[0001] The invention relates to a system for actively reducing sound from a primary noise
source, such as traffic noise, comprising: a loudspeaker connector for connecting
to at least one loudspeaker for generating anti-sound for reducing said noisy sound;
a microphone connector for connecting to at least a first microphone placed adjacent
to said loudspeaker; a control unit coupled to said first microphone connector, for
providing an error signal, based on the output of said first microphone; and a control
unit for outputting a signal to said loudspeaker connector, for controlling said loudspeaker
based on said error signal of said control unit.
[0002] Recently, such systems have been considered as active noise barriers for reduction
of traffic noise. The goal of the active noise barriers is to reduce the noise on
positions distant from the noise barrier, such as on the facades of houses. Microphones
at these positions can be used in order to create error signals to be minimized by
the controller. A problem with this is the vulnerability of connections and microphones
and a therefore expected short economic life. In another configuration, microphones
were placed directly near the active noise barrier. This leads to obvious advantages
due to the absence of sensors at inconvenient positions. This can lead to a possibly
more robust system with respect to fluctuations of the wind. However, it has been
found that the minimization of sound near the loudspeakers is not always appropriate
for reducing the sound pressure at more distant locations.
[0003] The invention has as one of its goals to provide a system that at the one time offers
a compact and robust way of implementing a noise barrier but where the hereabove described
disadvantages are mitigated.
[0004] This is achieved by providing a system according to the features of claim 1.
[0005] Conveniently, it has been found that by deriving a simulated error signal of a virtual
microphone placed in the far-field of the loudspeaker, the noise suppression is much
more successful than conventional active noise barriers and offers reductions even
up to 10 dB or more compared to a sound reduction of a conventional system with adjacently
placed microphones.
[0006] Since the primary and secondary sources may have a very different nature they may
lead to rather different requirements for a transfer function. One big difference,
for example, is that the nearfield sensors are located rather far away from the primary
sources while the nearfield sensors are, by definition, in the nearfield of the secondary
sources. So, although a single transfer function Hyz may have acceptable performance
in some systems, such as active panels, the performance for active noise barriers
can be improved further. Therefore, in a further preferred embodiment, said simulated
error signal of said virtual microphone is a result of a combination of a first transfer
function expressing the far field character of the primary source near said first
microphone and a second transfer function expressing the near field character of the
secondary source near said first microphone.
[0007] The loudspeaker may be arranged to produce anti-sound in a direction away from said
primary noise source. Further, said microphone is preferably placed adjacent in front
of said loudspeaker.
[0008] In a still further preferred embodiment, a second microphone may be placed between
said loudspeaker and said primary source. Said second microphone may be arranged to
provide a second error signal that is used as a feed forward error signal in order
to achieve an expected sound pressure level of the primary source near said loudspeaker.
[0009] Said at least one loudspeaker and said at least first microphone may be part of an
array of loudspeakers and an array of microphones respectively, wherein loudspeakers
and said microphones are placed at a distance less than 5 times an interspacing between
two adjacent loudspeakers. Such an array may serve as an active noise barrier on the
side of a road etc. Still more preferably, said at least said at least one loudspeaker
and said at least one microphone are part of an array of loudspeakers and an array
of microphones respectively, wherein said loudspeakers and said microphones are placed
relative to each other in a range between 10% and 100% of an interspacing between
two adjacent loudspeakers. It has been found that a position of said microphone at
such a general "close" distance of the loudspeaker, is still able to derive a sufficient
error signal in order to be able to calculate a far field sound pressure.
[0010] As a practical embodiment, said microphone may be formed integral with said loudspeaker
in a panel to be placed on the side of a road. To this end, the invention is also
related to a panel for a noise screen to be placed on the side of the road, for actively
reducing noise from a primary source, comprising: a loudspeaker to be directed away
from the primary source
a microphone attached to said panel and placed on a distance away from said panel;
and a system according to any of the preceding aspects.
[0011] Furthermore, said panel may further comprise a second microphone placed opposite
to said first microphone, viewed in a direction away from said panel. Furthermore,
said loudspeaker is placed on top of said panel. In this way the influence of wind
is more or less the same for the primary sources and the secondary sources. This can
lead to a more robust system with respect to fluctuations of the wind.
[0012] In another aspect, the invention is related to a method as described in claims 12
and 13.
[0013] The invention will be further clarified with reference to the figures. In the figures:
Figure 1 shows a basic configuration according to an embodiment of the invention;
Figure 2, 3 and 4 show prior art system diagrams wherein noise control is actively
used;
Figure 5 is a schematic system diagram of the system according to the invention;
Figure 6 shows a sectional diagram of Figure 5 explaining the invention in more detail;
Figure 7 shows a configuration of an active noise control system for reduction of
noise at the position of the farfield sensors by using information derived from nearfield
error sensors showing the region for identification related to the primary noise sources;
Figure 8 shows a comparison of a primary farfield signal with an estimate obtained
from the nearfield sensor signals;
Figure 9 shows a comparison of a secondary transfer function to the farfield with
an estimate obtained from the nearfield sensor signals;
Figure 10 shows a reduction of the broadband sound pressure for three independent
broadband primary noise sources at z= -4 m; the error signal equals the nearheld pressure;
Figure 11 shows a reduction as in Figure 10; the error signal equals a measured farfield
pressure;
Figure 12 shows a reduction as in Figure 10; the error signal equals a virtual farfield
pressure.
[0014] In the description, like or corresponding elements in will be referenced with the
same reference numerals.
[0015] A proposed configuration of the system according to the invention can be found in
Figure 1. In this figure reference sensors 1 (microphones) are present near secondary
sources 2 (loudspeakers), and nearfield error sensors 3 (microphones). In an initial
calibration phase also farfield error sensors 4 (microphones) can be used. One section
(n) of the active noise barrier 5 is considered. In order to assess the stability
and performance of the active noise barrier 5, including the interactions between
different subsystems(n-1, n+1), also the sensors and sources of adjacent sections
should be taken into account.
[0016] One of the aspects of the system is that virtual error signals are derived from the
nearfield error sensors. These virtual error signals should represent the degrees
of freedom in the farfield of the angular sector of interest. In the configuration,
we distinguish the following components: the primary disturbance
d, reference signals
x, actuator control signals
u, nearfield error signals
e_
y, and farfield error signals
e_
z such that

[0017] The action of the control operator W is described by

[0018] Lower case variables denote vectorial quantities that are a function of time, whereas
upper case variables denote matrices that operate on the time-dependent vectors. The
assumption is that
Gux is compensated in the controller, leading to a control structure based on Internal
Model Control. Furthermore, aspects related to decentralization are not taken into
account yet but will be discussed in the remainder.
[0019] For purposes of comparison, some existing systems will be described with reference
to the Figure 2 -4. Figure 2 shows a system using a single transfer function H
yz. Such systems are used for actively controlling noise from panels. However, the performance
for the present application is insufficient, as can be demonstrated in simulations.
Figure 3 shows a system where the basic strategy is to minimize the nearfield pressure,
using a feedforward controller, as in Figure 2. Figure 4 shows a system in which the
nearfield pressure is directly fed back to the controller, where the basic strategy
is also to minimize the nearfield pressure. The latter configuration has been tested
by Japanese researchers. In the present application, reduction of the nearfield pressure
is very different from minimizing the farfield pressure. The simulations described
below will show that the configuration of Figure 5 performs substantially better than
the configurations of Figure 2-4.
[0020] Turning to Figure 5, in a first approach for the reduction of farfield sound pressure
the sound pressure is reduced near the noise barrier. This is the approach that is
used in, for example, a Japanese active noise barrier as illustrated in Figure 4.
A disadvantage of this system is that a reduction of the nearfield pressure not necessarily
leads to a reduction of the farfield pressure. This holds for both feedback configurations
and for feedforward configurations. According to the invention, an improved technique
is based on the estimation of the farfield error signals
e_
z (which are assumed to be unavailable when the system is operational) from the nearfield
error signals
e_
y.
[0021] In a first approach a transfer function H
yz is designed which makes a real-time estimate of
e_z from measured data
e_y. However, it was found that the maximum performance with such a transfer function
is limited by the fact that we are dealing with two sets of sources, the primary sources
and the secondary sources. These two sets of sources may have a very different nature
and may lead to rather different requirements for the transfer function H
yz. One big difference, for example, is that the nearfield sensors are located rather
far away from the primary sources while the nearfield sensors are, by definition,
in the nearfield of the secondary sources. So, although a single transfer function
H
yz may have acceptable performance in some systems, such as active panels, the performance
for active noise barriers can be improved further.
[0022] In Figure 5 and Figure 6 is illustrated, that, for further improving the noise control
in the far field, a new strategy was devised in which the primary sources and the
secondary sources were treated separately. This estimation proceeds as follows. The
actual contributions on
e_
y and
e_
z consist of a part due to the primary signal and a part due to the secondary signal.
As the transfer functions from the primary source and secondary source to both
e_
y and
e_z are quite different, these contributions are taken into account separately. Therefore,
we define G
dy as the transfer function between primary source and nearfield error sensor, G
dz as the transfer function between primary source and farfield error sensor, G
uy as the transfer function between secondary source and nearfield error sensor, and
G
uz as the transfer function between secondary source and farfield error sensor. The
farfield signal
e_z can be expressed as

in which the primary signal
d_z is given by

and the secondary signal s_z is given by

[0023] The estimate
d^z of the signal
d_z is derived from an estimate
d^y of the primary nearfield error signals as follows

where
Hyz {d} is obtained from system identification. The estimate
d^y is obtained by subtracting the secondary signal from the measured signal
e_
y:

where
s^y is an estimate of the secondary signal on the nearfield error sensors, which is given
by

[0024] An estimate of the contribution of the secondary sources on the farfield sensors
is given by

where
Hyz {s} is also obtained from system identification. Finally, the estimate
e^z of the farfield sensor can be obtained from the summation of the estimates of the
primary sources and secondary sources:

[0025] The estimate e^z can be written as

[0026] A block diagram of the above procedure is given in Figure 6. The nearfield error
signal is given by

[0027] Therefore, using

and

the estimate e^z can be written as

[0028] In case of perfect modeling we have G
uy = G^uy. Then the above expression reduces to

[0029] If the secondary sources are close to the diffraction sources and if they radiate
in a similar way then we have H
yz{d} = H
yz{s} = H
yz, leading to

[0030] Stability robustness and performance robustness are governed by the sensitivity of
the control system to changes in the secondary path. The propagation distances involved
in the secondary path are considerably reduced if nearfield sensors are used instead
of the farfield sensors. Therefore, it is expected that the reduced change of the
propagation path will have a positive effect on the stability robustness of the system.
Furthermore, if the primary sources are effectively diffraction sources on top of
a noise barrier and if the secondary sources are close to these diffraction sources
then we obtain a system in which a large part of the possibly varying transmission
path is common to both the primary signals and the secondary signals. This results
in an expected improved performance robustness with respect to changes in wind, temperature,
etc.
[0031] E
q. (17) suggests that minimizing
e_
y would be sufficient to reduce
e^z. This only holds if the primary sources exactly coincide with the secondary sources.
In practical situations it is very difficult to exactly realize the condition
Hyz{d} = Hyz{s} = Hyz. Differences are caused by, for example, the distributed nature of the primary (diffraction)
sources while the secondary sources are concentrated sources. These different behaviors
are especially seen on the nearfield sensors. Also the exact position of the primary
sources may be difficult to determine. Therefore, for a practical implementation a
more general approach is required, such as based on the scheme of Figure 6 where the
estimation of the primary path and the estimation of the secondary path are treated
separately. For the estimation of the primary transfer functions we define a region
with identification sources as shown in Figure 7.
[0032] Furthermore, only those (and precisely those) nearfield pressure distributions should
be controlled that contribute to the farfield, as governed by the transfer functions
Hyz. If
e_y is minimized directly then usually only sound pressure reductions in the nearfield
are obtained. An additional reduction of the order of H
yz can lead to systems that are potentially more robust, have less spillover and require
less actuator effort.
[0033] In order to be able to compute the virtual error signals an initial calibration phase
is assumed in which sources are placed at random positions in the region where the
primary noise sources are to be expected. The estimate of the primary signal on one
of the farfield sensors is shown in Figure 8. The estimate of one of the secondary
source transfer functions to the farfield as estimated from the secondary transfer
functions to the nearfield sensors is shown in Figure 9. In an example 50 calibration
sources have been used, which are indicated as green asterixes in Figure 12. Three
independent primary sources are used, also shown in Figure 12. The identification
procedure is based on the solution of the equations resulting from an assumed multi-input,
multi-output Finite Impulse Response model leading to a block-Toeplitz structure.
In order to improve the stability of the solution, the mean-square value of the coefficients
of the FIR-model are weighted with a normalized coefficient weighting factor β = 10
-2 as compared to the mean-square prediction error of the farfield pressure. This value
of β was obtained by reducing the value of β until the prediction error for primary
signal validation data did not reduce anymore. The latter prediction error was obtained
with validation data from an independent set of primary sources at random positions
in the same region as for the solution of the system of equations.
[0034] Although the transfer functions
Hyz{d} and
Hyz{s} relate the signals from identical nearfield sensors and identical farfield sensors,
they should be obtained from separate calibration procedures, as mentioned previously.
The necessity of this is demonstrated in Table 1, which gives the estimation results
on farfield sensor for different combinations of calibration sources and the two transfer
functions
Hyz{d} and
Hyz{s}. The configuration is identical to that in Fig. 12. It can be seen that the estimator
Hyz{d} only gives accurate results if the calibration sources are placed in the same region
as where the primary sources are to be expected. The same holds for
Hyz{s}, which is only accurate if the calibration sources are taken to be the secondary sources.
[Table 1]
calibration sources |
Hyz{d} estimation
error[dB] |
Hyz{s} estimation
error[dB] |
sources in primary source region |
-14.6 |
10.5 |
secondary sources |
-2.6 |
-36.3 |
[0035] The example of Figure 10 is an active noise barrier in which the error signal equals
the pressure as measured on microphones near the secondary sources. In the present
case this distance, i.e. the difference between the z-coordinate of the 5 secondary
sources and z-coordinate of the 5 error sensors is 0.5 m. In addition, 5 reference
sensors are used, which are positioned, as seen from the secondary sources, 1 m towards
the primary sources. The primary noise sources are three independent broadband noise
sources which are positioned at a distance of 4.5 m from the secondary sources. The
position of the passive noise barrier could be at z=0 but for reasons of simplicity
there is no such noise barrier. The sound pressure in the farfield is evaluated with
5 microphones at z = 20 m. The distance between these microphones is 1.5 m. The sampling
frequency is f_s = 1 kHz, the number of controller coefficients equals 128, the distance
between the secondary sources is 0.3 m, as are the distances between the reference
sensors and the error sensors. All transducers and sources are at a height y = .2
m. Figure 10 shows the configuration and the resulting sound pressure in the x,z-plane.
It can be seen that sound pressure reductions are mainly obtained near the error sensors.
The reduction at the error sensors is 24.2 dB but the reduction in the farfield is
much less, being 1.9 dB.
[0036] If the error sensors are moved to the farfield, viz. to the positions of the evaluation
microphones, then the reductions in the farfield become much higher. The average reduction
of the error signal and consequently the reduction at the evaluation microphones now
becomes 11.3 dB. The results are shown in Figure 11.
[0037] Figure 12 shows the results in case the error signal equals virtual farfield signals,
using the same microphone positions as in Figure 10. The virtual sensor signals are
obtained by processing the nearfield error signals
e_y with a fixed operator which is determined in a calibration phase. The sources as
used in this calibration phase are positioned in the region where the primary noise
sources are to be expected (see Figure 12). It can be seen that the resulting farfield
sound pressure reductions are slightly less than in Figure 11, being 9.0 dB. However,
the farfield sound pressure reductions are considerably higher than obtained with
the nearfield error sensors as used in Figure 10. The reduction of the error signals
is approximately equal in case of minimizing the virtual error signals, being 11.6
dB, and in case of minimizing the true farfield error signals (11.3 dB). The numerical
results for the three cases are collected in Table 2.
[Table 2]
Error signal |
error signal av. reduction [dB] |
farfield pressure av. reduction [dB] |
Figure 10 |
24.2 |
1.9 |
Figure 11 |
11.3 |
11.3 |
Figure 12 |
11.6 |
9.0 |
Robustness of the controllers was evaluated for the following cases: for a different
order of the controller, for a new set of primary sources in the same region as for
the nominal case, for a change of the primary spectrum as compared to the nominal
case, and for a moving source. It was found that a reduction of the number of controller
coefficients leads to less reduction for the nominal case but also to less performance
degradation for changing primary source distributions. Similar trends were found for
the case of 32 controller coefficients. Hence, as compared to the controllers with
128 coefficients, the performance of the systems with 32 and 64 coefficients is less
dependent on the actual primary source distribution and is therefore more stable and
predictable. The influence of a simultaneous change of the positions of the primary
sources and a changing primary spectrum was investigated using a modified primary
spectrum that was obtained by applying a fourth-order bandpass filter with Butterworth
characteristic with lower and upper cutoff frequencies of 0.15 f
s and 0.35 f
s, respectively, to the primary source signals. With the modified primary source positions
and spectrum, a slight improvement was found of the performance on the farfield evaluation
microphones from 10.6 dB to 11.4 dB. Furthermore, the performance for a moving source
was investigated. The radiation characteristics of the moving sources were simulated
by taking into account the doppler shift as well as the changing radiation characteristics
depending on the direction of the movement. Also, simulations were performed in order
to study the performance variations for a change in the height of the primary sources.
The invention is not limited to the hereabove described embodiments but may comprise
variations and modifications thereto while falling under the scope of the annexed
claims.
1. A system for actively reducing sound from a primary noise source, such as traffic
noise, comprising:
- a loudspeaker connector for connecting to at least one loudspeaker for generating
anti-sound for reducing said noisy sound;
- a microphone connector for connecting to at least a first microphone placed adjacent
to said loudspeaker;
- a control unit coupled to said first microphone connector, for providing an error
signal, based on the output of said first microphone; and
- a control unit for outputting a signal to said loudspeaker connector, for controlling
said loudspeaker based on said error signal of said control unit; characterized in that said control unit is provided with a processor adapted to provide said error signal
as a simulated error signal of a virtual microphone placed in the far-field of the
loudspeaker.
2. A system according to claim 1, wherein said simulated error signal of said virtual
microphone is a result of a combination of a first transfer function expressing the
far field character of the primary source near said first microphone and a second
transfer function expressing the near field character of the secondary source near
said first microphone.
3. A system according to claim 1, wherein said loudspeaker is arranged to produce anti-sound
in a direction away from said primary noise source.
4. A system according to claim 1, wherein said microphone is placed adjacent in front
of said loudspeaker.
5. A system according to any of the preceding claims, wherein a second microphone is
placed between said loudspeaker and said primary source.
6. A system according to claim 5, wherein said second microphone is arranged to provide
a reference signal that is used as a feed forward signal in order to achieve an expected
sound pressure level of the primary source near first microphone.
7. A system according to claim 1, wherein said at least one loudspeaker and said at least
first microphone are part of an array of loudspeakers and an array of microphones
respectively, wherein loudspeakers and said microphones are placed at a distance less
than 5 times an interspacing between two adjacent loudspeakers.
8. A system according to claim 1, wherein said at least one loudspeaker and said at least
one microphone are part of an array of loudspeakers and an array of microphones respectively,
wherein said loudspeakers and said microphones are placed relative to each other in
a range between 10% and 100% of an interspacing between two adjacent loudspeakers.
9. A system according to claim 1, wherein said microphone is formed integral with said
loudspeaker in a panel to be placed on the side of a road.
10. A panel for a noise screen to be placed on the side of the road, for actively reducing
noise from a primary source, comprising:
- a loudspeaker to be directed away from the primary source
- a microphone attached to said panel and placed on a distance away from said panel;
and
- a system according to any of the claims 1-10.
11. A panel according to claim 10 further comprising a second microphone placed opposite
to said first microphone, viewed in a direction away from said panel.
12. A panel according to claims 10-11 wherein said loudspeaker is placed on top of said
panel.
13. A method of calibrating a system according to any of the claims 1-9 comprising:
- placing a panel according to claim 10 near the side of the road;
- placing a far field microphone at a distance away from said panel and said road;
- measuring an error signal derived from a first microphone placed adjacent to said
loudspeaker;
- measuring a reference signal derived from the far field microphone;
- calibrating said system by a comparison between said error signal and said reference
signal.
14. A method for actively reducing sound from a primary noise source, such as traffic
noise, comprising:
- deriving an error signal, based on the output of a first microphone placed adjacent
to a loudspeaker; and
- outputting a driving signal to said loudspeaker based on said error signal for generating
anti-sound for reducing said noisy sound;
- wherein said error signal is a simulated error signal of a virtual microphone placed
in the far-field of the loudspeaker.
15. A method according to claim 14, wherein said simulated error signal of said virtual
microphone is obtained by combining a first transfer function expressing the far field
character of the primary source near said first microphone and a second transfer function
expressing the near field character of the secondary source near said first microphone.