 placed adjacent the right wall 20 of the enclosure 10. The 
left audio loudspeaker 26 and the right audio loudspeaker 28 can be spaced 
from the walls 16, 20 to varying degrees, provided that the loudspeakers 
26, 28 are spaced apart to allow the observer 30 to sit or stand between 
them. While it is preferred that the left audio loudspeaker 26 and the 
right audio loudspeaker 28 be located directly to the sides of the 
observer 30, it is within the scope of the invention that the loudspeakers 
26, 28 may be forward or rearward of these exact positions, but the left 
audio loudspeaker 26 and the right audio loudspeaker 28 are preferably 
located rearward of the central audio loudspeaker 24 relative to the wall 
front 12. Moreover, a plurality of loudspeakers having the same frequency 
parameters as the left audio loudspeaker 26 and the right audio 
loudspeaker 28 can be arranged along the left and right walls 16, 18, 
respectively. 
According to the invention, said left audio loudspeaker 26 and said right 
audio loudspeaker 28 each having an input filtered to range in frequency 
from substantially 900 Hz to at least substantially 12 kHz, whereby the 
left audio loudspeaker 26 and the right audio loudspeaker 28 create a 
maximum width of the acoustic image and produce a stereophonic effect. The 
frequency range of the left audio loudspeaker 26 and the right audio 
loudspeaker 28 can extend to 16 kHz. The left and right audio loudspeakers 
can be any of a variety of loudspeakers capable of performing in the 
frequency range specified but are preferably selected to have an optimal 
sensitivity and performance in the input range. 
In combination with the left audio loudspeaker 26 and the right audio 
loudspeaker 28, the central audio loudspeaker 24 creates a central image 
and greater depth to the sound field. 
The embodiment for immersive observation preferably further comprises at 
least one sub-woofer audio loudspeaker 32 having at least one low pass 
filtered input having a cutoff frequency less than 1000 Hz and preferably 
below 600 Hz. According to the invention, it is desired to limit the 
sub-woofer audio loudspeaker performance to below 600 Hz to avoid 
localization of the low frequency signal while allowing production of the 
overtones approaching 550 Hz that contribute to the realism of the low 
frequency sound. The sub-woofer input can be further limited to below 250 
Hz. 
According to the invention, the sub-woofer audio loudspeaker 32 is coupled 
to dynamics of the enclosure by being placed adjacent a wall of the 
enclosure. The sub-woofer audio loudspeaker 32 is preferably disposed in 
one of the corners 14. The system can include a second sub-woofer audio 
loudspeaker 34, placed in the other corner 18. The sub-woofer loudspeaker 
can be any of a variety of loudspeakers capable of performing in the 
frequency range specified but is preferably selected to have an optimal 
sensitivity and performance in the input range. The sub-woofer audio 
loudspeaker can be driven by an output channel of a separate amplifier 
that combines the two channel input from the audio source. Alternatively, 
the sub-woofer audio loudspeaker can be driven by one of the outputs of a 
multichannel amplifier that processes the two channel input from the audio 
source. 
The preferred embodiment of the immersive sound system further includes a 
high frequency device or transducer 36 with a frequency bandwidth 
extending from approximately 4-6 kHz to the limit of the device, which is 
typically greater than 20 kHz, but at least 15 kHz. The amplifier for the 
high frequency device, be it a part of a multi channel amp or a dedicated 
amplifier for the high frequency device, preferably is equipped to sum the 
two signal input from the audio source to a mono output to the high 
frequency device. 
The high frequency device 36 is preferably mounted at the rear and 
centrally in the ceiling of the enclosure, that is, vertically higher than 
the left and right audio loudspeakers. The high frequency device 36 should 
be placed rearwardly from the front wall 12 no less than the distance the 
left audio loudspeaker 26 and the right audio loudspeaker 28 are placed 
rearwardly from the front wall 12. The high frequency device 36 can be 
provided by any of a variety of transducers capable of providing high 
quality sound in the specified range. 
Referring to FIG. 2a, the audio system for providing driving signals to the 
loudspeakers includes an audio generating source 38 for generating a 
plurality of channels or audio signals and may be a CD player, film 
soundtrack, VCR player or tape deck. The audio source 38 is fed to signal 
processing electronics 40 which can include preamplifiers and cross over 
networks to amplify the signal and use either active or passive crossover 
networks to separate the frequencies but with predetermined overlaps for 
the different loudspeakers. The crossover network can produce two or more 
channels in the frequency range from substantially 900 Hz to 12 Khz for 
the left and right audio loudspeakers. The signal processing electronics 
40 also produces a summing monophonic signal from the two or more channels 
from the high frequency signals above 5 kHz to drive the high frequency 
device. The signal processing electronics further produces a summing 
monophonic signal from the two or more channels from the low frequency 
signals to drive the sub-woofer and the central audio loudspeaker and use 
two or more overlapping frequency bands. 
The signals generated by the signal processing electronics 40 are amplified 
by an amplifier system 42 to drive the transducers or loudspeakers 44 of 
the system. The amplifier system 42 can include a single audio amplifier 
for receiving two or more channel input and producing multiple channel 
output. Alternatively, the central audio loudspeaker 24 can be driven by a 
first audio amplifier and the left audio loudspeaker 26 and the right 
audio loudspeaker 28 can be driven by a second audio amplifier. The 
central audio loudspeaker, the sub-woofer and the high frequency device 
can likewise be driven by separate amplifiers supplied with the 
appropriately filtered and summed-to-mono signals. 
The novel positioning (spatial signal processing) and frequency bandwidth 
(temporal signal processing) of each transduction device illustrated in 
FIG. 2 is critical in the development of a four dimensional sound field 
with a greater perceived sonic width and depth than conventional 
loudspeaker systems and thus an expanded "sweet spot" within the 
enclosure. The electronic signals sent to drivers 24, 32, 34 and 36 are 
preferably all mono, as opposed to stereo. The only stereo signals of the 
preferred embodiment are sent to drivers 26 and 28. The left and right 
stereo signals sent to transducers 26 and 28 are required by the binaural 
auditory system to effectively "locate" or "position" the stimuli audibly. 
In the azimuthal plane, there are two principal mechanisms involved in 
determining the direction from which a sound emanates: 1) interaural time 
difference (ITD) and 2) interaural intensive difference (IID). Interaural 
time difference (ITD) utilizes the time delay between sound entering each 
opposing ear to resolve the direction from which it emanates. This method 
functions best at frequencies below approximately 1667 Hz, assuming the 
width of the head is approximately 0.2 m since the wavelength (.LAMBDA.) 
of sound at 1667 Hz is approximately 0.2 m in air where the speed of sound 
(c) is approximately 340 m/s. However, depending upon the angle of 
incidence, ITD processing can have a limited effect up to approximately 
3000 Hz. Interaural intensive difference (IID) utilizes variation in the 
sound intensity at each ear to resolve the direction from which the sound 
emanates. The head of the observer 30 serves as a baffle, causing incident 
sound waves to reflect and diffract at higher frequencies (greater than 
3000 Hz), resulting in significantly different levels of intensity 
depending upon the angle of incidence. As might be expected, there is a 
bandwidth over which neither works most effectively (approximately 1000 Hz 
to 3000 Hz) since the ITD is too large to accurately determine the 
direction and the IID is too small to determine the direction. Stevens, S. 
S., and E. B. Newman, 1936. "The localization of actual sources of sound," 
American Journal of Psychology, 48, pp. 297-306. 
According to the invention, the central loudspeaker 24 positioned at 
"center stage" can be supplied with a mono signal between 150 Hz and 3000 
Hz, which fills the listening environment with low to mid frequency sound 
waves without deteriorating the stereophonic image created by the left 
audio loudspeaker 26 and the right audio loudspeaker 28. 
This bandwidth of sound is important with respect to the characteristic 
frequency response of the biomechanical transduction which takes place in 
the ear. The frequency response of the ear is generally represented by the 
A-weighting curve as illustrated in FIG. 3. A-weighting is a generally 
accepted method of assigning a weight to a measurement obtained with a 
transduction device such as a microphone that is related to the 
sensitivity of the ear at that frequency. Kinsler, L. E., A. R. Frey, A. 
B. Coppens and J. V. Sanders, 1982. Fundamentals of Acoustics, Third 
Edition, John Wiley & Sons, Inc., Canada, pp. 246-278. Kinsler et al., 
1982). As illustrated in FIG. 3, the peak sensitivity of the ear occurs 
between 2000 Hz and 3000 Hz, and thus the central audio loudspeaker 24 can 
be used to "fill" the "center stage" with sound without deteriorating the 
sonic image because it is centrally located. 
At very low frequencies, below approximately 350 Hz, the ear relies on ITD 
to resolve the location from which sound emanates; however, the wavelength 
of sound so far exceeds the dimension of the head at frequencies below 350 
Hz that low frequency sounds appear omni-directional. Thus, the 
sub-woofers 32 and 34 can be provided with a mono signal and used to 
generate the entire bass response without deteriorating the perception of 
the sonic image. The sub-woofers 32 and 34 are located in the front corner 
of the enclosure to take advantage of spatial signal processing as well. 
Placing the sub-woofers 32 and 34 in the corners provides a mechanism for 
coupling to all of the room modes at very low frequencies and increasing 
the effective sound power in a region where the sensitivity of the ear is 
diminished, as illustrated in FIG. 3. 
To confirm this statement, consider a modal model of room acoustics at low 
frequencies where the modal density is sufficiently low to support such a 
model. The modal model can be derived from the homogeneous wave equation: 
EQU .gradient..sup.2 +k.sup.2 !p(x,t)=0. (1) 
where .gradient..sup.2 is the Laplacian in an appropriate coordinate system 
(i.e., rectangular, cylindrical, etc., depending upon the shape of the 
enclosure respectively), k is the acoustic wavenumber, and p(x,t) is the 
acoustic pressure at the vector field point x. Assuming a series solution 
to the partial differential equation which is separable in space and time: 
##EQU1## 
where p.sub.n (t) is the response in generalized coordinates and 
.psi..sub.n (x) is the n-th acoustic mode shape of the enclosure. It is 
well documented, as in Morse, P. M. and K. U. Ingard, 1986. Theoretical 
Acoustics, Princeton University Press, pp. 576-599; Pierce, A. D., 1989. 
Acoustics, Acoustical Society of America, pp. 284-286; Fahy, F. 1985. 
Sound and Structural Vibration, Academic Press, New York, pp. 241-260 that 
the acoustic mode shapes of a rectangular enclosure can be expressed as 
follows: 
##EQU2## 
where A.sub.n is the modal amplitude, L.sub.x is the dimension of the 
enclosure in the x-direction, n.sub.x is the modal index for the 
x-direction and similarly for the remaining variables. 
The critical observation to be made is that if the radiating surface is 
placed in a corner of the enclosure, regardless of the modal index, each 
cosine term is unity since the spatial position corresponds to a maximum 
of the cosine function. This mathematical result demonstrates that the 
acoustic source can effectively couple uniformly to all acoustic modes of 
the enclosure and excite the modes with uniform phase below the first 
resonance frequency of the enclosure (excluding the rigid-body mode). For 
a typical enclosure with dimensions of 3.5 m by 4 m by 2.4 m, the 
resonance frequency (f.sub.n) of the first acoustic mode can be computed 
from the following expression: 
##EQU3## 
Hence, for the dimensions provided, the first resonance occurs at 
approximately 70 Hz. In addition, the acoustic source can be physically 
placed at some finite distance from the corner to spatially "roll-off" the 
response of the enclosure to the loudspeaker by virtue of spatial signal 
processing since the magnitude of the cosine term diminishes as the 
distance from the surface increases. 
In order to discuss the radiation characteristics of speakers, a simplified 
speaker radiation model will now be presented. For the purpose of this 
discussion an acoustic driver will be modeled as a piston source. The far 
field pressure radiating from a vibrating piston source can be expressed 
as Fahy, F. 1985. Sound and Structural Vibration, Academic Press, New 
York, pp. 241-260: 
##EQU4## 
where p is the farfield pressure, t is time, j is the square root of -1, 
.rho..sub.o is the density of air, c is the speed of sound of air, k is 
the acoustic wavenumber, a is the piston (speaker) diameter, v.sub.n is 
the velocity of the piston, J.sub.1 is the first order Bessel function of 
the first kind, and .theta. is the angle from the normal direction to the 
piston surface. The function in brackets is known as the directional 
factor, H(.theta.), and can be expressed as: 
##EQU5## 
which is equal to one for a given diameter speaker at sufficiently low 
frequencies. A plot of the directional factor, H, as a function of ka sin 
(.theta.) is presented in FIG. 5. Note that the independent variable, ka 
sin (.theta.), is a function of angle and frequency (through the 
wavenumber, k). This frequency and angular dependence give rise to a 
directivity pattern that changes with frequency. Most speakers are 
quasi-omni-directional at low frequency, but at higher frequency have one 
or more distinct lobes which cause the SPL emitted to be strong for 
.theta. equal to zero and decrease rapidly with increasing angle, .theta.. 
IID is used to position the sonic image and reproduce the stereophonic 
sound field at frequencies exceeding 3 kHz, thus the left audio 
loudspeaker 26 and the right audio loudspeaker 28 are preferably 
positioned midway in the enclosure as shown in FIG. 2 to maximize the IID. 
For the proposed "center stage," placing transducers on either side of the 
listener's head will result in the maximum IID, which from a psycho 
acoustic perspective serves to increase the width of the sonic image. 
This concept will be explained in the following few paragraphs. A plot of 
directivity for a typical 2 inch diameter driver that can be used for the 
left audio loudspeaker 26 and the right audio loudspeaker 28 is presented 
in FIG. 6 for frequencies ranging from 1000 Hz to 5000 Hz. Note that up to 
about 2 kHz the driver is omni-directional, but at 4 kHz the response at q 
equal to plus or minus 90 degrees is reduced by approximately 30 dB 
compared to the response at 0 degrees. It can be seen at 5 kHz that the 
response consists of 3 lobes with a nodal cone at approximately plus or 
minus 50 degrees. A three dimensional representation of the 1 kHz 
directivity is shown in FIG. 7. Note that the response is nearly 
omni-directional. In contrast, the directivity for the 5 kHz case is 
presented in the three dimensional plot shown in FIG. 8. As stated 
previously, the majority of the response is concentrated at q less than 50 
degrees. The directivity of the same driver at frequencies ranging from 5 
kHz to 15 kHz, plotted in increments of 2.5 kHz is shown in FIG. 9. For 
this frequency range, the response contains between 3 and 9 lobes with 
sound pressure responses 20 dB or more down for angles over 45 degrees 
away from the normal. 
As previously discussed, the perception of direction is found using the 
method for frequencies above 3 kHz. Moreover, for transient signals the 
perception of direction is most affected by the direction associated with 
the first arrival of a sound (Moore, 1989). Thus, if a sound first arrives 
from a direct path from the speaker to the ear, and also at some time 
later arrives from a reflected path (due to room reverberance), then the 
binaural auditory system perceives the source to be located at a direction 
corresponding to that of the first arrival. For the left audio 
loudspeakers 26 and the right audio loudspeaker 28 with 2 inch diameters, 
the acoustic response is highly directional at frequencies greater than 4 
kHz, and thus the location dictated in FIG. 2 will enhance the perceived 
stereo separation due to the direct path from the driver to the ear, and 
from the increased response of the direct signal (due to the directivity 
of the driver) versus the reflected signals. This concept is illustrated 
diagrammatically in FIG. 10 which depicts the typical directivity patterns 
of the left audio loudspeaker 26 and the right audio loudspeaker 28 
(assumed 2 inch drivers), and the high frequency loudspeaker 36 (assumed 1 
inch driver) at a frequency of 5 kHz. As one moves around within the 
enclosure, a clear left and right stereo image prevails and the audience 
thus benefits from an expanded "sweet spot." One no longer needs to sit in 
a very narrow region to enjoy the full audio experience. In some sense, 
the audience is now "on stage" during the performance as opposed to being 
far removed. This results from the perceived increase in width of the 
room. If one were to consider the left audio loudspeaker 26 and the right 
audio loudspeaker 28 as an array of transducers, it is clear that 
regardless of the audiences' position within the room, a full left and 
right channel is perceived. The bandwidth of stereophonic signals 
delivered to these transducers ranges from 900 Hz to 16 kHz. Except for 
very youthful audiences, the typical audible bandwidth ranges from 
approximately 50 Hz to 15 kHz, so the stereo image is clear within the 
bandwidth supplied to the left audio loudspeaker 26 and the right audio 
loudspeaker 28. Most of the stereophonic imaging techniques currently used 
in industry rely on differences in signal magnitude between channels and 
not time delays as noted in Moore, B. C. J., 1989. An Introduction to the 
Psychology of Hearing, Third Edition, Academic Press, New York, and thus 
are perfectly suited for the arrangement of the left audio loudspeaker 26 
and the right audio loudspeaker 28. 
The directivity of a preferred 8 inch diameter driver which can be used for 
sub-woofer loudspeakers 32 and 34 of FIG. 2 is presented in FIG. 11 for 
frequencies corresponding to 100, 200, and 300 Hz. It can be seen that the 
driver is quite omni-directional up to 300 Hz, and thus its orientation 
with respect to the room is relatively unimportant. 
Finally, to complete the acoustic envelope created for the listener 30, the 
high frequency driver 36 is positioned in the rear of the enclosure or 
centrally overhead. The signal supplied to this device is summed mono and 
ranges from 4-6 kHz to the limit of the device, exceeding 20 kHz by 
design. The psychoacoustic purpose of this device is to create the sonic 
illusion of a more reverberant sound field. High frequency sound is 
typically absorbed by the audience, carpet, seating and other such 
absorptive materials within the enclosure. The high frequency device 36 
creates the illusion of a more live sound field without deteriorating the 
sonic image since the pinna naturally attenuates sounds emanating from the 
rear of the head, Hebrank, J. H. and D. Wright, 1975. "Spectral cues used 
in the localization of sound sources on the median plane," Journal of the 
Acoustical Society of America, 58. Hebrank and Wright, 1975). This 
attenuation is consistent with attenuation which would occur from natural 
reflections of sound waves off boundaries in a reverberant enclosure. 
The four dimensional acoustical audio system according to the invention is 
supported by the principles outlined in psycho acoustics and utilizes both 
spatial and temporal signal processing consistent with the method by which 
humans resolve the direction from which sound emanates to maximize the 
psycho acoustic impact. The transducers are positioned and supplied with 
temporally filtered signals to increase the sonic width and depth of the 
enclosure and produce an acoustic field more consistent with a live 
performance in a reverberant enclosure. 
An alternative embodiment of the invention developed consistent with these 
principles is particularly directed to achieving an immersive experience 
in connection with audiovisual presentations, such as viewing a motion 
picture. Referring to FIG. 4, the arrangement can be constructed in a 
fashion similar to the immersive embodiment discussed above. In 
particular, the system can include a central audio loudspeaker 24, a left 
audio loudspeaker 26, a right audio loudspeaker 28, a sub-woofer audio 
loudspeaker 32 and a high frequency device 36 according to the 
specifications set forth above. Additionally, the alternative embodiment 
can further include a left rear audio loudspeaker 46 and a right rear 
audio loudspeaker 48 having substantially the same frequency input ranges 
as the right audio loudspeaker 28 and the left audio loudspeaker 26. The 
rear left audio loudspeaker 46 and the rear right audio loudspeaker 48 are 
preferably positioned from the front wall 12 rearward of the left audio 
loudspeaker 26 and the right audio loudspeaker 28. The enclosure 10 can 
include a motion picture viewing screen 50 or a video monitor on the front 
wall 12 to orient the observer toward the front wall 12 and provide visual 
information. 
Although details of preferred embodiments of the invention have been 
described herein, it is not intended that the invention is limited to 
these details. Alternative applications and embodiments of the invention 
are possible and will likely become apparent in view of this disclosure. 
Accordingly, the scope of the invention should only be determined by the 
following claims.