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JPH0646499

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DESCRIPTION JPH0646499
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an
audio reproduction apparatus, and more particularly to a sound field correction apparatus
suitable for use in a narrow space such as a car.
[0002]
2. Description of the Related Art Conventionally, when playing music in a car, correction of
frequency characteristics has been performed subjectively by a listener with a graphic equalizer.
Also, in recent years, a device that measures frequency characteristics using a microphone and
corrects it with a parametric equalizer has been developed. A block diagram of an example of this
device is shown in FIG. In FIG. 10, 1 is a signal source such as a CD player, 2 'is an R channel
equalizer, 3' is an L channel equalizer, 4 is an R channel amplifier, 5 is an L channel amplifier, 6
is an R channel speaker, 7 is a speaker for L channel, 8 is a microphone for measurement, 9 is a
frequency characteristic measuring device, 10 is a CPU, 11 is a storage device (memory), and 12
is a vehicle compartment.
[0003]
The apparatus shown in FIG. 10 outputs a dedicated measurement signal (for example, white
noise) from the speakers 6 and 7, which is collected by the measurement microphone 8 installed
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near the listener's head, and the frequency characteristic measuring device 9 The frequency
characteristic is analyzed by the above, and the CPU 10 controls the equalizers 2, ', 3' according
to the result. This operation is performed for each seat, and the respective coefficients are stored
in the memory 11 to be in the seat mode, and the coefficients are called and used so as to obtain
the optimum characteristics for the seat desired by the user.
[0004]
In addition, correction of the arrival time of sound from the left and right speakers 6, 7 to the
listener is performed by delaying the signal with a delay circuit, and the delay time is measured
by measuring the distance between the left and right speakers 6, 7 and the listener. Was
calculated. FIG. 11 is an explanatory view of the calculation example, and in FIG. 11, x is the
distance between the R channel speaker 6 and the listening position (m: meters), y is the L
channel speaker 7 and the listening position (measuring microphone 8 And a delay time (s;
seconds) for correcting dt, dt = (y−x) / 340. However, the sound velocity is set to 340 m / s.
[0005]
Then, as in the case of the above-described frequency characteristic correction, as the seat mode,
the delay time can be stored in the memory 11, and the user can call and use it as needed. FIG.
12 is a block diagram of this arrival time correction system, wherein 1 is a signal source such as
a CD player, 2 ′ ′ is an R channel delay circuit, 3 ′ ′ is an L channel delay circuit, 4 is an R
channel amplifier, 5 Is an L channel amplifier, 6 is an R channel speaker, and 7 is an L channel
speaker.
[0006]
However, the frequency characteristic automatic correction apparatus shown in FIG. 10 and the
arrival time difference correction apparatus shown in FIG. 12 have the following problems. In the
arrival time difference correction device of FIG. 12, it is necessary to actually measure the
distance for calculating the delay time, and it takes a long time to actually measure the distance,
and the accuracy is not high. In the frequency characteristic automatic correction device shown
in FIG. 10, the characteristic of the correction is disturbed when noise enters from the outside, so
that automatic correction can not be performed under the noise. In the frequency characteristic
automatic correction device of FIG. 10, a computer (microcomputer) must be used to calculate
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the coefficients of the equalizer (filter), and the coefficients of the filter must be held. The point
where the burden on the listener is large because the measurement signal (white noise etc.) is
emitted from the speaker.
[0007]
The present invention has been made in view of the above problems, and it is an object of the
present invention to provide a sound field correction device capable of creating better listening
conditions for music in a vehicle compartment as compared to the prior art.
[0008]
SUMMARY OF THE INVENTION In order to achieve the above object, a sound field correction
apparatus according to the present invention reproduces an acoustic signal from an acoustic
source with respect to a reproduction sound listening position in a predetermined sound field
reproduction space. A reproduction output means for reproducing and outputting as a signal, an
acoustic input means for obtaining a reproduction acoustic signal located in the vicinity of a
reproduction sound listening position, and an acoustic signal from an acoustic source are
inputted, and reproduction is performed based on the taken reproduction acoustic signal. Output
means and adaptive filter means for adaptively controlling at least one of frequency
characteristics, arrival time difference and impulse response of the reproduced sound signal
shifted according to the sound field reproduction space according to the reproduction sound
listening position It is characterized by
[0009]
In the sound field correction apparatus described above, the adaptive filter means examines a
filter coefficient corresponding to an arrival time at which the reproduction acoustic signal
reaches the position of the listener, and when the coefficient is a predetermined level, it
corresponds to the level Adaptive signal arrival time test means for setting arrival time difference
by delay time, delay processing of acoustic signal from acoustic source by delay time, and
capture reproduced acoustic signal by acoustic input means to estimate impulse response of
sound field reproduction space Adaptive impulse response estimation processing means to be
set, and adaptive inverse filter means for setting inverse filter characteristics with a
predetermined measurement signal by impulse response and obtaining frequency characteristics
by adaptive control processing, at the position of the listener Automatically adjust and control
the frequency characteristic, the reproduction sound arrival time, and the impulse response
characteristic.
[0010]
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According to the above construction, the sound field correction apparatus of the present
invention causes the reproduction output means to reproduce and output the acoustic signal
from the acoustic source as a reproduction acoustic signal to the reproduction sound listening
position in the predetermined sound field reproduction space.
Further, the sound input means is located near the reproduction sound listening position, and the
reproduction sound signal is taken in, and the adaptive filter means inputs the sound signal from
the sound source and the reproduction output means based on the taken reproduction sound
signal. And adaptively control at least one of the frequency characteristic, the arrival time
difference, and the impulse response of the reproduction acoustic signal shifted according to the
sound field reproduction space according to the reproduction sound listening position.
[0011]
The adaptive filter means uses the adaptive filter means to pick up the signal by the sound input
means installed near the head of the listener while reproducing normal music one channel at a
time by the adaptive signal arrival time test means. The difference in arrival time is measured,
the sound image is localized in front of the listener, and the impulse of the system (vehicle
interior) by the adaptive filter means even when the extraneous noise is large when traveling by
the external noise cancellation means (adaptive impulse response estimation means) The
response is estimated and correction processing is performed, and by the adaptive inverse filter
function, the disturbed frequency characteristic can be corrected by the adaptive filter means.
[0012]
1 is a block diagram showing an example of the configuration of a sound field correction
apparatus according to the present invention. In FIG. 1, 1 is a signal source such as a CD player, 2
is a signal processor for R channel, 3 is an L channel 4 for the R channel amplifier, 5 for the L
channel amplifier, 6 for the R channel speaker, 7 for the L channel speaker, 8 for the
measurement microphone, 10 for the CPU, 11 for the storage device (memory) , 12 is a cabin.
Although the case of two channels is described in this embodiment, the number of channels of
the sound field correction device of the present invention is not limited to this.
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[0013]
Further, the signal processing unit of the present invention has adaptive signal arrival time test
means, adaptive impulse response estimation means and adaptive inverse filter means.
The adaptive processing in the adaptive signal arrival time test means and the adaptive impulse
response estimation means is estimation of the impulse response of the system (identification of
an unknown system (system)), and the adaptive inverse filter means is an inverse filter of the
system. Design (inverse modeling).
These will be described later.
[0014]
FIG. 2 is a block diagram showing an example of the configuration of the signal processing unit.
In the present embodiment, although described as the L channel signal processing unit 3 for the
sake of explanation, the configurations of the R channel signal processing unit 2 and the L
channel signal processing unit 3 are equal. . In FIG. 2, 1 is the same signal source as FIG. 1, 21
and 23 are delay circuits, 22 is a reverse circuit (R channel) delay circuit (21), 24 and 25 are
adaptive FIR filters, 26 is a switch, 27 is It is a system (unknown system) to be corrected. In the
embodiment, the system 27 to be corrected is an example in which the L channel speaker 7 and
the measurement microphone in FIG. 1 are configured. FIG. 3 is a flowchart showing the
operation of the sound field correction apparatus of FIG. The operation of the sound field
correction apparatus will be described below with reference to FIG.
[0015]
<Operation of Sound Field Correction Device> [Step 31] Measurement of Arrival Time
Measurement microphones installed near the listener's head from the speakers 6 and 7 on the
left and right of the cabin 12 by adaptive signal arrival time verification means described later
Check the signal arrival time up to 8. As shown in the operation explanatory diagram of the
signal processing unit of FIG. 4, the signal source 1, the delay circuit 23, the adaptive FIR filter
24, the switch 26, and the system 27 to be corrected are operated. In FIG. 4, thick lines indicate
the flow of signals in the operation under the adaptive signal arrival time verification means
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(also, the block diagrams and symbols in FIGS. 4 to 6 are the same as in FIG. 2).
[0016]
First, as an initial setting, the delay time of the delay circuit 23 is set to 0 (zero), and all the
coefficients of the adaptive FIR filter 24 are set to 0 (zero). Music is output from only one
channel (in this case, L channel) from the signal source 1 (for example, a CD player), and the
arrival time is verified according to the principle of the adaptive signal arrival time verification
means (described later). The test result is set in the delay circuit 23 and sent to the CPU 10 and
stored in the memory 11. The same operation as described above is performed in the R channel
signal processing unit to obtain the arrival time to the measurement microphone 8.
[0017]
[Step 32] Estimation of Impulse Response Next, the impulse response between the speakers 6, 7
and the measuring microphone 8 is estimated. As for the impulse response between the L
channel speaker 6 and the measurement microphone 8, as shown in the operation explanatory
diagram of the signal processing unit of FIG. 4, the signal source 1, the delay circuit 23, the
adaptive FIR filter 24, the switch 26, and The system 27 is operated and switched to the switch
26a side.
[0018]
As an initial setting, all coefficients of the adaptive FIR filter 24 are set to 0 (zero), and the delay
circuit 23 performs a delay set by the adaptive signal arrival time checking means. The music of
only the L channel is output from the signal source 1 and the impulse response is estimated by
the principle of adaptive impulse response estimation means described later. The coefficients of
the adaptive FIR filter thus obtained are the estimated impulse response. The impulse response
can be similarly estimated for the reverse channel (R channel).
[0019]
[Step 33] Setting of Inverse Filter As shown in the operation explanatory diagram of the signal
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processing unit in FIG. 5, the signal source 1, delay circuit 21, adaptive FIR filter 25, delay circuit
23, adaptive FIR filter 24 and switch 26 are operated. Then, the inverse filter of the transfer
function (impulse response) between the speakers 6 and 7 and the measuring microphone 8
estimated in the above step 32 is designed. In FIG. 5, thick lines indicate the signal flow in the
operation at the time of designing the inverse filter of the impulse response.
[0020]
As an initial setting, the delay circuit 21 delays half of the maximum delay time of the adaptive
FIR filter 25 so that all the filter coefficients of the adaptive FIR filter 25 are 0, and the delay time
of the delay circuit 23 is 0. The filter coefficients of the FIR filter 24 set the result estimated by
the impulse response estimation means. Next, the measurement signal (for example, white noise)
is output from the signal source 1.
[0021]
Here, FIG. 2 corresponds to FIG. 9 (a block diagram of the inverse filter of the system described
later), and the adaptive FIR filter 24 of FIG. As the adaptive FIR filter 93 of FIG. 9, the inverse
filter is realized by the adaptive FIR filter 25 of FIG. 2 in accordance with the principle of the
adaptive inverse filter described later. This process is performed on the L channel and the R
channel. In this case, since it is not necessary to output a signal from the speaker, by turning off
the switch 26, the listener can be spared from hearing the unpleasant measurement signal.
[0022]
[Step 34] Setting of Arrival Time Difference The signal arrival time difference from the left and
right (R channel and L channel) speakers 6 and 7 is set. Test by adaptive signal arrival time
verification means and set arrival time (refer to step 31) stored in the memory 11 to the delay
circuit 23 of the reverse channel (this means that the arrival time of the reverse channel is set to
the delay circuit 23) Is the same as As a result, the signals reproduced from the left and right
speakers 6, 7 reach the listener simultaneously.
[0023]
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[Step 35] Output of a signal whose transfer function has been corrected As shown in the
operation explanatory diagram of the signal processing unit of FIG. 6, the signal source 1,
adaptive FIR filter 25, delay circuit 23, adaptive FIR filter 24 and switch 26 Make it work. The
delay circuit 23 delays the time corresponding to the arrival time of the reverse channel set in
the above step 34, and the adaptive FIR filter 24 makes only one coefficient and makes the other
zero. The adaptive FIR filter 25 has the inverse filter coefficient set in the process of step 33 set.
Also, the switch 26 is switched to the b side. As a result, the speakers 6 and 7 output a signal
whose transmission coefficient has been corrected. A thick line indicates the flow of signals in the
operation at the time of output of the signal whose transfer function has been corrected.
[0024]
The coefficients for each seat obtained by the above processing are stored in the memory 11 via
the CPU 10 so that the user can call them as needed. In the case of realizing this system in a
normal passenger car, assuming that the sampling frequency of the (digital) signal processing
unit is 44.1 kHz, the number of taps of the adaptive FIR filter 24 and the adaptive FIR filter 25 in
FIG. The order of 2000 taps is appropriate.
[0025]
Further, in the present embodiment, although the arrival time obtained by the principle of the
adaptive signal arrival time test means is stored in the memory via the CPU, the present
invention is not limited to this. For example, the arrival time is stored in the register of the signal
processing unit. It may be configured as follows.
[0026]
<Adaptive signal arrival time test means> FIG. 7 is a block diagram of an impulse response of a
system by an adaptive filter.
In FIG. 7, 1 is a signal source, 71 is a space to be estimated (unknown system), 72 is a speaker,
73 is a measurement microphone, 74 is an adaptive FIR filter, 75 is a delay circuit, s' is an
original signal, and n is Extraneous noise, s is a signal that has passed through the system, x is an
input signal (n + s) of the microphone, y is a signal obtained by filtering s', and e is an error signal
(xy).
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[0027]
In FIG. 7, first, the delay time of the delay circuit 75 is set to 0 (or the minimum value), music is
output from the signal source 1, and adaptive processing is performed. In this case, since the
delay time at the delay circuit 75 is zero, the time for the direct sound from the speaker 72 to
reach the measuring microphone 73 in the unknown system 71 ideally converges to zero. There
is a level at which the coefficients of the adaptive FIR filter correspond to the time at which.
Therefore, the adaptive FIR filter is tested while performing adaptive processing. In the test, as
shown in FIG. 13, whether the filter coefficient is equal to or higher than the threshold level is
examined sequentially from the shortest delay time. When the filter coefficient reaches the
threshold level, the number of delay steps of the filter coefficient corresponds to the delay time.
Also, the threshold level here is determined by the levels of x and s'. In FIG. 13, TL is a threshold
level, and K0 to Kn-1 correspond to the coefficients of the adaptive FIR filter of FIG.
[0028]
The maximum delay time of the adaptive FIR filter of this system must be longer than the arrival
time of the signal from the speaker 72 in the unknown system 71 to the measuring microphone.
This relationship is shown in the following equation (1). 1 / f · tap> t (1) where f: sampling
frequency, tap; the number of taps of the adaptive FIR filter, t: the arrival time of the sound from
the speaker to the measurement microphone.
[0029]
<Adaptive Impulse Response Estimation Means> The block diagram of the impulse response of
the system according to the adaptive filter of FIG. 7 is a system for estimating the impulse
response of the unknown system 71 by the adaptive FIR filter 74. FIG. 8 is a block diagram of an
adaptive processing algorithm, where x is an input signal, d is a desired signal, y is a signal
obtained by filtering x, e is an error signal (xy), and Z-1 is a one sample delay. , K are coefficients
of the adaptive FIR filter.
[0030]
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In the configuration of FIG. 8, the transfer function of the adaptive filter can be made close to the
transfer function of the system by making s'-filtered y closer to s in x. This can be realized by
minimizing the mean square error of the error e between x and y. Assuming that the mean
square error of e is E [e2], the mean square error of FIG. 8 can be expressed as the following
equation (2). E [e2] = E [(x-y) 2] = [n2] + 2E [n (s-y)] + E [(s-y) 2] (2) Signal s' and extraneous noise
in the system of FIG. When there is no correlation in n, the second term of the right side of
Equation (2) is zero. Therefore, Formula (2) becomes E [e2] = E [n2] + E [(sy) 2] (3). Here, to
minimize E [e2] is to minimize the second term of the right side of equation (3). Thus, the impulse
response of the adaptive filter is that of the system.
[0031]
The adaptive impulse response estimation means sets the arrival time of the signal from the
speaker 72 in the unknown system 71 to the measurement microphone 73 obtained by the
above-mentioned adaptive signal arrival time verification means in the delay circuit 75 of FIG.
Output music from source 1 Then, the above-described estimation process of the impulse
response of the system is performed. When the estimation result converges, the process ends.
The impulse response can be estimated more accurately if the maximum delay time of the
adaptive FIR filter of this system is closer to the reflected sound duration time of the unknown
system 71 (the case may be longer than the reflected sound duration time).
[0032]
<Design of Inverse Filter of System> FIG. 9 is a block diagram for designing an inverse filter by
adaptive processing. 1 is a signal source, 91 is a delay circuit, 92 is a system to be corrected, 93
is an adaptive FIR filter, s, d , X, y, e have the same meanings as the symbols in FIG. 7 and FIG. In
the block diagram of FIG. 9, an adaptive FIR filter 93 corrects and outputs a frequency
characteristic which has been disturbed by passing through a system 92 to be corrected. In
addition, the delay circuit 91 suitably has a half length of the adaptive FIR filter. The error signal
e is the difference between the desired output signal d of the system and the actual output y. In
the adaptive processing, the coefficients of the adaptive filter are adjusted so as to minimize e2.
[0033]
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<Adaptive Inverse Filter Means> In the adaptive inverse filter means, a system 92 to be corrected
as shown in FIG. 9 is created by the adaptive FIR filter estimated by the above-mentioned
adaptive impulse response estimation means. A measurement signal (for example, white noise) is
output from the signal source 1 and an inverse filter for the system 92 to be corrected is realized
by the adaptive FIR filter 93 by inverse filter design processing. Note that the number of taps of
the adaptive FIR filter 93 is preferably about the same as the number of taps of the FIR filter for
which the system 92 to be corrected is formed. Also, the measurement signal is preferably a
signal with a wide frequency band.
[0034]
As described above, according to the present invention, the following effects can be obtained.
Adaptive signal arrival time verification means (by application of an adaptive filter) can
automatically adjust the arrival time of the sound from the left and right speakers to the listener.
The external noise canceling means (adaptive impulse response means) can estimate the impulse
response of the system (in the vehicle compartment) with the adaptive filter even when the
external noise is large, for example, during traveling. By means of the adaptive inverse filter
means, it is possible to automatically correct the disturbed frequency characteristic with the
adaptive filter and listen to the sound of the optimum frequency characteristic at each seat.
Further, since correction is performed automatically by the adaptive filter, it is not necessary to
calculate the filter coefficients by the microcomputer and to hold the filter coefficients in the
memory as in the prior art. There is less burden on the listener since there is no need to emit the
measurement signal from the speaker during inverse filter design. The corrected result for each
seat can be stored in the storage device, and the user can recall the result as needed.
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