U.S. patent application number 12/978763 was filed with the patent office on 2011-04-21 for artificial ambiance processing system.
This patent application is currently assigned to Harman International Industries, Incorporated. Invention is credited to Barry A. Blesser.
Application Number | 20110093104 12/978763 |
Document ID | / |
Family ID | 36576586 |
Filed Date | 2011-04-21 |
United States Patent
Application |
20110093104 |
Kind Code |
A1 |
Blesser; Barry A. |
April 21, 2011 |
ARTIFICIAL AMBIANCE PROCESSING SYSTEM
Abstract
An apparatus and method simulates more accurately the natural
statistics of a physical reverberation process. A new filter design
is provided having a comb shaped group delay. Gain minimums at a
plurality of frequencies are combined with a delay line to create a
constant reverberation time independent of frequency while allowing
for temporal spreading. In addition, the connection topology
between the plurality of energy transmission networks is temporally
randomized to facilitate energy distribution within the
reverberation apparatus. Both the temporal and spectral responses
are actively changed on each iteration of the energy recirculation.
By making the response have a high echo density and a lack of
spectral coloration in the decay, the illusion of a natural process
is enhanced.
Inventors: |
Blesser; Barry A.; (Belmont,
MA) |
Assignee: |
Harman International Industries,
Incorporated
Northridge
CA
|
Family ID: |
36576586 |
Appl. No.: |
12/978763 |
Filed: |
December 27, 2010 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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11331370 |
Jan 12, 2006 |
7860590 |
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12978763 |
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09922816 |
Aug 6, 2001 |
7062337 |
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11331370 |
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60226884 |
Aug 23, 2000 |
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Current U.S.
Class: |
700/94 ;
381/63 |
Current CPC
Class: |
H04S 1/002 20130101;
G10H 2210/281 20130101; H04R 5/04 20130101 |
Class at
Publication: |
700/94 ;
381/63 |
International
Class: |
G06F 17/00 20060101
G06F017/00 |
Claims
1. A signal processing system comprising: a first delay module; a
first multiplier, coupled to the first delay module, capable of
creating a real frequency line of poles; and a second multiplier,
coupled to the first delay module, capable of creating a real
frequency of the line of zeros; where for every pole located left
of an imagery y axis in an s-plane representation, there is a
corresponding zero in the right half plane of the s-plane
representation at a same imaginary frequency and the zeros are
closer to the imaginary y axis than the poles.
2. A signal processing system comprising: a first delay module
capable of creating a real frequency line of poles; and a second
delay module capable of creating a real frequency of the line of
zeros; where for every pole locate left of an imaginary y axis of
an s-plane representation, there is a corresponding zero in the
right half plane of the s-plane representation at a same imaginary
frequency and the zeros are closer to the imaginary y axis than the
poles.
3. A filter comprising: a delay module capable of creating a line
of poles and zeros, where for every pole located left of an
imaginary y axis in an s-plane representation, there is a
corresponding zero in the right half plane of the s-plane
representation at a same imaginary frequency; and the zeros are
closer to the imaginary y axis than the poles.
4. A filter comprising: a delay means for delaying a signal and for
creating a line of poles and zeros, where for every pole located
left of an imaginary y axis in an s-plane representation, there is
a corresponding zero in the right half plane of the s-plane
representation at a same imaginary frequency; and the zeros are
closer to the imaginary y axis than the poles.
5. A signal processing system comprising: a first means for
delaying a first signal and for creating a line of poles; a second
means for delaying a second signal and for creating a line of
zeros; and where for every pole located left of an imaginary y axis
in an s-plane representation, there is a corresponding zero in the
right half plane of the s-plane representation at a same imaginary
frequency and where the zeros are closer of the imaginary y axis
than the poles.
6. A signal processing system comprising; a first means for
delaying a signal; a second multiplier means, coupled to the first
means for delaying the signal, for creating a real frequency line
of zeros; and where for every pole located left of an imaginary y
axis in an s-plane representation, there is a corresponding zero in
the right half plane of the s-plane representation at a same
imaginary frequency and where the zeros are closer to the imaginary
y axis than the poles.
7. An energy dispersive transmission network, comprising: (A) a
notchpass filter means capable of receiving an input signal and
generating a first output signals, the notchpass filter comprising
a first delay means for creating a line of poles and zeros, where
for every pole located left of an imaginary y axis in an s-plane
representation, there is a corresponding zero in the right half
plane of the s-plane representation at a same imaginary frequency
and the zeros are closer to the imaginary y axis than the poles;
(B) a second delay means capable of receiving the first output
signal generated by the notchpass filter means and generating a
second output signal; and (C) a multiplier means for scaling the
second output signal according to a gain to generate a third output
signal, an energy decay rate of the third output signal relative to
the input signal being substantially identical at all frequencies.
Description
CROSS-REFERENCE TO RELATED APPLICATIONS
[0001] This application is a continuation of, and claims priority
to, U.S. application Ser. No. 11/331,370, filed on Jan. 12, 2006,
titled ARTIFICIAL AMBIANCE PROCESSING SYSTEM, which is a divisional
of, and claims priority to, U.S. patent application Ser. No.
09/922,816, filed on Aug. 6, 2001, titled ARTIFICIAL AMBIANCE
PROCESSING SYSTEM, which claims priority under 35 U.S.C.
.sctn.119(e) to U.S. Provisional Patent Application No. 60/226,884,
filed Aug. 22, 2000, titled ARTIFICIAL AMBIANCE PROCESSING SYSTEMS,
all applications of which are incorporated by reference in this
application in their entirety.
RELATED APPLICATION
[0002] This application claims priority to U.S. provisional
application No. 60/226,884, Attorney docket number 5764, entitled
"Artificial Ambiance Processing System, by Barry A. Blesser, filed
Aug. 22, 2000.
BACKGROUND OF THE INVENTION
[0003] 1. Field of the Invention
[0004] The invention relates to artificial reverberation and
ambiance systems that create the illusion of real acoustic spaces,
and more specifically to systems for simulating more accurately the
natural statistics of a physical reverberation process.
[0005] 2. Background of the Invention
[0006] In an acoustic space, the sound travels from the source to
the listener via many different paths. The direct path, referred to
as the "dry" signal, corresponds to the signal of an anechoic
space, an outdoor performance, or a close microphone. The indirect
path, referred to as the "wet" signal, bounces off the walls or
other surfaces multiple times and appears at the listener delayed,
attenuated, and spectrally modified. The process continues with
more and more reflections arriving at the listener later and later.
Thus, all reverberation processes can be thought of as a rapidly
increasing series of reflections or echoes. In audio signal
processing, "ambiance" is the general sense of creating an illusion
of a space. "Reverberation" is more specifically the decay process
once the process has become so complex that only a statistical
representation is useful. Ambiance includes reverberation and
spatial location of the listener. Since the earliest days of radio
broadcasting and recorded music, the need for artificial
reverberation and spatial ambiance has been well known. When a
microphone is placed close to a performer to avoid picking up
background noise, very little of the environmental reflections
appear in the signal. Reverberation, created by an artificial
system, is added to the microphone signal to recreate the
perception of the acoustic space.
[0007] Typically, signal processing systems often fail to
accurately duplicate natural acoustics for both practical and
theoretical reasons. The echo pattern or impulse response is
extremely complex and only the early part of the process can be
characterized in detail. Impulse response measurement decays into
the noise level because the amount of source energy is limited to a
fraction of the atmospheric pressure. Averaging cannot be used
because of the lack of thermal equilibrium, which causes minor
changes in the speed of sound. For the same reason, frequency
response measurements never reach a stable steady state. Also, the
computational burden on ray tracing algorithms grows exponentially.
Moreover, the detailed echo pattern is different for each seat in
every space and the physics of a 3-dimensional space is
fundamentally different from the 1-dimension properties of signal
processing. Along a given axis, the speed of sound varies as a
function of the angular orientation, giving a sound wave two extra
degrees of freedom. In a signal processing system, signals travel
at a fixed rate through all delay lines. Fortunately, the physical
complexity of the process does not match the human's ability to
perceive minor differences. As a result, there is a large body of
research that teaches which attributes are perceptible.
[0008] Because of these factors, artificial systems attempt to
create the perceptual illusion of a real acoustic space without
trying to be physical simulations thereof. Large spaces have been
better analyzed in terms of statistics rather than in terms of an
explicit impulse response. When an artificial system has similar
statistics the illusion is good; when those statistics differ, the
illusion is weak. However, the prior art shows a tendency to either
use an artistic or purely mathematical approach. There is very
little prior work that provides a formalism that can relate the
statistical properties of an artificial system to the perceptually
relevant properties of acoustic performance spaces.
[0009] The current generation of artificial reverberation systems
is based on a few digital signal processing primitives including
delay lines, multipliers, and adders. From these primitives, more
complex elements are created. A comb filter, that is implemented
with a delay line having feedback of less than 1, derives its name
from the fact that the frequency response is comb shaped. An
allpass filter, which is a comb filter combined with a feedforward
path, has a flat frequency response, hence the name allpass. An
energy transmission network, which is composed of a large delay
line, holds and transmits the energy between different parts of the
system. These transmission networks are directly analogous to a
linear path through cubic sections of air in a real acoustic space,
which also hold and transmit acoustic energy. Other elements, such
as filters, are added to provide the more subtle attributes such as
high frequency absorption of wall surfaces and the air itself.
[0010] Historically, reverberation systems were based on one of two
basic topologies. One of these topologies uses a multiplicity of
comb filters fed in parallel from a series of allpass filters. The
other topology uses a single large loop composed of a multiplicity
of delays, lowpass filters, and allpass filters. About 15 years
ago, these structures were represented as static mixers using a
matrix notation. Not only can the matrix notation represent all of
the basic reverberation topologies by selecting appropriate
numbers, but the notation can also be used to create other
topologies as long as the numbers are constrained to a set of
mathematical rules. The matrix column vectors should have unity
magnitude and each of these vectors should be orthogonal to all the
others. Any set of numbers that satisfies these rules is the to be
a unitary orthogonal matrix. However, there are an infinity of
matrix numbers that satisfies the mathematical rules. The prior art
does not teach a selection criterion. Constraining the mixer
coefficients to be consistent with these rules is necessary, but
not sufficient, to create a high quality reverberation system
because the rules ignore additional perceptual issues that are
unrelated to the mathematical formalisms. The various necessary
perceptual optimizations often conflict with one another;
optimizing one, while de-optimizing the other.
[0011] Another problem of the prior art is that it is limited to
using a relatively small number of energy transmission networks
because of their high economic burden. It is essentially impossible
to fill these networks uniformly because statistical averaging
requires a large number of such elements. In a real acoustic space,
the energy density becomes uniform after the reverberation process
has continued for a modest amount of time. Artificial systems show
a much weaker tendency to produce a uniform energy distribution
and, hence, often produce energy periodicity in the reverberation,
which is perceptible to the listener.
[0012] Because there are different classes of defects in
reverberators, it is useful to first consider two extreme classes
of sound: broadband pulses and narrow band continuous signals. The
former is typical of a pluck on a guitar or a bang of wood blocks,
while the latter is typical of a flute, organ and other instruments
that have a long steady state. Most music falls between these two
cases. In discussing reverberations systems, professional audio
engineers will often refer to the system behavior when excited
either by an impulse or by a steady state sinewave, representing
the two extreme cases. A defective reverberation system will show
undesirable properties with one or both of these cases. Typically,
an audio engineer first looks for the perceptual smoothness of the
reverberation tail as the dominant quality criterion. Secondly, he
also looks for spectral coloration in the tail. Does the spectrum
of the reverberation have the same spectral content as the
original? The untrained listener does not detect these defects
explicitly but has the sense that the reverberation is not quite
right. Professional sound engineers are, however, very sensitive to
even the most minor defects. Experts in the field can catalog
dozens of critical cases that form the tool chest for evaluation.
The fundamental difficulty in creating the illusion of
reverberation derives from the fact that optimizing one set of
properties often de-optimizes others.
[0013] The prior art has a strong proclivity to describe artificial
reverberation systems as being a complex linear and time-invariant
filter. The linearity property dictates that scaling the input by a
factor will typically scale the output by the same factor. The
time-invariant property dictates that shifting the input in time
will typically shift the output by the same amount. A similar
approach has been rejected when applied to a large acoustic space,
such as a concert hall, because it is very misleading and
unproductive. It is useful only in the degenerate example of a
small and rectangular shaped acoustic space. A concert hall might
easily have more than 100,000 resonances and 100,000 discrete
echoes. Scientists therefore often use a statistical notation that
talks about the frequency response in terms of its average,
standard deviation, slope rates, etc. The reverberation decay is
described in terms of the spectrum of the amplitude envelope and
spectrum of the phase changes. An impulse response can be described
in terms of the spectrum of energy variation within a 1 msec. time
window. The scientific literature shows that there have been some
notable successes in mapping the statistical metrics to the
perceptual properties.
[0014] In contrast, artificial systems are generally very limited
in their complexity. Audio engineers have traditionally not
described the reverberation response in terms of statistics but
have stayed with the deterministic notion of a linear,
time-invariant construct. Some reverberation designers abandon all
structured methods and resort to a purely artistic creation
process. Because artificial systems are generally built out of less
than 20 network modules with only some 50 free parameters, there is
no obvious method to incorporate a statistical approach into the
design process. There are simply not enough degrees of freedom.
Consider, for example, how the resonance density behaves when
excited with a narrow band musical note. A large concert hall can
easily have a density of well over 10 resonance per Hz, whereas an
artificial system might have only 0.3 resonances per Hz. When the
density is extremely high, many excited resonances contribute to
the response resulting in a random envelope and phase response.
With only 2 excited resonances, the envelope will have a
characteristic beat tone at a frequency equal to the difference
between the two resonances, which sounds very unnatural. This
problem has been intuitively understood but has not been
extensively studied. Historically, the solution has involved some
kind of isolated parameter randomization. For example, a delay can
be slowly changed, or a delay output can be panned between random
values. All of these methods have limited utility and some negative
artifacts. Very few methods work within the main recirculation
processing because any artifact, such as needle generation, will
also recirculate. The prior art has not solved these problems. The
prior art generally ignores the subject of statistical
randomization, even though it is critically important.
SUMMARY
[0015] This invention improves artificial reverberation and
ambiance processing systems by providing a mechanism to create the
appropriate random envelope and phase statistics in the output
signal. The inventive system uses two new building blocks: (1) a
notchpass filter as a constituent part of an energy dispersive
transmission network and (2) an energy preserving modulation mixer
in a recirculation topology, both of which can be used separately,
but which reinforce each other when used together. In a typical
embodiment, the modulation mixer creates feedback by randomly
routing the output energy from a plurality of transmission networks
back to their respective inputs. As the energy recirculates, both
the networks and the mixer repeatedly change the audio in a
statistically desirable way thereby avoiding the more typical
structured periodicity and beat tones typical of prior art
systems.
[0016] The notchpass filter may comprise a delay line with both
feedback and feedforward appropriately adjusted to have gain
minimums at the same plurality of frequencies as the group delay
maximums. The notchpass filter is then combined with a second,
longer delay line and an attenuator to form an energy dispersive
transmission network. This network has the dual desirable
properties of both increasing the echo density for pulse signals
and differentially delaying the spectral components for steady
state signals. By appropriate selection of the network parameters,
the reverberation time can be made independent of frequency even
though the filter's gain, by itself, is not spectrally constant.
Having a constant reverberation time may be critical because any
spectral component with a longer decay time would remain audible
while all other components had decayed. It is desirable for the
spectral balance of the reverberation decay to remain constant and
to be the same as the input audio to avoid coloration, a situation
where a few tones completely dominate. Because most reverberation
topologies are based on recirculation, the energy dispersive
transmission network will operate repeatedly on the previous
signals as they circulate around the feedback loop. On each
iteration, the echo density will be dramatically increased and the
spectral components will be further spread in time. Perceptually,
the constantly changing recirculated signals take on a random
quality without any periodicity.
[0017] An energy preserving modulation mixer may also be used to
create n audio output signals, called an audio output vector, from
n audio input signals, called the audio input vector, using a
transformation by which each output signal is composed of a
changing weighted sum of the input signals. The transformation
mapping between the audio input vector and the audio output vector
may be continuously changed such that the amount of a given audio
input signal that appears in a given audio output signal is not
static. The transformation, driven by a set of m randomizer
signals, is constrained so that the energy in the audio output
vector is typically the same as the energy the audio input vector.
The modulation mixer's n audio outputs feeds a set of n energy
dispersive transmission networks and their respective n outputs
feed the mixer's n audio inputs, thus creating an energy
recirculation loop. Because the transformation may be continuously
changing, the energy from a given network is distributed
differently to each of the other networks on each iteration. The
modulation mixer does not increase or decrease the energy in the
audio vectors; hence the reverberation time is not influenced by
the time varying transformation. The m randomizer signals serve to
randomize the interconnections between the plurality of energy
dispersive transmission networks rather just randomizing a single
parameter. The reverberation response can be made to have natural
statistics by the appropriate choice of the m randomizer signals.
With a static mixer, the reverberation would have a characteristic
beat envelope equal to the difference between neighboring
resonances. Such undesirable property is avoided by randomization
of the topology, or equivalently by randomizing the resonances, but
without changing the energy decay process. Randomly moving
resonances emulate the high resonance densities of large acoustic
spaces.
[0018] Since the invention is based on randomization, the total
system energy is the only attribute that remains relevant as the
individual signal's phase and amplitude are continuously modified
during recirculation. Achieving a linear decay process at the
output is equivalent to having the total system energy decay at a
logarithmic rate. The energy preserving modulation mixer does not
change the total energy in the system but does change the
distribution among the plurality of networks. Each energy
transmission network reduces the energy at a constant rate but does
change its signal's phase and amplitude. Both mechanisms have a
random quality to create natural statistics at the final
output.
BRIEF DESCRIPTION OF THE DRAWINGS
[0019] The invention can be better understood with reference to the
following drawings and description. The components in the figures
are not necessarily to scale, emphasis instead being placed upon
illustrating the principles of the invention. Moreover, in the
figures, like reference numerals designate corresponding parts
throughout the different views. In the drawing:
[0020] FIG. 1 is a block diagram illustrating an implementation of
a reverberation system using a notchpass filter and an energy
transmission network;
[0021] FIG. 2 is a block diagram illustrating an implementation of
a notchpass filter where the module for implementing the filter
poles is distinct from the module to implement the filter
zeros;
[0022] FIG. 3 is a block diagram illustrating an alternate
implementation of a notchpass filter with the inner loop producing
the filter poles and an outer feedforward path producing the filter
zeros;
[0023] FIG. 4 is a block diagram illustrating an alternate
implementation of a notchpass filter with an inner feedforward path
to produce the filter zeros and an outer feedback loop to produce
the filter poles;
[0024] FIGS. 5-6 are graphs illustrating the group delay and
amplitude, respectively, as a function of frequency, for the
inventive notchpass filter used in a reverberation topology
system;
[0025] FIG. 7. is a block diagram illustrating a reverberator with
four energy dispersive transmission networks connected with
feedback using a mixer to combine the n outputs with the m inputs
to create the n source signals to feed the four network inputs;
[0026] FIG. 8 is a block diagram illustrating a reverberator with a
plurality of energy dispersive transmission networks in
parallel;
[0027] FIG. 9 is a block diagram illustrating a a reverberator with
a plurality of energy dispersive transmission networks in a single
large loop;
[0028] FIG. 10 is a block diagram illustrating a notchpass filter
modified to allow for a frequency dependent notch depth, that may
be needed to implement a reverberation time as a function of
frequency;
[0029] FIG. 11 is an s-plane representation of the pole-zero
pattern for the inventive notchpass filter designed to work with an
energy transmission network that has a flat frequency response;
[0030] FIG. 12 is an s-plane representation illustrating a
modification of the real part of the line of zeros for various
filters that might be used in the energy transmission network;
[0031] FIG. 13 is an s-plane representation illustrating a
modification of the real part of the line of zeros for various
filters that might be used in the energy transmission network;
[0032] FIG. 14 is a block diagram illustrating a 2.times.2 energy
conserving modulation mixer that can be used by itself or as a
building block to create higher order modulation mixers;
[0033] FIG. 15 is a block diagram illustrating four sets of
2.times.2 energy conserving modulation mixers connected to form a
4.times.4 version, with a generator to create the modulation
signals;
[0034] FIG. 16 is a graph illustrating samples of generator signals
that can be used to drive a modulation mixer to produce random
topology changes;
[0035] FIG. 17 is a block diagram illustrating additional detail of
a 4.times.4 energy preserving modulator having sixteen modulators
and obtaining a coefficient generator from the four modulation
signals;
[0036] FIG. 18 is a graph illustrating the modulation signals with
each signal having a different rate, but only one signal changing
at each instant of time, and each signal stopping when hitting a
threshold corresponding to 45, 135, 225 or 315 degrees;
[0037] FIGS. 19-21 collectively form a block diagram illustrating a
reverberation system using the dispersive energy transmission
networks and the randomizing modulation mixer of the present
invention;
[0038] FIG. 20 is a block diagram illustrating a reverberation
system using the dispersive energy transmission networks and the
randomizing modulation mixer;
[0039] FIG. 21 is a block diagram illustrating a reverberation
system using the dispersive energy transmission networks and the
randomizing modulation mixer; and
[0040] FIG. 22 is a block diagram illustrating a 4.times.4
modulation mixer with an audio output vector supplied to
loudspeakers, which in conjunction with an acoustic space and
microphones, function as an energy transmission network.
DETAILED DESCRIPTION
[0041] This invention provides a system capable of actively
creating the required random temporal and spectral statistics of
natural reverberation for audio signals, especially speech and
music. An impulse and a sinewave are generally used to illustrate
the behavior of the invention because they span the complete range
of interesting audio cases. The former exists only in short instant
of time but has a very broad spectrum; the latter exists for a long
time but has a very narrow spectrum. The invention allows for the
simultaneous optimization of both the impulse and sinewave
response. The prior art puts these two optimizations in conflict.
An unstructured high echo density no longer produces spectral
coloration and vice versa. These conclusions may apply to a vast
number of reverberation topologies and architectures.
[0042] Most reverberation systems contain a multiplicity of long
delay lines with attenuation, called energy transmission networks.
They hold and transmit the acoustic energy. Except for the case of
a very long series chain of energy transmission networks, the
networks are configured to recirculate energy via one or more
feedback paths. In order to achieve a uniform system reverberation
time, all energy transmission networks should have the same ratio
of attenuation, when expressed in dB, to its matching delay time.
System reverberation time, abbreviated RT, is defined as the time
to decay 60 dB. A 100 msec. delay line having an attenuation of 3
dB has the same decay rate as a 200 msec. line with 6 dB of
attenuation. Both decay at a rate corresponding to a 2 second RT.
Since both the delay length and the attenuation are constant,
independent of frequency, the resulting ratio is also constant.
Thus, all frequencies typically have the same RT.
[0043] This invention modifies the standard energy transmission
networks to also produce dispersion. Frequency components are each
delayed by a different amounts (spectral spreading) and a single
impulse is transformed into a plurality of impulses (temporal
spreading). Because these energy transmission networks are embedded
in a feedback topology, the temporal and spectral dispersion will
modify the signal on each iteration. The continuous change achieves
the requirement of preventing coherent patterns in the decay
process. Typically, a comb filter added to a delay line will create
dispersion. However, the comb filter's frequency response has a
grossly non-uniform amplitude and group delay shape. An allpass
filter could be used instead to create a flat frequency response,
but its group delay would still have a comb filter shape. These
structures cannot be used, because they will result in an
unacceptable comb shaped RT as a function of frequency. These
structures produce a high level of spectral coloration. Only a few
frequencies dominate with their much longer RT.
[0044] To achieve the requirement of a spectrally flat RT while
adding dispersion, this invention creates a new structure, called a
notchpass filter. It is built from a comb filter with feedforward,
but has the added property that its frequency response is the
mirror image of its group delay response. The gain is made to have
periodic minimums at the same frequencies as the group delay
maximums. Because both the gain minimums and delay maximums can be
controlled independently, the energy transmission network's RT can
be made to be independent of frequency. The notchpass filter in
combination with a standard energy transmission network becomes a
new building block, an energy dispersive transmission network, that
may replace standard energy transmission networks in reverberation
topologies.
[0045] For example, a notchpass filter may have a 10 msec. delay
line that produces group delay peaks at 100, 200, 300, 400, etc.
Hz, and group delay minimums at 50, 150, 250, 350, etc. Hz. A
typical notchpass filter group delay is shown in FIG. 5. Assume
that the delay of a standard energy transmission network has a
value of 97 msec. and assume that the notchpass filter has a group
delay of 53 msec. and 3 msec. at the maximums and minimums,
respectively. To make all frequencies have the same 2 second RT,
parameters are selected so that the base attenuation is 3 dB and
the notchpass filter gain minimums produce an additional 1.5 dB. A
typical notch pass gain is shown in FIG. 6. A 100 Hz spectral
component experiences a composite delay of 150 msec. with a
composite attenuation of 4.5 dB, while the 50 Hz component
experiences a shorter 100 msec. delay but with a smaller 3 dB of
attenuation. Both spectral components have the same decay rate.
However, assuming that the signal has a flat frequency response in
each network would be wrong. Rather, the RT may be made to have no
frequency dependence. It is necessary only to have a flat spectrum
in the reverberation output. Internal signals are irrelevant. All
spectral components in the output should decay at the same rate.
Historically, prior art systems achieved the flat spectrum by
making both attenuation and gain independent of frequency. However,
this prevents the desirable property of dispersion from being
included within the recirculation. This invention makes them
proportional.
[0046] When the energy dispersive transmission networks are used in
a reverberation system, each spectral component experiences a
different reverberation process because the group delays are
different for each musical overtone. Each overtone, being at a
different frequency, has its own a unique recirculation pattern. On
repeated iterations, each overtone is time shifted relative to the
others by the difference in group delays at the overtone
frequencies. The phase coherence in the decay is completely
removed. Impulse signals have the same property. One each
iteration, the signal is dispersed anew by the notchpass filter.
Furthermore, the echo density at the output increases dramatically
as would happen in a natural process because there are k impulse in
the notchpass filter output for each input impulse. On the next
iteration, each of those k impulse itself produces k more impulses,
etc. Even though there is no active randomization, the invention
allows each part of the reverberation to be very different from
every other part in both its time and spectral structure. RT is
defined as the time for the signal to decay by approximately 60 dB.
Thus, it is often determined by extrapolating the decay rate, which
is the change in signal per unit time with units of dB per
second.
[0047] The previous discussion illustrated the invention under the
assumption that having a spectrally flat decay rate was ideal. In
practice, it is more desirable to have a longer RT at low
frequencies than at high frequencies. A modest change in RT
duplicates real acoustics because the higher frequencies experience
more absorption from air attenuation and surface reflections. The
desired RT frequency response generally has a downward slope with,
for example, values of 3 seconds at 100 Hz, and 1.5 seconds at 5
kHz being typical. In a standard energy transmission network, a
simple filter was added to make the main attenuation have the same
kind of gradual increase as a function of frequency. The invention
may include a matching compensation filter within the notchpass
filter so that the depth of its gain minimums also increase with
frequency. System RT now has the desired shape, a gradual decrease
with increasing frequency, but without any of the comb shape had
the notchpass filter not been modified with a matching compensation
filter.
[0048] There is a direct analogy between an energy transmission
network in an artificial system and a volume of air in a natural
acoustic space since they both hold and transmit acoustic energy.
Because sound travels in 3-dimensions, the energy in a given volume
of air can chaotically move to any of its 6 neighboring regions:
above, below, left, right, front, and back. Conceptually, the
physical space can be thought of as having thousands of individual
cubic energy storage volumes. As the natural reverberation process
continues, the energy gets uniformly distributed to all of
them.
[0049] Many prior art systems have achieved the equivalent complex
connection between a large number of energy transmission networks
by using a 2 or 3 dimensional mesh topology. Such systems remain
laboratory curiosities because of their extremely high compute
cost. Economic limitations generally require that only a few energy
transmission networks be used. It is also well known that good
quality reverberation may require a resonance density of at least 3
per Hz, implying a total delay in all energy transmission networks
of 3 seconds. By having four delay networks of 750 msec. each to
incorporate 3 seconds of delay, it would take a very long time for
the reverberation process to distribute energy uniformly to such
large delay lines.
[0050] This invention achieves the equivalent statistics using only
a modest number of energy transmission networks with only a modest
amount of total delay. The smaller delays allows for rapid filling
while the small number of networks satisfies the economic
requirements. This invention provides a methodology for time
randomization of the topological connections between energy
transmission networks. Its is well known that a static mixer based
on the unitary orthogonal matrix properties can provide a energy
distribution but the statistics only work when there are a large
number of energy transmission networks. The invention solves these
problems by randomly changing the topological connections
represented by the modulation mixer. Without such randomization,
and with a limited amount of total delay, the decay envelope will
have a very disturbing periodic beat. That frequency of the beat
tone will be equal to the difference between neighboring
resonances.
[0051] The need for randomizing in artificial reverberation systems
has been well known but the techniques currently in use are either
insufficient or have undesirable properties. Random delay is very
difficult to implement in a digital signal processing environment
because the interpolation between taps produces a high rapid time
amplitude modulation of high frequency signals. Interpolation
artifacts limit the use of the changing delays to be mostly outside
of the major recirculation loop. This invention provides a new
scheme for producing the desired statistics by changing the nature
of the recirculation rather than by changing isolated delay
values.
[0052] A multiplier can be used for two distinct purposes when one
of the inputs is an audio signal. The second input can be a fixed
number, which produces a linear scaling of the audio signal in the
form of an attenuator or amplifier, or the second input can be
another time changing signal, which creates a modulation process.
While implemented in a similar fashion, they serve fundamentally
different functions. An attenuator is not the same as an amplitude
modulator. In this invention, each of the fixed constants in a
mixer is replaced with a corresponding modulator. That modulator is
driven by a process that creates continuously changing
coefficients. When the process is correctly designed, the mixer
continuously changes the topological connections between the energy
transmission networks without producing any artifacts.
[0053] In one embodiment of the invention, the n.sup.2 fixed
coefficient multipliers of the static mixer with are replaced with
corresponding modulators each having a second time varying input
signal. As a black box, the complete modulation mixer should be
considered as having n audio input signals, called the audio input
vector, n audio output signals, called the audio output vector, and
n.sup.2 coefficient modulation input signals, called the
coefficient modulation vector. The invention contains an additional
network to drive the coefficient modulation vector so that the
modulation signals are constrained to satisfy the energy
conservation property of the audio output vector. When the energy
in the audio output vector is constrained to be the same as the
energy in the audio input vector, the modulation mixer can be used
in the feedback recirculation topology of a reverberator containing
a plurality of energy transmission networks. The modulation signals
can be random, pseudo-random, or periodic but within the
constraint.
[0054] The invention functions to randomize the topology of the
recirculation. At different times, each of the energy transmission
networks feeds a different amount of its energy to the other
networks, including itself. Since these ratios change with time,
the connections are thus randomized. At one instant of time,
network 1 might feed 20% of its energy to itself, 40% to network 2,
10% to network 3, and 30% to network 4. At another instant of time,
network 1 might feed 50% to itself, 5% to network 2, 25% to network
3, and 20% to network 4. As long as the distribution energy sums to
a constant, namely 100%, such a structure will produce smooth
reverberation decay. A large energy burst in one energy
transmission network is relatively quickly distributed to the other
networks.
[0055] The two parts of the invention are closely related in that
the illustrative embodiment contains a modulation mixer, that is
energy preserving. The modulation mixer is also combined with one
or more energy transmission networks, each of which has a uniform
energy decay rate. As a result, the reverberation decays uniformly;
it is not degraded by either of the two processes. Neither the
modulation process of the mixer nor the dispersion of the energy
transmission networks produces spectral coloration of the
reverberation time; but they both break up repeating patterns.
[0056] An implementation of the reverberation apparatus, as shown
in FIG. 1, may be based on the single energy dispersive
transmission network 31, comprising notchpass filter 26 and
standard energy transmission network 29, embedded in a single
feedback loop 32. The output of the energy dispersive transmission
network 31 is coupled back to input adder 20 to form the main
feedback loop 32. Feedback loop 32 recirculates the energy that
enters from input 19 with a gradual decay determine by the
coefficient gRT input of multiplier 28 and by the gain of notchpass
filter 26. In the illustrative embodiment, delay 27 may be
relatively long, ranging from 30 to 300 msec., while delay 23 may
be typically set to a value of about 25% of delay 27. The composite
delay of notchpass filter 26 and energy transmission network 29
determines how long it takes the energy to recirculate.
[0057] Notchpass filter 26 comprises a feedback loop around delay
23, multiplier 24 and adder 21. The amount of dispersion and the
peak group delay is determined by the value of gP input of
multiplier 24. The closer to 1.0, the larger the value. The
feedforward path via multiplier 22 with coefficient gZ allows for
controlling the gain minimum at those frequencies where the group
delay peaks. FIGS. 5 and 6 illustrate the group delay and amplitude
respectively as function of frequency for typical values of gZ and
gP. The composite delay around the main feedback loop 32 comprises
delay 27 and the group delay notchpass 26, while the composite gain
around loop 32 comprises gRT and the gain of notchpass 26.
Generally, reverberation time around a recirculating delay line is
defined by Equation (1) below
RT = - 3 * T log g ( 1 ) ##EQU00001##
where g is the composite gain and T is the composite delay. Given
any two parameters, Equation (1) can be used to compute the third.
The minimum group delay in the notchpass filter is defined by
Equation 2 below
T min = 1 - g 2 1 + g 2 + 2 g * T ap = 1 - g 1 + g * T ap ( 2 )
##EQU00002##
where T.sub.ap is the delay line in the notchpass filter and g is
the average of gP and gZ, as defined in Equation (7). Similarly,
the maximum group delay is defined by
T max = 1 - g 2 1 + g 2 - 2 g * T ap = 1 + g 1 - g * T ap ( 3 )
##EQU00003##
Equations (2) and (3) represent the frequencies where the delay
produces an out of phase condition and an in-phase condition,
respectively. Notice that on a linear scale, Equation (3) dominates
because T.sub.max rapidly goes to infinity as g approaches 1.0. At
the frequency where the group delay is a maximum, the minimum gain
is defined by Equation 4 below
G min = 1 - g Z 1 - g P ( 4 ) ##EQU00004##
where gZ is the gain coefficient for the zeros and gP is the gain
coefficient for the poles. Similarly, the maximum gain is defined
by Equation 5 below
G max = 1 + g Z 1 + g P ( 5 ) ##EQU00005##
at a frequency where the group delay is a minimum. When the energy
storage network, having a gain of gRT, is combined with the
notchpass filter, the composite g is equal to gRT*G.sub.min and
gRT*G.sub.max for the minimum and maximum, respectively. Similarly
the composite delay is T.sub.RT+T.sub.min and T.sub.RT+T.sub.max
for the same two cases where T.sub.RT is the delay of the energy
storage network. Since both the composite gain g and composite
delay T are a function of frequency, the values of gP and gZ are
selected so that RT is remains constant. Effectively, Equation (1)
is solved twice, once at the in-phase frequency and once at the out
of phase frequency.
[0058] The values of delay 27, delay 23, gain gRT, and g are design
choices with a set of reasonable values being 100 msec, 25 msec,
0.7, and 0.5, respectively. The equations are then used to compute
the values of gP and gZ. Equations (2) and (3) evaluate directly
since all of the independent parameters are given. The differences
between gP and gZ is approximately given by Equation (6) below
g Z - g P = ( 1 - g ) ( 1 - 1 g RT * 10 - 3 ( T max + T RT ) RT ) (
6 ) ##EQU00006##
which can be used to determine gP and gZ when equation (6) is added
or subtracted from the definition of g by Equation (7) below
g.sub.Z+g.sub.P=2*g (7)
[0059] A numerically exact solution can be obtained by using any
number of mathematical tools such as Mathcad software, commercially
available from MathSoft, Inc., Cambridge, Mass. 02142, but Equation
(6) is almost perfect because G.sub.max is both very close to 1 and
not very sensitive to changes in the gZ and gP values. An
alternative solution method requires the computation to be done
twice, initially assuming G.sub.max is 1, and then using the actual
G.sub.max based on the previously computed values of gP and gZ.
When these values are used in the system of FIG. 1, energy will
recirculate and decay at a constant rate for all frequencies. Those
frequencies having a longer delay will also have a higher
attenuation.
[0060] The same result will be achieved if notchpass filter 26 is
replaced with any of the alternative implementations shown in FIG.
2-4 or 10. The notchpass filter may be implemented as a linear,
time-invariant filter, which can be described in terms of its poles
and zeros. The location of the poles and zeros in the circular Z
plane for sampled data systems, or in the rectangular S plane for
continuous systems, complete defines the resulting filter. Each of
the topologies may have at least two independent gains and at least
one delay line.
[0061] FIG. 2 illustrates an alternative embodiment of the
notchpass filter of FIG. 1. In this embodiment, filter network 44
which implements the poles is separated from filter network 48 that
implements the zeros. The output from filter network 44 feeds
filter network 48, but the same result is achieved if filter
network 48 feeds filter network 44. Delays 42 and 45 have
approximately the same delay values in order to make the delay
maximums coincide with the same frequency of the gain minimums. The
multiplier 43 is used to set the pole value gP. Multiplier 46 is
used to set the zero value gZ. Adder 41 completes the feedback path
for the poles and adder 47 adds the direct and delayed signals to
create the zeros. Note that adder 25 of FIG. 1 and adder 47 are
both adding the same signals.
[0062] FIGS. 3 and 4 show alternative implementations of the
notchpass filter of FIG. 1. Delays 52 and 63 may have, the same
value. Both topologies have a feedback loop created by multiplier
53 with adder 51 and multiplier 62 with adder 61. Both have a
feedforward path created by multiplier 64 with adder 65 and
multiplier 54 with adder 56. The actual values for the multiplier
coefficients are shown in terms of the basic poles location gP and
zero location gZ. Of the four implementation shown in FIGS. 1-4,
generally the configuration of FIG. 1 is more versatile but
different applications may make the alternative topologies more
efficient.
[0063] In the invention, the notchpass filter parameters are
adjusted to produce a group delay and amplitude characteristics
shown in FIGS. 5 and 6. The group delay may have has a very wide
range, often with a maximum to minimum ratio of as much as 25:1.
When the notchpass filter's group delay is combined with the larger
delay of the energy transmission network, the composite delay
variability is greatly reduced. The notchpass filter's gain
minimums are typically on the order of 0.8 and its maximums may be
very slightly above 1.0, with the exact values dependent on the
parameters. These values are for illustrative purposes only.
[0064] The more general case of feedback using the energy
dispersive transmission networks is shown in FIG. 7 with four such
networks 404, 405, 406, and 407. The n outputs in signal path 403
are connected to mixer 401 that combines the n signals with one or
more of the m inputs in signal path 400 to create the n inputs in
signal path 402. The mixer adds and scales one or more of the
signals in signal path 403 with one or more of the inputs in signal
path 400 for each signal in signal path 402. By changing the nature
of the mixer, the connections illustrated in FIGS. 8 and 9 can be
created. Typically, mixer 401 is of the energy preserving type.
[0065] Applications for the energy dispersive transmission network
are shown in FIGS. 8 and 9. These are real reverberation topologies
that traditionally used a standard energy transmission network.
Elements 76-79 and 92-95 may be implemented with the energy
dispersing transmission network 31 of FIG. 1 or with the modified
notchpass filter design of FIGS. 2-4 and 10. In FIGS. 8 and 9 the
energy dispersive transmission networks may be of identical design,
but with different parameters. The topologies of FIGS. 8 and 9 both
will produce quality reverberation. The four output channels 72,
73, 74, and 75, for the parallel design, and outputs 96, 97, 98,
and 99 for the large loop, can either feed four loudspeakers or can
be mixed to provide a stereophonic or monophonic output. The input
is shown as monophonic signal 70 and 90 but can be expanded to
include provisions for a multiplicity of inputs. These
architectures may have other optional pre-processing elements 71
and 91, including one or more allpass filters, lowpass filters,
etc.
[0066] In order to duplicate the gradual decrease in RT with
increasing frequency, a very gentle lowpass filter replaces the gRT
coefficient of multiplier 28 of FIG. 1. The exact shape of that
lowpass may be controlled by the user, e.g. from the front control
panel. Given the advantages of avoiding a comb shaped response in
RT, a matching frequency dependence may replace the gZ coefficient
of multiplier 22. While somewhat arbitrary, the matching filter is
constrained by the set of equations (1) through (7). Such filters
have no required topology.
[0067] FIG. 10 illustrates an alternative design of a notchpass
filter. Filter 101 replaces gZ and can be implemented as a standard
FIR type (finite impulse response) of any odd order. The order is
the number of taps. If those coefficients in filter 101 are
symmetric, the filter is phase linear, which can be modeled as a
simple delay. Delay 102 may be inserted to match the delay of
filter 101. Both signal paths entering adder 103 may have the same
phase. Hence, the filter is equivalent to a frequency dependent gZ.
In practice, the delay line in filter 101 and delay 102 are merged
with delay 105 in a single element to save computation time and are
shown separately for illustrative purposes.
[0068] The inventive notchpass filters, regardless of the
implementation or topology, can be described using standard signal
processing notations in the form of the s-plane representation for
continuous-time systems, or, in the form of the z-plane
representation for discrete time systems. It is well known that
s-plane and z-plane representations are equivalent. It is also well
known that a linear, time-invariant filter can be converted to one
of these two representations. The notchpass filters described
herein can be equivalently described either in terms of the s-plane
poles and zeros or in terms of a network diagram. In the s-plane
representation, the poles of the notchpass typically appear on a
vertical line and are equally spaced along the vertical or
imaginary axis. The zeroes are also on a line but, unlike the poles
which are in the left half plane, the zeros are in the right half
plane. The line of zeroes is typically closer to the vertical axis
than the line of poles in order to achieve the periodic comb-like
attenuation of the notchpass filter. FIGS. 11 to 13 only show the
upper two quadrants of the s-place because the lower two quadrants
are mirror images of the upper quadrants. Poles and zeros appear as
complex conjugates of each other, except if there is a single pole
and zero at Direct Current.
[0069] FIG. 11 illustrates the location of the poles and zeros for
a notchpass filter in accordance with the present invention. The
pole line 350 contains a sequences of poles with the real part 353
which is determined by the feedback gP and zero line 351 contains a
sequence of zeros with real part 354. Both poles and zeros are
spaced the imaginary frequency distance 352 that is determined by
the size of the delay line. In the illustrative embodiments of the
inventive notchpass filter, there is typically a pole with the same
imaginary frequency as the corresponding zero. As used herein,
especially with reference to the s-plane representation of FIGS.
11-13, the term "line" is not limited to a straight line.
[0070] When the gRT has been replaced by a filter, and when the
notchpass is modified to take this property into account, the poles
or the zeros will typically no longer be on a straight line. FIG.
12 and FIG. 13 illustrate two examples of a bent line of zeros to
achieve optimum ripple free results. The shape of the bend is
determined by the nature of the filter that replaced gRT. Bending
the zero line is equivalent to bending the pole line since the
attenuation is determined by the ratio of the real part of the
zeros to the real part of the poles. Increasing one distance is
equivalent to decreasing the other. In FIG. 12, the zero line
distance from the imaginary axis starts at the value of 361 and
gradually decreases along line 360. In FIG. 13, the zero line
distance starts at the value of 371, gradually increases, and then
decreases along line 370.
[0071] While optimum ripple free reverberation time is desirable,
economic considerations may allow more ripple. Such would be the
case when gRT had been replaced by a filter, but the line of poles
and zeros was left straight as shown in FIG. 11. With an optimum
design, the comb-like attenuation ripples illustrated in FIG. 6 are
made to vary with frequency by applying equations (1) to (7) at
each frequency. Generally, the frequency changes in RT are
sufficiently gentle that these equation only need be applied at a
few select frequencies. With a non-optimizes design, the ripple
depth may be constant as shown in FIG. 6.
[0072] FIG. 14 illustrates a 2.times.2 modulation mixer 128. Mixer
128 can be used by itself, or can be used as a building block to
create higher order mixers. An input audio vector, composed of
audio signals 130 and 131, feeds the four modulators multipliers
139A-C, that are driven by the coefficient vector 137 to create an
audio output vector, composed of output signals 132 and 133. The
energy in the audio output vector is typically the same as the
energy in the audio input vector as long as the coefficients are
created by network 135. Energy conservation comes from the simple
trigonometric rule:
sin.sup.2+cos.sup.2=1
for all angles.
[0073] The modulation process comprises modulation multipliers
139A, 139B, 139C, and 139D in mixer 138. The randomization signal
which appear at input 134 is created by signal generator 129.
Within converter 135 the randomization signal 134 is supplied to
trigonometric functions, that given an input value, generate a
corresponding function value as the output thereof. Such
trigonometric outputs serve as the coefficient vector 137 by which
the audio input vector is modulated by the modulation mixer
136.
[0074] The resulting output audio vector 138 comprises signals 132
and 133. In FIGS. 9-15, circuit paths that cross are not assumed to
be connected.
[0075] FIG. 15 illustrates a 4.times.4 modulation mixer 145 using
four modulation mixers 140-143, each of which includes a
coefficient converter and modulation mixer, similar to converter
135 and modulation mixer 136, respectively of FIG. 14. Since each
basic modulation mixer is energy conserving, pairs of signals can
be passed through any number of the devices. Modulator mixers 140
and 142 each randomize pairs of the input signal while modulation
mixers 141 and 143 randomize their corresponding outputs. The input
audio vector comprises the four input signals 144A, 144B, 144C, and
144D, while the resulting output audio vector comprises signals
146A, 146B, 146C, and 147D. The four randomization signals 147A,
147B, 147C, and 147D are created by generator 144. The approach
shown in FIG. 15 is a two-stage implementation. An infinite variety
of configurations can be created cascading any number of modulation
mixer. A 5.sup.th modulation mixers could be added to the outputs,
or the 4.sup.th mixer removed. The amount and type of randomizing
may be left to the designers choice. Generally, it is desirable to
have as few modulation mixer stagers as possible while still
randomizing most or all the audio input signals. Moreover, by
keeping the number of cascading generations to a small value, there
is no build up of modulation sidebands.
[0076] FIG. 16 illustrates typical signals 150-153 that may be used
to drive the modulation mixers 140-143 of FIG. 15. In this figure,
the input signal is viewed as having phase units, namely that the
signal runs from 0 to 360 degrees and then wraps back to 0. Using a
phase wrapping notation is not a requirement but simplifies the
implementation. In digital signal processing, wrapping can be
easily achieved by making the largest fixed-point digital number
correspond to 360 degrees and the smallest correspond to 0. Since
fixed-point digital numbers wrap, phase wrapping happens
automatically. Signal 150 is shown having a random slope that
changes at random intervals. Such a signal will result in
coefficients that appear to be a sequence of random sinewave
segments at different frequencies. Signal 151 is shown as a pure
linear phase that creates a sinewave coefficient. Signal 152 is
shown with a linear
[0077] FIG. 17 illustrates the implementation of a 4.times.4
modulation mixer 125. An input vector, comprising audio input
signals 110, 111, 112, and 113, feeds the 16 modulators that create
an audio output vector comprising signals 115, 116, 117, and 118. A
modulation vector, comprising modulation inputs 121, 122, 123, and
124, provides the basis from which coefficients are generated by
generator 120. Generator 120 may implement the two mathematical
rules for energy conservation. The magnitude of the vertical
coefficient vector should be 1.0 at all times, and the dot product
of all the vectors with each other, except itself, should be 0.0 at
all times. The design shown in generator 120 is an expansion of the
connection of 2.times.2 modulation mixers shown in FIGS. 14 and 15.
As illustrated in FIG. 17, generation of the coefficient values is
partially achieved with trigonometric functions prior to
scaling.
[0078] FIG. 18 illustrates other modulation signals 160-163. Each
of the four signals shown has a different slope corresponding to a
different frequency. Only one signal changes at each instant of
time and runs until hitting the threshold of 45, 135, 225, or 315
degrees. After the running signal reaches its stopping point, the
next signals start changing. The signal 160 is shown hitting the
threshold of 315 degrees, which then results in signal 161 running
until 215 degree threshold. The process continues at the reference
points of signals 162 and 163. In this example, the stop-start
algorithm is sequential and follows a fixed order. The order shown
is 1, 2, 3, and 4 with repeat. Alternatively, the stop-start
ordering may be random, or the slopes may be random. The major
advantage of the sequential generator is that only one modulation
process is active at a given time. The start-stop approach not only
reduces the computation burden but allows the spectral spreading to
be controlled explicitly. In FIG. 15, the modulation products of
140 add to those of 141 if both modulation inputs are changing at
the same time. In an alternative implementation, the modulation
signals 160-163 be sub-sampled, and can run at half or a quarter of
the rate of the audio signal since they are low frequency
signal.
[0079] The major advantage of the start-stop algorithm is that it
avoids a defect when using periodic or random signals as the source
of the modulation coefficients. A mixer produces the maximum amount
of energy spreading when all of the coefficients have a value of
+/-0.5; each energy transmission network feeds 25% of its energy to
each network. The mixer produces the least amount of mixing when
all of the four diagonal terms are 1.0 and all off-diagonal terms
are 0.0. Each network feeds 100% of its energy to itself and none
to the others. Because of the randomization, the mixer will
occasionally exist in one state and occasionally in the other. A
given musical note may be perceived differently in these two
extreme cases. The start-stop algorithm solves this problem because
the four thresholds all correspond to coefficients with a magnitude
of 0.5. Only the running signal is not at one of these
thresholds.
[0080] The nature of the phase randomization signals is critical to
the operation of the invention. If they are changing too slowly,
there will be no perceived randomization of the reverberation
topology. If they are changing too rapidly, there will undesirable
modulation artifacts introduced, i.e. new spectral side-bands are
created. The higher the modulation frequency, the more spectral
spreading. The amplitude and phase of natural reverberation also
has also has spectrum spreading. The randomization signals should
be selected to match the spreading that would appear in natural
reverberation, which is a function of the RT. For typical
reverberation times, the spreading corresponds to a spectrum
between 0.3 to 3 Hz. Hence, the randomizer should be adjusted to
produce a similar amount of spectral spreading. Phase randomization
signals may generally be in the range from 0.3 to 3.0 Hz, to make
the reverberation tail have an unstructured and non-periodic
personality.
[0081] FIGS. 19, 20 and 21 show an alternative embodiment
containing both the energy dispersive transmission networks,
described previously with reference to FIGS. 1-10, and the random
modulation mixer, described previously with reference to FIGS.
14-18. Stereo audio input signals are fed into lines 248 and 249
where they are diffused by allpass networks 242 and 243,
respectively, each of which has two allpass filter diffusors. The
outputs of networks 242 and 243, respectively, feed delay lines 244
and 245 to allow for unreverberated delayed signals to be fed
directly to the output. Lines L, M, N, O, P and Q appear directly
in the output summations 251 and 255. This kind of preprocessing is
typical of reverberation system, representing a simulation of the
individual early reflections. The choice of preprocessing is based
on the type of acoustic space and the location of the listener. In
the early reflections, the statistics process has not yet begun.
The randomization modulation mixer 241 is part of the main
recirculation loop that includes four sections of processing. The
four sections comprise the energy dispersive transmission networks
200, 210, 220, and 230, as well as filtering sections 201, 211,
221, and 231. Networks 200, 210, 220, and 230 may be implemented
with the energy dispersing transmission network 31 of FIG. 1 or
with the modified notchpass filter design of FIGS. 2-4 and 10.
Filtering sections 201, 211, 221, and 231 may implement a frequency
dependent reverberation time by changing the gRT for high and low
frequencies. The feedback path 202, 212, 222, and 232 around a unit
delay of a respective filter section implement a 1.sup.st order
lowpass filter that serves to reduce the feedback of high
frequencies. The feedback path 204, 214, 224, and 234 around
another unit delay of a respective filter section create a lowpass
filter at very low frequencies, typically 200 Hz. By adjusting
gains 203, 213, 223, and 233, the amount of low frequency RT can be
adjusted separately from the amount of high frequency RT. Each of
the main delay lines 200, 210, 220 and 230 in the energy dispersive
transmission networks is shown with four taps, each feeding the
output summation networks 251 and 255. These summation networks
represent the fully reverberated signals from the recirculation
networks. Finally, the ratio of reverberation to dry input signal
is adjustable by mixers 252 and 256 to form the stereo output
signals 253 and 257.
[0082] Any of the signal processing circuits and, including the
inventive notchpass filter, energy dispersive transmission
networks, and modulation mixer shown in FIGS. 1-22 may be
implemented using a commercially available Digital Signal Processor
(DSP) integrated circuit, such as the Motorola 56004 chip from
Motorola, Inc, Schlumberger, Ill., The published programming
instructions and library routines for such DSP may be used to
construct algorithms which function according to the illustrated
signal processing circuits, such algorithms being within the scope
of those skilled in the arts in light of this invention. Other
products with in the Motorola 56000 family of products may be
similarly used. In addition, DSP product from other manufactures,
such as the SHARC Series, commercially available from Analog
Devices, Norwood, Mass., the TMS 320 Series, commercially available
from Texas Instruments, Dallas, Tex., may be similarly used, or any
other available DSP product may be used to implement this
invention.
[0083] An application will often have additional processing
components inserted at various point, e.g. there may be components
of the energy dispersive transmission network inter-mingled with
other functional blocks. In some cases, the functionality is
implicit. For example, the main attenuation can be embedded within
the notchpass filter without being explicitly represented. These
variations do not change the basic invention but are designers
implementation preferences.
[0084] The actual parameters in the embodiment are a few strong
function of the application. Is the system to simulate a symphony
hall, a bathroom, a train station, a small chamber or an opera
hall? Is the listener sitting close to the stage or in the 3.sup.rd
balcony? Is the music classical, popular or hard rock? The number
of questions is so large, and each of the answers determines the
optimum settings for the network parameters. Furthermore, economic
considerations determine the amount of signal processing that can
be used. The design shown in FIG. 20 has four outputs of delay taps
on each of the four delay sections. Should there be more processing
resources, more taps may be added. With less resources, three taps
might be used instead of four.
[0085] In the preferred embodiment, additional optimization can be
performed to achieve better performance. Smoothness in the
reverberation decay can be improved if the energy in each 1 msec.
interval is essentially the same as the neighboring intervals. To
use this approach, the proposed design is simulated in a computer
with starting values for all parameters. An impulse is entered into
the input and the complete output is analyzed as a sequence of 1
msec. windows. This represents the average energy or amplitude in
the output envelop. A metric such as the peak variation in envelop
or the power in the envelop is applied to the output to create a
figure of merit. The process is repeated a large number of times
with each iteration using slightly different design parameters.
This is known as a Monte Carlo simulation.
[0086] A simple example illustrates the process. Consider having 4
parameters that need to be optimized and that each parameter can
have 3 values. The middle value is the starting reference and each
of the other values represents an incremental increase or decrease
in that value, for example, 1% change. Since each of the 4
parameters can each have one of three possible values, there are a
total of 3.sup.4, or 81 cases. After the figure of merit has been
computed for each, the optimum is selected as the new starting
reference and the process repeats. One of those 81 cases will be
more optimum than all the others. Because computers are so
inexpensive, a very large system can be so optimized by long
computer runs. There are many other equivalent numerical techniques
that can be used. The exact technique for finding the optimum is
not important. It is important that the reverberation decay have
minimum energy variations in the envelop over a 1 msec. averaging
window. The size of the window is not critical since it is only an
approximation to human hearing.
[0087] FIG. 22 illustrates a 4.times.4 modulation mixer 305 with
audio output vector 302 feeding four loudspeakers 310, 311, 312 and
313 located in an acoustic space 304, such as concert hall. The
four microphones 320, 321, 322, and 323 send their signals back to
modulation mixer 305 as the audio input vector 301. The mixer 305
is driven by four randomizer signals 330, 331, 332, and 333 in the
modulation vector 300. The audio source, not shown, may be located
inside of acoustic space 304. Alternatively, a prerecorded audio
signal can be supplied to the loudspeakers. In this embodiment, the
loudspeakers, acoustic space and microphones function as an energy
transmission network, as an alternative those purely electronic
implementations described herein.
[0088] A software implementation of the above-described embodiments
may comprise a series of computer instructions either fixed on a
tangible medium, such as a computer readable media, e.g. diskette,
CD-ROM, ROM, or fixed disk, or transmittable to a computer system,
via a modem or other interface device, such as communications
adapter connected to the network over a medium. Such medium can be
either a tangible medium, including but not limited to optical or
analog communications lines, or may be implemented with wireless
techniques, including but not limited to microwave, infrared or
other transmission techniques. The series of computer instructions
embodies all or part of the functionality previously described
herein with respect to the invention. Those skilled in the art will
appreciate that such computer instructions can be written in a
number of programming languages for use with many computer
architectures or operating systems. Further, such instructions may
be stored using any memory technology, present or future,
including, but not limited to, semiconductor, magnetic, optical or
other memory devices, or transmitted using any communications
technology, present or future, including but not limited to
optical, infrared, microwave, or other transmission technologies.
It is contemplated that such a computer program product may be
distributed as a removable media with accompanying printed or
electronic documentation, e.g., shrink wrapped software, preloaded
with a computer system, e.g., on system ROM or fixed disk, or
distributed from a server or electronic bulletin board over a
network, e.g., the Internet or World Wide Web.
[0089] Although various exemplary embodiments of the invention have
been disclosed, it will be apparent to those skilled in the art
that various changes and modifications can be made which will
achieve some of the advantages of the invention without departing
from the spirit and scope of the invention. It will be obvious to
those reasonably skilled in the art that other components
performing the same functions may be suitably substituted. Further,
the methods of the invention may be achieved in either all software
implementations, using the appropriate processor instructions, or
in hybrid implementations that utilize a combination of hardware
logic and software logic to achieve the same results. Such
modifications to the inventive concept are intended to be covered
by the appended claims.
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