RELATED APPLICATION INFORMATION
[0001] This application claims the benefit of U.S. Provisional Application Serial No.
61/800,566, filed on March 15, 2013, hereby incorporated by reference as if set forth
fully herein.
BACKGROUND OF THE INVENTION
Field of the Invention
[0002] The field of the invention pertains to sound reproduction systems and, more
specifically, methods and systems for modifying audio signals from two or more sound
sources creating a sound field within a bounded or semi-bounded listening space to
achieve a desired sound field distribution between and within specified listening
positions.
2. Background of Related Art
[0003] Audio systems are commonplace in households, automobiles and other
environments. Often, audio system components such as amplifiers and speakers are
selected for certain desired characteristics such as high sound fidelity. However, the
audio system components are only one factor affecting sound quality in a particular
environment. Other factors include, among other things, the listening environment itself,
the number and location of speakers, and the position of the listener.
[0004] For example, while many rooms are rectangular, usually one dimension
(length or width) is longer than the other, meaning that sound unfolds differently across
the different dimensions of the room and may reflect at different times off different walls.
This effect is more pronounced with rooms that are not perfectly rectangular in shape. In
addition, the presence of openings or doorways in a room can affect the way in which
sound is reflected or re-directed. Semi-bounded rooms or spaces, such as an outdoor
stage, may have only one or two walls and hence quite asymmetric characteristics for
sound reproduction. Also, the presence of objects or physical features within the room or
listening space, or the existence of surfaces of different types (e.g., windows or hard
surfaces as compared to upholstery or soft surfaces) along the same or different walls,
may also impact the way in which sound unfolds or is reflected within the area.
[0005] In addition to the particular characteristics of the listening area, the listener's
position within the room or listening space also influences the audio experience and
determines the quality and characteristics of the sound experienced by the listener. For
example, it is known that modes may exist within a room or other bounded area at
wavelengths generally comparable to the dimensions of the length or width of the room
or area. These modes may cause constructive or destructive interference that and hence
create acoustic suppression at certain specific frequencies related to the size (or shape) of
the room or other area. These modes are hard to predict for non-rectangular rooms or
areas with odd shapes or physical obstructions. The number and placement of speakers
will also affect what a listener experiences at a particular location in the listening space.
Speakers closer to a listening position will generally be louder than speakers farther
away, and thus, at different listening positions, the aggregate effect of multiple speakers
may differ quite dramatically. Certain speakers, such as dipoles, also have a directional
component, and hence the relative orientation of the listening position as to the speakers
can, in some cases, also affect the listener's experience.
[0006] The above issues may manifest as a detectable difference in power level over
one or more frequencies or frequency bands as between different listening positions
within a prescribed listening area. Where such variability in power level exists, the audio
system may be viewed as inefficient or wasteful, among other things, because maximum
power is experienced in fewer than all listening positions.
[0007] An example of a bounded listening area presenting particular challenges is the
enclosed space within an automobile or other vehicle where the listening positions are
predetermined and suitable locations for the low frequency drivers are restricted. In
addition, the listening positions are restricted to the seating positions provided (usually 4
or 5) and all of these are very asymmetrically placed with respect to the speaker
positions. Space is always at a premium within a car interior and as a result the speakers
are often placed in physically convenient positions that are nevertheless often very poor
from an acoustic point of view, such as the foot wells and the bottom of the front and rear
side doors.
[0008] Some features are provided in automobile audio systems, or other sound
systems, which can partially mitigate the aforementioned problems for one listening
position but at the detriment of another. For example, an occupant can manually adjust
the sound balance to increase the proportional volume to the left or right speakers. Some
automobile audio systems have a "driver mode" button which makes the sound optimal
for the driver. However, because different listening axes exist for left and right occupants
or listeners, an adjustment to the balance that satisfies an occupant (e.g., driver) on one
side of the listening area will usually make the sound worse for the occupant seated on
the other side of the listening area. Moreover, balance adjustment requires manual
adjustment by one of the occupants or listeners, and it is generally desirable to minimize
user intervention. Various types of equalization may also be used, but these are typically
global in nature and hence do not adequately address the different experience at different
listening locations. In addition, a global equalization may improve the sound quality or
experience at one location, but be detrimental to the sound quality or experience at other
locations in the listening space.
[0009] Other techniques propose moving speakers around to find optimal speaker
locations, but those techniques are not effective when speaker locations are fixed.
[0010] Similar asymmetries in sound experience and other related problems may
occur in any other partially or wholly bounded listening space as well, such as in
household rooms, auditoriums, arenas, and other defined listening areas. In some cases
there is flexibility with respect to listening positions, but often the listening positions are
generally fixed. Similarly, it is often the case that speaker locations are fixed and hence
moving speakers is not an option.
[0011] In some cases, as opposed to the goal of having similar sound quality and
level at the listening positions in a particular listening area, it may be desirable to provide
different listening experiences for different occupants or listeners. For example, it may
be desirable to have a quiet zone for one or more occupants, while maintaining good
sound quality for the remaining occupants.
[0012] Accordingly, it would be advantageous to provide an improved sound system
which overcomes one or more of the foregoing problems or shortcomings, and which can
provide improved sound quality or selected sound field variability
SUMMARY
[0013] Embodiments of the invention may include, in one aspect, a technique for
sound allocation within a prescribed listening area, such as an semi-bounded or bounded
listening space. The sound allocation technique may be employed to minimize variance
in frequency response or audio level at different listening positions whilst optionally also
obtaining maximum output capability, or alternatively may be employed to achieve a
desired sound level pattern or sound field variability while optionally obtaining
maximum power output. The sound allocation technique may also be used to achieve
particular zones of generally uniform frequency response (i.e., transfer functions) or
audio level at a specified listening position.
[0014] In a first aspect, an audio system with predefined speaker locations may be
configured to achieve maximum or optimal power output with minimum variance (within
a selected tolerance, for instance) at the listening positions.
[0015] In another separate aspect, an audio system with predefined speaker locations
may be configured to achieve maximum or optimal power output when producing a
desired sound level pattern or sound field variance.
[0016] In yet another separate aspect, an audio system with predefined speaker or
acoustic output source locations may be configured to produce zones of uniform
frequency response or audio level within a prescribed listening space, such as a bounded
or semi-bounded listening area.
[0017] According to one or more embodiments as disclosed herein, an audio system
with predefined acoustic output source (e.g., speaker) locations includes a sound
allocation processor that modifies the signal sent to each speaker so that the vector sum
of the all of the sound sources gives desired response characteristics at each listening
position. The technique is generally applicable to any type of speakers, whether
directional or not, and including monopole or dipole speakers for example.
[0018] Further embodiments, variations and enhancements are also disclosed
herein
BRIEF DESCRIPTION OF THE DRAWINGS
[0019] FIG. 1 is a diagram of an embodiment of a sound allocation system in
accordance with one embodiment as disclosed herein.
[0020] FIG. 2 is a flow diagram illustrating a process for sound allocation in
accordance with one example as described herein.
[0021] FIG. 3 is a top diagram illustrating an example of sound measurement
locations for determining the sound reproduction characteristics of a listening area
relative to different listening locations.
[0022] FIG. 4 illustrates a possible implementation of a sound allocation
processor as may be used in connection with a sound allocation system in accordance
with one or more embodiments as disclosed herein.
[0023] FIG. 5 is a conceptual diagram showing how the aggregate modified
speaker outputs combine at each listening position within a listening area to generate a
modified sound field or frequency response at each listening position, according to one
example.
[0024] FIG. 6 is a diagram illustrating a bounded listening area with a set of
speakers, and various graphs illustrating examples of sound measurements taken at
specified listening positions in the listening area.
[0025] FIG. 7 is a diagram illustrating the same listening area as in FIG. 6, but
with sound allocation as provided according to an example herein, and accompanying
graphs showing modified audio characteristics or frequency responses at each of the same
listening positions after the modified sound signals are played through the various
speakers.
DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS
[0026] According to one or more aspects of embodiments disclosed herein, an audio
system is provided having a plurality of acoustic output sources disposed in or around a
listening area, with a sound allocation processor receiving an audio source signal. The
sound allocation processor may include a plurality of audio modifying elements, one for
each acoustic output source, modifying certain characteristics (e.g., a gain and/or a phase)
of the audio source signal with respect to frequency for each acoustic output source to,
for example, create a uniform sound level over the listening area or within defined zones
within the listening area. The sound allocation processor may, in certain circumstances,
be configured to maximize or optimize the output capability the acoustic output sources
whilst at the same time minimizing the inter-seat response variability and the in-band
response uniformity, within a selected tolerance.
[0027] In various embodiments, each of the audio modifying elements may comprise
one or more custom filters for each acoustic output source, and may optionally further
include a custom gain stage for each acoustic output source. The audio modifying
elements may, for example, include a delay and/or non-minimum phase shift adjustment
that is specifically tailored for each speaker or sound source. In addition, the sound
allocation processor may comprise a global equalization adjustment applied to the audio
source signal for all of the acoustic output sources.
[0028] In a preferred embodiment, the acoustic output sources include low frequency
drive units, and the sound allocation processor is configured to affect primarily low
frequencies of the audio source signal.
[0029] In another separate aspect, a method for sound allocation in an audio system is
provided, comprising receiving an audio source signal and, for each of a plurality of
acoustic output sources, independently modifying a gain and/or a phase of the audio
source signal with respect to frequency to create a substantially uniform sound level or a
desired sound field variability over the listening area or within defined zones within the
listening area. The modified audio source signals are then conveyed to each respective
acoustic output source.
[0030] According to another separate aspect, a method for sound modification in an
audio system having a plurality of acoustic output sources in or around a prescribed
listening area, comprises the steps of characterizing a sound transfer function for each of
the acoustic output sources, and employing an annealing algorithm to identify parameters
providing a specified sound level variance at defined listening positions within the
listening area. The identified parameters may be durably stored in the audio system for
future use, and may later be utilized in the audio system to modify an audio source signal
so as to achieve the specified sound level variance within the listening area.
[0031] In certain embodiments, the identified parameters are applied to adjust a gain
and/or a phase of different spectral components independently for each of the acoustic
output sources. One or more custom filters as well as a custom gain for each acoustic
output source may be used to independently modify the audio source signal for that
acoustic output source. The identified parameters may include a speaker-specific delay
and/or a non-minimum phase shift as applied separately and independently to each
speaker or sound source.
[0032] In a preferred embodiment, as explained in greater detail herein, the annealing
algorithm may involve selecting candidate sound modification parameters for each
acoustic output sources, applying the sound modification parameters to determine a
sound output level at the defined listening positions within the listening area; and
determining a variance in sound output level between the different listening positions. If
the variance in sound output is within a specified tolerance, the candidate sound
modification parameters may be accepted. The sound modification parameters may
include a selected gain associated with each acoustic output source, and/or a selected
phase for different spectral components associated with each acoustic output source. For
example, the selected phase adjustment may involve a frequency-dependent phase pattern
using a component providing a non-minimum phase shift.
[0033] According to certain embodiments, a sound allocation technique is provided
that may maximize or optimize the output capability in an audio system. The sound
allocation technique may also or alternatively, for example, minimize sound variation
among different listening positions, within a selected tolerance, or produce a desired
sound level pattern or sound field variability. The sound allocation technique may also
be used to create "relatively quiet spots" or "relatively quiet zones" and/or produce zones
of uniform frequency response or audio level within a prescribed listening space. These
quiet zones may have a specified sound level reduction as compared to other areas of the
prescribed listening space. Conversely, the sound allocation processor may be used to
create zones of relatively boosted sound or volume level, having a specified sound level
increase as compared to other areas of the listening space.
[0034] The sound allocation techniques and related embodiments described herein
may find particularly advantageous use for listening spaces in which the wavelength at
the maximum frequency of interest subject to processing are greater than 1/1 0th of the
maximum dimension of the listening space. For example, for an automobile interior as
the listening area, it may be desirable to perform the disclosed sound processing on
frequencies in and below the neighborhood of 200 Hertz, which corresponds to
wavelengths in the range of roughly 5-6 feet. In other embodiments, such as for
residential rooms of ordinary size, the sound processing may be performed primarily in
the low frequency range, below some selected threshold such as below 400 Hertz, below
250 Hertz, or 150 Hertz. Conversely, for smaller enclosed spaces, such as a telephone
booth for example, the sound processing may be performed over a larger or higher
frequency range, such as up to 1kHz or 2 kHz for instance.
[0035] In an embodiment in which level allocation is applied by an audio sound
system, a set of four low frequency drive units at predefined locations within an enclosed
listening space are provided with processed audio signals in order to provide near
constant sound levels across frequencies or a desired sound field variability at different
listing positions within the enclosed space.
[0036] Although one or more preferred embodiments are described having four low
frequency drive units, it is to be understood that such a configuration is merely
exemplary. Embodiments of the invention can be practiced with a fewer number (e.g.,
two or three) low frequency drive units, or a greater number, or with other types of sound
sources having characteristics of a monopole, dipole or combination thereof as well of
any arbitrary number so long as the number of speakers is sufficient to create the desired
sound level pattern or sound field variability. The sound allocation is preferably
performed over non-directional frequency bands such as the frequency band below 200
Hertz; thus, the speakers or other sound sources are optimally, but need not be, low
frequency drive units.
[0037] Fig. 1 shows an embodiment of a sound allocation system 100 in accordance
with one aspect of the instant disclosure. In Fig. 1, an audio source 121 provides an
audio signal 122 to an audio sound allocation processor 125 which, as explained in more
detail below, individually modifies the sound for each of a plurality of speakers in a
bounded or enclosed listening area 101. The audio source 121 may include or be derived
from any source of audio content, such as, for example, a conventional radio (including
FM, AM or satellite radio), a CD player, an MP3 player or source, a DVD soundtrack, or
any other source of audio content. The audio source 121 may also include other audio
components, such as amplifiers or pre-amplifiers, equalizers, filters, and the like.
[0038] As further illustrated in Fig. 1, a set of speakers 105A - 105D (which, in this
example, are four in number, although the invention may be practiced with any number
of two or more speakers or other acoustic output sources), which may be monopole or
dipole sources or a combination thereof, are spaced about the bounded or enclosed area
101. While in this example the speakers 105A - 105D are spaced symmetrically around
the bounded area 101, this configuration is not a requirement. An audio input signal 102
is supplied to an audio sound allocation processor 125 which, as described in more detail
hereafter, provides individualized modifications to the phase and/or amplitude of the
audio input signal 102 in order to provide more balanced and even sound at selected
listening positions, or else to provide a sound field of a particular shape or characteristics
over a selected range or band of frequencies. The audio sound allocation processor 125
includes audio modifying elements 131 - 134 which adjust the phase and/or amplitude of
audio input signal 102 respectively for each of speakers 105A - 105D, which are fed by
audio signals 107A - 107D, respectively, output by audio modifying elements 131 - 134.
The nature of audio modifying elements 131 - 134 is discussed by way of illustrative
examples below.
[0039] According to one embodiment that may be implemented in accordance with
the example shown in Fig. 1, the audio sound allocation processor 125 modifies the phase
and/or amplitude of the complex spectra associated with the audio speaker outputs in
order to achieve a substantially uniform audio level at the various listening positions, or a
sound field variability pattern of desired properties, while seeking to maximize total
audio output. In this example, the audio sound allocation processor 125 is configured to
provide a substantially uniform audio level or sound field pattern at six primary listening
positions 140A - 140F, although any number of listening positions may be selected.
[0040] According to another embodiment that may be implemented in accordance
with the example shown in Fig. 1, the audio sound allocation processor 125 modifies the
phase and/or amplitude of the complex spectra associated with the audio speaker outputs
in order to reallocate or readjust the sound levels across different frequencies within a
bounded or semi-bounded listening space 101. In this embodiment, the audio sound
allocation processor 125 may provide different sound experiences at different listening
positions; for example, it may be employed to create a "hole" or "dead zone", i.e., a zone
of relative quiet, in the overall sound field at the location of one or more of the primary
listening positions 140A - 140F. This type of operation can be advantageous, for
example, where one or more of the listeners do not want to hear the audio content.
[0041] In either embodiment, the audio modifications described herein may be
provided on an ongoing basis, or may be applied dynamically for particular situations.
[0042] An illustration of one technique for sound level allocation is illustrated in the
flow diagram 200 of Fig. 2, which may be explained by way of example with reference to
the audio system 100 illustrated in Fig. 1 which, in this case, includes four speakers 105A
- 105D although, as noted earlier, the process may work with any arbitrary number of
speakers of sufficient quantity to suitably effect the listening area. As shown in Fig. 2,
the process 200 begins with a first step of selecting a set of listening positions within an
enclosed or bounded listening space (e.g., area 101 shown in Fig. 1), as represented by
block 205 in Fig. 2. By way of example, the six listening positions 140A - 140F may be
selected. While in this example, six listening positions 140A - 140F are selected, any
number of listening positions may be chosen. Next, sound measurements are taken in
order to characterize the unmodified sound field in the absence of audio processing as
described herein. These sound measurements may involve obtaining a spectral profile of
the speaker output at each measurement location, characterized in the form of a complex
transfer function, using any of the well-known methods for measuring the complex
transfer function between a sound source and receiver. The sound measurements may be
taken for each speaker independently, and may be made at only the listening positions or
else at other locations in the listening area as well, as illustrated in FIG. 3 for example
(measurements taken at locations 310A-C, 315A-C, 320A-C, 325A-C, 330A-C, and
335A-C).
[0043] Once the sound measurements have been taken for each speaker 105A - 105D
in the current example (i.e., with the sound measurement pattern of Fig. 3), the sound
measurements at a given listening position or other sound measurement point are
summed vectorially for each of the sound measurement points, preferably characterized
in the form of a composite transfer function at each sound measurement point
[0044] Next, as illustrated in the following steps in Fig. 2, a sound allocation
algorithm is run on the composite sound profiles 219 in order to generate parameters to
be used with audio electronic equipment in order to create a modified sound field or
sound level pattern following certain desired characteristics. In this example, as an initial
aspect of the sound allocation algorithm (as indicated by step 235), a tolerance value may
be selected (in terms of dB, percent, or other value) by which the sound levels at the
various listening positions or other sound measurement locations may be compared. The
selected tolerance value will affect how many candidate solutions are generated, and is
preferably set so that a meaningful set of candidate solutions is obtained.
[0045] In a next step 240, a search is run in order to identify a candidate set of
solutions to achieve a desired sound level allocation over a given range of frequencies.
The desired sound level pattern may be one, for example, that is as even or uniform as
possible across the different listening positions. Alternatively, the desired sound level
pattern or sound field variability pattern may be one in which certain listening positions
have a drop off in sound level or are substantially quiet. A multivariate algorithm may
employed to select different phase and/or amplitude adjustment values for each speaker
105A - 105D, using the composite transfer functions to determine the predicted output at
each listening position or sound measurement location. If too many candidate solutions
are obtained during the process, then the tolerance value may be tightened in order to
reduce the number of possible solutions.
[0046] A candidate solution may be tested to determine whether the modified sound
level pattern is relatively even across the different listening positions, i.e., the predicted
sound output is within the selected tolerance across the different listening positions
(assuming the goal is to make the sound levels even across the listening area) over the
desired frequency range, as indicated by step 250. The smoothness or uniformity of the
sound field, either globally or within a selected sound zone, may be evaluated by, e.g.,
looking at the standard deviation of the combined sound output at each of the listening
positions or sound measurement points. The process compares the predicted sound
output at each of the different listening positions or sound measurement points with one
another to see if the sound output is within the selected tolerance. If not, then the
candidate solution is discarded (step 251). Otherwise, the candidate solution is tested to
see if the predicted sound output is relatively smooth over the desired frequency range, as
indicated by step 255. If not, then the candidate solution may be discarded (step 25 1).
Alternatively, steps 250 and 255 may be replaced by steps that test whether the candidate
solution is one which provides a sound level pattern or sound field variability of desired
shape, and those that deviate from the desired shape by more than a selected tolerance
may be discarded.
[0047] If no candidate solutions are obtained by the above process, the tolerance may
have been set too tight. In such a case, the tolerance may be increased and another
attempt made to identify candidate solutions.
[0048] To create a "relatively quiet zone" in a particular location within the
prescribed listening area, it is possible to apply an error weighting function to the
measurement points in the quiet zone area in order to reduce the sound output within that
zone. For example, an error weighting function may be applied in the quiet zones so that
the sound produced by the collective sound sources will be suppressed by, for example,
lOdB or 20dB within that region whilst retaining the same frequency response and seat to
seat variation. In terms of running the above candidate solutions, the inverse of the
weighting function, i.e., +10 dB or +20 dB, would be added to the measured values at the
sound measurement points. Then, when the candidate solutions are tested to determine
the predicted sound output, the actual sound output in the "relatively quiet zones" will
actually be less by the value of the error weighting function.
[0049] In one embodiment, a converging algorithm may be employed to identify
candidate solutions by perturbing the phase and/or amplitude individually for each of the
speakers and predicting the sound output at the different sound measurement points, over
the frequencies of interest, by using the measured transfer functions. In particular, an
annealing algorithm may be employed to identify candidate solutions and converge on a
best fit candidate. An annealing algorithm has the benefit of being more likely to avoid
local minima and instead identify a solution that constitutes a global minimum variance.
Annealing algorithms are known generally in the art and are used, for example, in aircraft
for noise reduction.
[0050] As represented now in step 260, the best result from the candidate set of
solutions is identified. This may be carried out as a discrete step or part of the
converging algorithm that is employed to identify candidate solutions. The best
candidate may be one that, through an added global equalization, may be suitable to
achieve the desired pattern of sound levels and characteristics. The sound level pattern or
sound field shape and structure may include desired zones of generally or substantially
uniform frequency response, created in part by utilizing both destructive and constructive
interference in combination. In some cases, the best result from the candidate set of
solutions is one which mitigates losses through destructive interference, evens the load as
much as possible on all of the speakers or other sound sources, and/or reduces peaks and
dips in local zones within the target listening area or globally therein.
[0051] Assuming a suitable solution has been determined, in a next step 270, an
audio modifying element implementation is selected for each speaker. Thus, in the
example of Fig. 1, an implementation would be selected for audio modifying elements
131 - 134 that supply audio signals to speakers 105A - 105D. A variety of different
types of electronic components or filters may be utilized for this purpose. For example,
the required equalization may be implemented by using any combination of finite
response filter (FIR), infinite impulse response (IIR) filters having minimum phase or
non-minimum phase, or other types of filters, in conjunction optionally with a delay
element and/or a gain adjustment applicable to the particular speaker. The audio
modifying elements 131 - 134 each apply the phase and/or amplitude adjustment that had
been determined for the best solution to providing the desired sound field according to
the previously run search algorithm. In certain embodiments, only amplitude adjustment
may be utilized, or only phase adjustment may be utilized.
[0052] In a next step 280, a global equalization characteristic may be selected for the
audio sound allocation processor 125. The global equalization collectively adjusts all of
the signals fed to speakers 105A - 105D so that the actual sound level pattern or sound
field better matches the desired sound pattern or field. Since the sound level at each
listening position is selected by the earlier process to be substantially identical within a
given tolerance (assuming a sound zone or region with generally uniform or even
frequency response or audio level is desired as opposed to one varying in frequency
response or audio level at different listening positions), a global equalization should not
change the fact that the relative sound level should remain approximately the same at
each listening position. The global equalization characteristic may be implemented as a
separate component within the audio sound allocation processor 125.
[0053] Figure 4 illustrates a preferred implementation of a sound allocation system
400 in accordance with one embodiment as disclosed herein. Although the embodiment
of Fig. 4 is similar to Fig. 1 in that it uses four speakers 404A - 404D, any number of two
or more speakers may be used. As illustrated in Fig. 4, the sound allocation system 400
in this example comprises an audio sound allocation processor 425 that is includes or is
coupled to an audio source 421, similar to audio source 121 described previously in
reference to Fig. 1. The audio source 421 provides an audio signal to an equalizer 415,
which applies a global equalization to the audio signal 422 that is ultimately fed, in
modified form, to each of speakers 405A - 405D.
[0054] The output of the equalizer 415 is provided delay elements 431 to 434 which
may apply delay adjustment that is individualized for each speaker 405A - 405D. The
output of the delay stages 431 - 434 are provided to filter stages 441 - 444, respectively,
each of which outputs one of a set of modified audio signals 481 - 484 to speakers 405A
- 405D, respectively. Filter stages 441 - 444 preferably are embodied or include a nonminimum
phase shift adjustment element, although they may generally comprise one or
more low-pass filters, high-pass filters, bandpass filters, bandstop filters, shelf filters,
non-minimum phase components, or other types of filters or elements. Filter stages 441 -
444 may be implemented as FIR or IIR filters, for example, or in other manners.
[0055] For purposes herein, a difference between a minimum phase shift filter and a
non-minimum phase shift filter may be described as follows. A minimize phase shift
filter is generally described by the transfer function:
s ¾
and which does not have zeros in the right half s plane. If, on the other hand, a filter's
transfer function has zeros in the right half s plane, then it would exhibit non-minimum
phase behavior. The modulus of the phase response for a non-minimum phase shift filter
is larger than that for a filter with minimum phase behavior having the same amplitude
response.
[0056] Each speaker 405A - 405D receives an output from one of the filter stages
441 - 444, and thereby receives an audio signal that is modified in terms of phase and/or
gain in order to contribute to a desired sound level pattern or sound field. FIG. 5 is a
conceptual diagram showing how the aggregate modified speaker outputs combine at
each listening position Ml - M4 within the listening area to generate a modified sound
field or frequency response at each listening position, according to one example. For
example, at listening position Ml, the outputs form speakers 405A - 405D combine such
that their aggregate outputs form a combined transfer function at listening position Ml,
according to the vector sum of all of the speaker outputs. A similar effect occurs at
listening positions M2, M3 and M4, but in each case dependent upon the relative audio
level and characteristics of each speaker output as perceived at the particular listening
position.
[0057] Of course, the invention disclosed herein is not limited to the particular
configuration illustrated in Fig. 4, and many other implementations are possible as would
be understood by those skilled in the art.
[0058] In one or more embodiments, the speakers 105A - 105D may be low
frequency drive units, and the adjustments or modifications provided by the sound
allocation processor may effectuate an even bass response across a plurality of listening
positions.
[0059] In some cases, such as where the speakers 105A - 105D are located in an
automobile, the listener can make manual adjustments to the relative volume levels as
amongst the speakers, for example by adjusting a fade control (which adjusts the relative
volume as between front and back speakers) or a balance control (which adjusts the
relative volume as between right and left speakers). Manual adjustments to the relative
speaker volume levels through fade or balance controls may affect the sound allocation
provided by the sound allocation processor. To adjust for the changes in relative volume,
it is possible to provide different parameters for the audio modifying elements 131 - 134
for different levels of fade and/or balance. For example, different filter parameters may
be provided at discrete fade and/or balance levels. Such parameters may be stored, for
instance, in a lookup table within the sound allocation processor 125, and loaded into the
audio modifying elements 131 - 134 in real time as the manual fade and/or balance
adjustments are made. There may be one lookup table for different fade levels and one
lookup table for different balance levels, or else the parameters may be combined into a
single two-dimensional lookup table that uses both the fade and balance levels as
selection inputs.
[0060] An example of a sound allocation process as applied to a particular
listening area may be explained with respect to FIGS. 6 and 7. FIG. 6 shows a top view
of a bounded listening area 601 with designated listening positions 640A-D (also
designated as Ml - M4) and speakers 605A-D at the specified locations near the corners
of the listening area 601 . In this example, the set of speakers 605A-D, which may be
low frequency drivers or subwoofers, are located at various fixed positions in the
listening area 601. All speakers 605A-D are driven equally, i.e., they each receive the
same audio source signal (whether from one amplifier or multiple amplifiers). Notably,
the listening positions 640A-D need not be symmetrical throughout the room, although
they could be; this is a matter of design and implementation choice. Also shown in FIG.
6 are graphs 650, 660, 670 and 680 that show, for each location Ml - M4, a respective
frequency response curve that depicts for each location the difference in response from
the mean or average response for all locations. It can be seen for example that at roughly
45 Hz, there is up to 15dB variation in response between the four locations. This is
generally undesirable from the standpoint of providing a uniform listening experience
regardless of listening position.
[0061] FIG. 7 shows the same arrangement of listening area 601, speaker and
listening positions, but with the speakers 605A-D driven by a sound allocation processor
725 utilizing techniques as previously described herein and, more specifically, that has
been configured to apply gain and/or phase adjustments independently for each speaker
over the frequency range of interest (in this case, below 100 Hertz), after employing a
search algorithm or related technique to arrive at suitable parameters for the sound
allocation processor 725. The sound allocation processor 725 receives an audio signal
from an audio signal source 721, and then uses audio modifying components 731 - 734 to
separately modify the spectral characteristics of the audio source signal individually for
each of the speakers 605A-D, by for example adjusting a gain and/or phase of the audio
source signal individually for each speaker 605A-D. As noted previously, the sound
allocation processor 725 may also apply a global equalization adjustment 715 for all of
the speakers 605A-D. The accompanying graphs 750, 760, 770, 780 correspond to
graphs 650, 660, 670, 680 respectively in FIG. 6, and show that the deviation from the
mean response is greatly reduced by the action of the sound allocation processor 725.
Thus, the operation of the sound allocation processor 725 in accordance with the
principles described herein may act to provide a substantially uniform sound level across
different listening positions, through means of audio processing and without necessarily
requiring a change in the speaker positions.
[0062] According to one or more aspects as disclosed herein, a sound allocation
system comprises a plurality of speakers disposed around a bounded or semi-bounded
listening area, an audio source coupled to a sound allocation processor, said sound
allocation processor comprising individualized sound modification components for each
speaker, wherein the sound modification components adjust the transfer function
individually for each speaker to obtain a desired sound level pattern or sound field
variability within the listening area, with respect to a particular frequency range or band
of interest. In one or more embodiments, the sound modification components are
selected so that the sound level is substantially identical, within a selected tolerance and
over a desired frequency range, at each of a plurality of listening positions. In other
embodiments, the sound modification components are selected so that the sound level
matches a desired non-uniform sound allocation pattern over a desired frequency range
across a plurality of listening positions.
[0063] In one aspect, an audio system is provided having predefined speaker
locations that achieves maximum or optimal power output with minimum detectable
variance (within a given tolerance) at a plurality of listening positions over a desired
frequency range. In another separate aspect, an audio system is provided having
predefined speaker locations that achieves maximum or optimal power output while
producing a desired non-uniform sound level pattern or sound field variability in the
listening area, over a desired frequency range.
[0064] While preferred embodiments of the invention have been described herein,
many variations are possible which remain within the concept and scope of the invention.
Such variations would become clear to one of ordinary skill in the art after inspection of
the specification and the drawings. The invention therefore is not to be restricted except
within the spirit and scope of any appended claims.

What is claimed is:
1. An audio system, comprising:
a plurality of acoustic output sources disposed in or around a bounded or semibounded
listening area; and
a sound allocation processor receiving an audio source signal, said sound
allocation processor including a plurality of audio modifying elements, one for each
acoustic output source, operable to modify the audio source signal for each acoustic
output source by applying at least a non-minimum phase shift adjustment tailored for
each acoustic output source to create a sound field with reduced variability or a desired
sound pattern over a prescribed frequency range within the listening area or defined
zones within the listening area.
2. The audio system of claim 1, wherein each audio modifying element
comprises one or more filters.
3. The audio system of claim 1, wherein each audio modifying element
comprises a customized delay for the acoustic output source.
4. The audio system of claim 1, wherein one or more of the audio modifying
elements comprises a customized gain for the acoustic output source.
5. The audio system of claim 4, further comprising a global equalization
adjustment for the audio source signal.
6. The audio system of claim 2, wherein the sound allocation processor
mitigates power losses caused by destructive interference of sound waves output from the
acoustic output sources.
7. The audio system of claim 1, wherein said acoustic output sources include
low frequency drive units.
8. The audio system of claim 1, wherein the audio modifying elements are
configured to operate primarily over a low frequency range of the audio source signal.
9. The audio system of claim 1, wherein the sound allocation processor
creates at least one relatively quiet zone within the listening area.
10. The audio system of claim 9, wherein the relatively quiet zone has a
specified volume reduction relative to a sound volume in other parts of the listening area.
11. The audio system of claim 1, wherein the sound allocation processor
creates a plurality of relatively quiet zones within the listening area.
12. The audio system of claim 1, wherein the sound allocation processor
creates a zone within the listening area having a specified volume increase relative to a
sound volume in other parts of the listening area.
13. A method for sound allocation in an audio system, comprising:
receiving an audio source signal;
for each of a plurality of acoustic output sources, independently modifying the
audio source signal over a prescribed frequency range by applying at least a nonminimum
phase shift adjustment tailored for each acoustic output source to create a
sound field with reduced variability or desired sound pattern over a prescribed frequency
range within the listening area or defined zones within the listening area; and
conveying the modified audio source signal to each respective acoustic output
source.
14. The method of claim 13, wherein the audio source signal is modified using
one or more filters customized for each acoustic output source.
15. The method of claim 13, wherein the audio source signal is modified using
a delay tailored for each acoustic output source.
16. The method of claim 14, wherein the audio source signal is subject to a
customized gain level for one or more of the acoustic output sources.
17. The method of claim 16, further comprising applying a global equalization
adjustment to the audio source signal for all of the acoustic output sources.
18. The method of claim 13, wherein the independent modification of gain
and/or phase of the audio source signal for each of the acoustic output sources mitigates
power losses caused by destructive interference of sound waves output from the acoustic
output sources.
19. The method of claim 13, wherein said acoustic output sources include low
frequency drive units.
20. The method of claim 13, wherein the modification of gain and/or phase of
the audio source signal is performed primarily over low frequencies of the audio source
signal.
2 1. The method of claim 13, wherein the independent modification of gain
and/or phase of the audio source signal for each of the acoustic output sources results in
creation of at least one relatively quiet zone within the listening area.
22. The method of claim 21, wherein the relatively quiet zone has a specified
volume reduction relative to a sound volume in other parts of the listening area.
23. The method of claim 13, wherein the independent modification of gain
and/or phase of the audio source signal for each of the acoustic output sources results in
creation of a plurality of relatively quiet zones within the listening area.
24. The method of claim 13, wherein the independent modification of gain
and/or phase of the audio source signal for each of the acoustic output sources results in
creation of a zone within the listening area having a specified volume increase relative to
a sound volume in other parts of the listening area.
25. A method for sound modification in an audio system having a plurality of
acoustic output sources in or around a prescribed listening area, comprising:
characterizing a sound transfer function for each of the acoustic output sources;
employing an annealing algorithm to identify parameters providing a specified
sound level variance at defined listening positions within the listening area over a
prescribed frequency range; and
durably storing the identified parameters in the audio system for future use.
26. The method of claim 25, further comprising utilizing the identified
parameters in the audio system to modify an audio source signal and achieve the
specified sound level variance within the listening area.
27. The method of claim 26, wherein the identified parameters are applied to
adjust a gain and/or a phase of the audio source signal independently for each of the
acoustic output sources.
28. The method of claim 27, further comprising using one or more custom
filters for each acoustic output source to independently modify the audio source signal for
that acoustic output source.
29. The method of claim 25, wherein the annealing algorithm includes:
selecting candidate sound modification parameters for each acoustic output
sources;
applying the sound modification parameters to determine a sound output level at
the defined listening positions within the listening area; and
determining a variance in sound output level between the different listening
positions.
30. The method of claim 29, further comprising determining whether the
variance in sound output is within a specified tolerance.
31. The method of claim 29, wherein the sound modification parameters
include a custom gain associated with each acoustic output source.
32. The method of claim 29, wherein the sound modification parameters
include a custom delay and non-minimum phase shift associated with each acoustic
output source.
33. An audio system, comprising:
a plurality of acoustic output sources disposed in or around a listening area; and
a sound allocation processor receiving an audio source signal, said sound
allocation processor including a plurality of audio modifying elements, one for each
acoustic output source, applying at least a non-minimum phase shift of the audio source
signal for each acoustic output source over a prescribed frequency range to create zones
of substantially uniform frequency response within the listening area while mitigating
power losses due to destructive interference of sound waves output from the acoustic
output sources.
34. The audio system of claim 33, wherein each audio modifying element
comprises one or more infinite impulse response (IIR) filters.
35. The audio system of claim 34, wherein each audio modifying element
comprises a delay tailored for the acoustic output source over the prescribed frequency
range.
36. The audio system of claim 34, wherein one or more of the audio
modifying elements comprises a dedicated gain stage tailored to the acoustic output
source.
37. The audio system of claim 36, further comprising a global equalization
adjustment applied to the audio source signal for all of the acoustic output sources.