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JPH0675591

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DESCRIPTION JPH0675591
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
voice input device for reducing or eliminating unwanted signals (noises) from directions other
than the desired voice input direction.
[0002]
2. Description of the Related Art For example, as an audio input device used in a camera
integrated type VTR, it is desired to pick up only the audio from the subject direction and remove
the audio from the photographer side and other directions. .
[0003]
As described above, a unidirectional microphone such as that shown in FIG. 7 is generally used as
an audio input device that picks up only audio input from a specific direction.
Also, using two omnidirectional microphones, adding a circuit that adjusts the phase in an analog
manner to the output of one of the microphones, combining the output of the circuit with the
output of the other microphone, There is also known a method of giving high sensitivity
directivity in a specific incident direction.
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1
[0004]
SUMMARY OF THE INVENTION However, in the above case, although uni-directional, there is a
null (zero sensitivity) for voice input from a specific direction, and the highest sensitivity of
incidence is obtained. It has sensitivity in directions other than the direction. Therefore, when the
incident direction of the highest sensitivity is set to the incident direction of the desired voice,
unnecessary voices from the opposite direction of the desired voice are eliminated or reduced, as
is apparent from the characteristic diagram of FIG. However, in the case where the input
direction of the unnecessary voice is in the lateral direction with respect to the desired voice
input direction, the voice is picked up with, for example, 1/2 the size of the desired voice.
[0005]
In addition, even if the unnecessary voice from the opposite direction to the desired voice
direction that can be removed and reduced, if the voice level is large, it can not be reduced and is
picked up at a predetermined sound pressure level I will.
[0006]
In view of the above, the present invention can always eliminate or reduce unnecessary speech
even if the input direction of the unnecessary speech changes, and eliminate or reduce it
regardless of the size of the input level of the unnecessary speech. It is an object of the present
invention to provide a voice input device capable of
[0007]
SUMMARY OF THE INVENTION In order to solve the above problems, the voice input device
according to the present invention is used as a main input for picking up a desired voice when
the reference numerals of the embodiment of FIG. The microphones 11 and the main input
microphones 11 are spaced apart from each other by a predetermined distance, and the voice
signals from the reference input microphones 21 are unnecessary signals in the output sound
signals of the main input microphones 11. Adaptive filter means 25 for forming a signal
approximating components, delay means 24 provided between the reference input microphone
21 and the adaptive filter means 25, and an audio signal output from the main input microphone
11. Combining means 14 for reducing or eliminating unnecessary signals using the output signal
of means 25; and output power of the combining means 14 Characterized in that it comprises a
means for adjusting the adaptive filter means 25 so as to be minimized.
[0008]
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2
In the configuration of the present invention described above, the desired voice is picked up by
both the main input microphone 11 and the reference input microphone 21, but the reference
input is only delayed for the installation interval of both microphones. The desired voice input to
the microphone 21 is delayed.
Further, since the output sound signal of the reference input microphone 21 is further delayed
by the delay means, it is uncorrelated with the desired sound component in the output sound of
the main input microphone 11.
[0009]
Therefore, even if the synthesis circuit subtracts the output signal of the adaptive filter means 25
from the output sound signal of the main input microphone 11, the desired voice is not reduced.
[0010]
On the other hand, unnecessary voices are also collected by the reference input microphone 21
and the main input microphone 11, and unnecessary voices input to the main input microphone
11 for the delay time according to the incident direction thereof are the reference input
microphone 21. It is delayed than the input point of.
However, the output sound signal of the reference input microphone is delayed by the delay
means 24.
Therefore, when looking at the timing of the two input signals of the synthesizing means 14, the
unnecessary signal component in the output sound signal of the main input microphone and the
unnecessary signal component in the output sound signal of the reference input microphone are
temporally It becomes the timing which agrees.
And since the amplitudes of both unnecessary signal components are adjusted to coincide with
each other by the adaptive filter means, the unnecessary signal components are eliminated and
reduced from the output sound signal of the main input microphone 11 in the synthesizing
means 14. is there.
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[0011]
In this case, even if the incident direction of the unnecessary sound collected by the reference
input microphone 21 changes, it is reduced. In addition, the unnecessary signal is reduced or
eliminated by the synthesizing means regardless of the input level of the unnecessary speech.
[0012]
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT An embodiment of a voice input
device according to the present invention will be described below with reference to the drawings.
In the present invention, the concept of adaptive noise reduction processing is used, so an
embodiment of the present invention will be described. This adaptive noise reduction process will
be described before.
[0013]
FIG. 3 is a block diagram of a basic configuration of the adaptive noise reduction processing
system, in which 1 is a main input terminal, 2 is a reference input terminal, and a main input
signal input through the main input terminal 1 is through delay circuit 3 Is supplied to the
synthesis circuit 4.
In the delay circuit 3, when there is no time delay between the main input signal input to the
main input terminal 1 and the reference input signal input to the reference input terminal 2, the
time delay in the adaptive filter circuit 5 is It is for correcting the minute. Further, the signal
input through the reference input terminal 2 is supplied to the combining circuit 4 through the
adaptive filter circuit 5 and is subtracted from the signal from the delay circuit 3. Then, the
output of the synthesis circuit 4 is fed back to the adaptive filter circuit 5 and also led out to the
output terminal 6.
[0014]
In this noise reduction device, the main input terminal 1 receives a signal obtained by adding the
desired signal s and the noise n0 uncorrelated thereto. On the other hand, the noise n1 is input
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to the reference input terminal 2. The noise n1 of the reference input is uncorrelated with the
desired signal, but is correlated with the noise n0.
[0015]
The adaptive filter circuit 5 filters the reference input noise n1 and outputs a signal
approximating the noise n0, that is, a signal y equal in amplitude to the noise n0. As an output
signal of the adaptive filter circuit 5, it is possible to obtain a signal -y having the same amplitude
as that of the noise n0. The synthesizing circuit 4 performs a process of subtracting the output
signal of the adaptive filter circuit 5 from the output signal of the delay circuit 3 (in the case
where the output signal is a signal -y in reverse phase to the noise n0).
[0016]
The adaptive algorithm in the adaptive filter circuit 5 serves to minimize the subtraction output
(residual output) e which is the output of the combining circuit 4. That is, assuming that s, n0, n1,
and y are statistically stationary and the average value is 0, the residual output e is e = s + n0−y.
Since the expected value of the square of this is uncorrelated with s and n0 and y, E [e2] = E [s2]
+ E [(n0-y) 2] + 2E [s (n0-y) ] = E [s2] + E [(n0-y) 2]. Assuming that the adaptive filter circuit 5
converges, the adaptive filter circuit 5 is adjusted so as to minimize E [e2]. At this time, since E
[s2] is not affected, Emin [e2] = E [s2] + Emin [(n0-y) 2]. That is, E [(n0-y) 2] is minimized by
minimizing E [e2], and the output y of the adaptive filter circuit 5 is an estimate of the noise n0.
The expected value of the output from the combining circuit 4 is only the desired signal s. That
is, adjusting the adaptive filter circuit 5 to minimize the total output power is equal to the
subtraction output e being the least squares estimated value of the desired speech signal s.
[0017]
The adaptive filter circuit 5 can be realized either by an analog signal or by a digital signal
processing circuit. An example of the case where the adaptive filter circuit 5 is realized using a
digital filter is shown in FIG. This example uses the so-called LMS (least mean square) method as
the adaptation algorithm.
[0018]
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As shown in FIG. 4, in this example, an adaptive linear combiner 300 of FIR filter type is used.
This is because a plurality of delay elements DL1, DL2,... DLm (m is a positive integer) each
having a delay time Z-1 (= τ) of a unit sampling time, an input noise n1, and each delay element
DL1, DL2. ,..., A weighting circuit MX0, MX1, MX2,... MXm for multiplying the output signal of
DLm and the weighting factor, and an adding circuit 310 for adding the outputs of the weighting
circuits MX0 to MXm. The output of the summing circuit 310 is y.
[0019]
Weighting coefficients to be supplied to the weighting circuits MX0 to MXm are formed based on
the residual signal e from the synthesizing circuit 4 by an LMS operation circuit 320 comprising,
for example, a microcomputer. The algorithm executed by this LMS arithmetic circuit 320 is as
follows.
[0020]
Now, let the input vector Xk at time k be Xk = [x0k x1k x2k ... xmk] T, as also shown in Fig. 4, the
output yk, and the weighting coefficient wjk (j = 0, 1, 2, ... Assuming that m), the input / output
relationship is as shown in the following equation 1:
[0021]
Then, if the weight vector Wk at time k is defined as Wk = [w0kw1kw2k... Wmk] T, the input /
output relation is given by yk = XkT.Wk.
Here, assuming that the desired response is dk, the residual ek is expressed as follows. In the ek =
dk-yk = dk-Xk T · WkLMS method, updating of the weight vector is sequentially performed
according to the equation Wk + 1 = Wk + 2μ · ek · Xk. Here, μ is a gain factor (step gain) that
determines the speed and stability of adaptation.
[0022]
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Next, FIG. 1 shows a block diagram of an embodiment of a voice input device according to the
present invention using the above-mentioned adaptive noise reduction processing. In this
example, the adaptive filter circuit uses the digital filter of the configuration of FIG. 4 described
above. Further, as the synthesis circuit 4, a subtraction circuit is used.
[0023]
In FIG. 1, reference numeral 11 denotes a main input microphone for collecting a desired voice,
and reference numeral 21 denotes a reference input microphone for collecting unnecessary voice
in a direction to be removed as noise. In the case of this example, both the main input
microphone 11 and the reference input microphone 21 are configured by omnidirectional
microphones as shown in FIG. Further, as shown in FIG. 2, the main input microphone 11 and the
reference input microphone 21 are disposed apart from each other by a distance d.
[0024]
In this example, the arrival direction of the desired voice is mainly from the top to the bottom in
the figure, as indicated by the arrow AR in FIG. In this example, a voice input device is realized
which reduces or eliminates voices from directions different by about 90 ° to about 270 ° with
respect to the direction AR as unnecessary voices (noises).
[0025]
Then, an audio signal obtained by being picked up by the main input microphone 11 and
converted into an electric signal is supplied to the A / D converter 13 through the amplifier 12
and converted into a digital signal, and is then supplied to the subtraction circuit 14. Supplied.
[0026]
Further, an audio signal obtained by being collected by the reference input microphone 21 and
converted into an electric signal is supplied to the A / D converter 23 through the amplifier 22
and converted into a digital signal.
The digital signal from the A / D converter 23 is supplied to the adaptive filter circuit 25 through
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the delay circuit 24. Then, the output signal of the adaptive filter circuit 25 is supplied to the
subtraction circuit 14. The output signal of the subtraction circuit 14 is fed back to the adaptive
filter circuit 25 and also converted back to an analog signal by the D / A converter 15 and is led
out to the output terminal 16. Note that the audio signal may be output as it is without passing
through the D / A converter 15.
[0027]
In the case of this example, the delay amount of the delay circuit 24 and the maximum delay
amount of the linear coupler 300 in the adaptive filter circuit 25 (unit delay amount τ of delay
elements × number of unit delay elements = τ × (number of taps−1) And the sum is selected
equal to the time for sound to propagate through the distance d between the two microphones
11 and 21.
[0028]
In the adaptive filter circuit 25, as described above, the reference input speech is controlled to
approximate the unwanted speech as noise contained in the main input speech.
As a result, assuming that the desired voice in the voice collected by the main input microphone
11 and the noise are uncorrelated, in the subtraction circuit 14, the noise signal collected by the
reference input microphone 21 is the main input. The speech signal from the microphone 11 is
subtracted and removed, and only the desired speech is obtained from the subtraction circuit 14.
[0029]
The above configuration is a configuration of an adaptive noise reduction system in which the
output voice signal of the main input microphone 11 is input as the main input and the output
voice signal of the reference input microphone 21 is supplied as the reference input noise. The
operation of this system will now be described.
[0030]
In this case, the desired voice is picked up by both the main input microphone 11 and the
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reference input microphone 21, but the delay time for the distance d between the two
microphones 11 and 21 (sound propagates through the distance d) It is time, and assuming that
the sound propagation speed is c, the desired voice input to the reference input microphone 21 is
delayed by d / c) from the desired voice input to the main input microphone 11 . Further, since
the output voice signal of the reference input microphone 21 is further delayed by the delay
circuit 24 and the adaptive filter circuit 25, it is uncorrelated with the desired voice component
in the output voice of the main input microphone 11. .
[0031]
Therefore, even if the subtraction circuit 14 subtracts the output signal of the adaptive filter
circuit 25 from the output sound signal of the main input microphone 11, the desired sound is
not reduced.
[0032]
On the other hand, unnecessary voices are also collected by the reference input microphone 21
and the main input microphone 21. However, since the main input microphone 11 and the
reference input microphone 21 are disposed apart from each other by the distance d, The
unnecessary voice input to the main input microphone 11 is delayed with respect to the
unnecessary voice input of the reference input microphone 21 by the delay time according to the
arrival direction of the unnecessary voice.
However, the output sound signal of the reference input microphone 21 is delayed by the delay
circuit 24 and the adaptive filter circuit 25. Then, the adaptive filter circuit 25 works so that the
time lag between the unnecessary signal component in the main input and the unnecessary
signal component in the reference input is eliminated in the subtraction circuit 14 because of the
above-mentioned nature.
[0033]
That is, when the unnecessary voice arrival direction is a direction different from the desired
voice arrival direction AR by 180 ° in FIG. 5, that is, in the back direction, the unnecessary voice
input to the main input microphone 11 is performed by the reference input microphone 21. Just
delay by d / c for the unwanted signal input. At this time, the adaptive filter circuit 25 works to
maximize the weighting factor of the final stage tap, and the delay amount in the adaptive filter
03-05-2019
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circuit 25 is maximized. Therefore, the sum of the delay amount with the delay circuit 24 is equal
to d / c, and in the subtraction circuit 14, there is no time lag between the unwanted signal
component from the main side and the unwanted signal component from the reference side.
Since the amplitude is also adjusted by the adaptive filter circuit 25, unwanted signal
components are removed from the main input signal.
[0034]
Next, when the unnecessary voice comes from the oblique back indicated by 'in FIG. 5, the
unnecessary voice reaches the microphone 21 assuming that the incident angle of the
unnecessary voice with respect to the desired voice direction AR at that time is θ. The time to
reach the microphone 11 to the microphone 11 is d · cos θ / c, which is shorter than in the case
of the direction. Therefore, the weighting coefficient of the adaptive filter circuit 25 is a tap
before the tap of the final stage, and the coefficient of the tap which passes through the delay
time corresponding to the delay time d · cos θ / c becomes large. Components are removed from
the main input speech.
[0035]
Further, when the unnecessary voice comes from the side direction indicated by 'in FIG. 5, the
delay between the input time of the microphone 11 of the unnecessary voice and the input time
of the microphone 21 becomes slight. In this case, when the delay amount of the delay circuit 24
is small, the adaptive filter circuit 25 eventually cancels unnecessary speech in the subtraction
circuit 14 because the first tap coefficient becomes large.
[0036]
FIG. 6 is a diagram for explaining the basic principle configuration of the present invention. That
is, as shown in FIG. 6, when the plane waves P1 and P2 arrive at a position separated by a
distance d at an incident angle θ with respect to the direction AR, P2 = P1 · Exp (−jωd · cosθ /
c) . Then, P2d obtained by adding a delay to P2 becomes P2d = P2 · Exp (−jωnT) (T: sampling
interval of audio signal). Accordingly, P = P1−P2d = P1 (1−Exp (−j (ωnT + ωd · cosθ / c))).
Here, for example, when the unnecessary voice arrives from the direction of FIG. 5, θ = 180 °,
nT = d / cP = P1 (1−Exp (−jωd / c (1 + cosθ))). When the characteristics represented by this
equation are illustrated, it becomes unidirectional as shown in FIG.
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[0037]
Similarly, when it is assumed that the unnecessary voice comes from the direction of FIG. 5, ', the
directivity characteristic of the voice input device of this example is as shown in FIG. Further,
when it is assumed that the unnecessary voice comes from the direction of FIG. 5 ′, the
directivity characteristic of the voice input device of this example is as shown in FIG. As
described above, the directivity characteristic is such that the sensitivity in the arrival direction of
the unnecessary signal is always zero, and the characteristic is reduced and eliminated regardless
of the input level of the unnecessary voice.
[0038]
In the device of the example of FIG. 1, the frequency response before processing is shown in FIG.
10 after processing when desired voice is input from the direction AR and a sine wave of 500 Hz
is added as an unnecessary signal from the rear direction different from 180 °. The frequency
response of is shown in FIG. From these FIGS. 10 and 11, it can be confirmed that the 500 Hz
sine wave as the unnecessary signal is sufficiently reduced.
[0039]
The sum of the delay amount of the delay circuit 24 and the maximum delay amount according
to the total number of taps of the adaptive filter circuit 25 has no reduction effect on the voice
input from the desired voice arrival direction, and the desired voice arrival direction The value is
selected to have a reduction effect for voice input from outside. Preferably, the unnecessary
signal can be favorably reduced if it is twice or less the time d / c during which the sound
propagates in the distance d between the two microphones 11 and 21. However, the sum of the
delay amounts may be twice or more of the time d / c.
[0040]
FIG. 12 shows an embodiment in which the voice input device according to the present invention
is applied to a stereo microphone device. In FIG. 12, the microphone 11L is for sound collection
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11
of the left channel, and the microphone 11R is for sound collection of the right channel, and the
two microphones 11L and 11R are disposed apart by a distance d. The arrival direction of the
voice of the left channel is the direction of the arrow aL in the drawing, and the arrival direction
of the voice of the right channel is the arrow aR of the drawing. Then, the microphone 11L serves
as a reference input for the sound of the right channel, and the microphone 11R serves as a
reference input for the sound of the left channel.
[0041]
First, to describe the configuration of the audio signal of the left channel, the audio signal of the
left channel obtained by being picked up by the microphone 11L and converted into an electrical
signal is supplied to the A / D converter 13L through the amplifier 12L. The digital signal is
converted to a digital signal and supplied to the subtraction circuit 14L.
[0042]
Further, the unnecessary audio signal for the left channel audio, which is collected by the
microphone 11R and converted into an electric signal, is supplied to the A / D converter 13R via
the amplifier 12R and converted into a digital signal. .
The digital signal from the A / D converter 13R is supplied to the adaptive filter circuit 25L via
the delay circuit 24L. Then, the output signal of the adaptive filter circuit 25L is supplied to the
subtraction circuit 14L. The output signal of the subtraction circuit 14L is fed back to the
adaptive filter circuit 25L, and is converted back to an analog signal by the D / A converter 15L
and is derived to the output terminal 16L. The sum of the delay amounts of the delay circuit 24L
and the adaptive filter circuit 25L is selected in the same relationship as the sum of the delay
amounts of the delay circuit 24 and the adaptive filter circuit 25 in the example of FIG.
[0043]
With this configuration, unnecessary voice from the reverse direction aR is reduced or eliminated
from the voice of the left channel from the output voice signal of the microphone 11L, and the
voice signal of the left channel from the voice arrival direction aL is output to the output terminal
16L. Only
[0044]
Next, the configuration of the audio signal of the right channel will be described. The audio signal
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of the right channel obtained by being picked up by the microphone 11R and converted into an
electric signal is supplied to the A / D converter 13R via the amplifier 12R. , Converted into a
digital signal, and supplied to the subtraction circuit 14R.
[0045]
Further, the digital signal of the unnecessary audio signal for the right channel audio from the A
/ D converter 13R is supplied to the adaptive filter circuit 25R via the delay circuit 24R.
Then, the output signal of the adaptive filter circuit 25R is supplied to the subtraction circuit
14R.
The output signal of the subtraction circuit 14R is fed back to the adaptive filter circuit 25LR and
is converted back to an analog signal by the D / A converter 15R and is derived to the output
terminal 16R.
[0046]
The sum of the delay amounts of the delay circuit 24R and the adaptive filter circuit 25R is also
selected in the same relationship as the sum of the delay amounts of the delay circuit 24 and the
adaptive filter circuit 25 in the example of FIG.
[0047]
With this configuration, unnecessary voice from the reverse direction aL is reduced or eliminated
from the voice of the right channel from the output voice signal of the microphone 11R, and the
voice signal of the right channel from the voice arrival direction aR is output to the output
terminal 16R. Only
[0048]
In the example shown in FIG. 1, the reference input microphone 21 is a unidirectional
microphone as shown in FIG. 13. The sensitivity to the desired voice arrival direction AR is small,
and high sensitivity is shown in the opposite direction. It may be arranged in
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Further, in the example of FIG. 1, a delay circuit may be further provided after the adaptive filter
circuit 25.
[0049]
The above example is an example in which the fixed delay circuit 24 is disposed in front of the
adaptive filter circuit, but a configuration using the delay element in the linear coupler of the
adaptive filter circuit without providing the delay circuit 24 and You can also
[0050]
The example of FIG. 14 is a block of one embodiment in that case.
In this example, the same parts as those in the example of FIG.
Also in this example, the main input microphone 11 and the reference input microphone 21 are
disposed apart from each other by a predetermined distance d. Further, as in the example of FIG.
1, as these microphones 11 and 21, both microphones may be configured with no directivity as
shown in FIG. 2, or as shown in FIG. The microphone 11 may be nondirectional, and the
reference input microphone 21 may be unidirectional. In this example, as shown in FIG. 15, both
the microphones 11 and 21 are both made of a single directional microphone. The main input
microphone 11 is disposed so that the direction of the maximum sensitivity is directed to the
desired voice arrival direction AR, and the reference input microphone 21 is disposed such that
the desired voice arrival direction AR is the lowest sensitivity (sensitivity zero). ing.
[0051]
Then, in this example, the output of the reference A / D converter 23 is directly supplied to the
adaptive filter circuit 30. The maximum delay amount in the adaptive filter circuit 30
corresponds to the sum of the maximum delay amounts of the delay circuit 24 and the adaptive
filter circuit 25 in the example of FIG. 1 described above, and preferably 2 of d / c. It will be a
value within double. In the example of the figure, the adaptive filter circuit 30 has a tap number
of 5 as a value satisfying the above condition, and four delay elements 31 to 34 and five weight
circuits 41 to 45, It comprises the LMS arithmetic circuit 50.
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14
[0052]
In the case of this example, a control signal is supplied to the LMS arithmetic circuit 50 from the
control terminal 51. According to the control signal, the LMS arithmetic circuit 50 forcibly sets
all weighting coefficients from the tap of the first stage to the taps of the i (i = 1, 2,... In this case,
the adaptive filter circuit 30 is equal to the state where the simple delay corresponding to the
number of stages is inserted up to the tap of the stage where the coefficient is made zero.
However, the amount of the simple delay is variable according to the number i of stages in which
the coefficient is forced to be zero. The number of stages i for which the coefficient is forced to
zero is determined by the control signal from the control terminal 51.
[0053]
The operation of the example of FIG. 14 will be described below. In the case where there is no
tap for which the weighting factor is forced to be zero, the adaptive filter circuit 30 selects the
weighting factor of the tap according to the unwanted signal arrival direction. It is adjusted to be
maximum to reduce unnecessary signals.
[0054]
That is, when the unnecessary signal arrives from the direction shown in FIG. 5, the unnecessary
signals arrive at the two microphones 11 and 21 at substantially the same time, so in the
adaptive filter circuit 30, the taps of the first stage The coefficient of the weighting circuit 41 is
maximized, and the temporal timings of the unnecessary signal component on the main side
input to the subtraction circuit 14 and the unnecessary signal component on the reference side
coincide, and the amplitude is also adjusted by the adaptive filter circuit 30 The unwanted signal
components are removed from the main input signal.
The directional characteristics of the output of the device of FIG. 14 at this time are as shown in
FIG.
[0055]
Further, when an unnecessary signal arrives from the direction of FIG. 5 ', the weighting factor of
the weighting circuits 42 to 44 of the taps of the second to fourth stages becomes maximum
03-05-2019
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according to the angle with the direction AR, and further, When an unnecessary signal arrives
from the direction of FIG. 5, the weighting factor of the weighting circuit 45 of the tap of the final
stage is maximized, and the unnecessary signal component is removed from the main input
signal in the same manner as described above.
[0056]
Directional characteristics of the output of the apparatus of FIG. 14 when removing unwanted
signals coming from direction, 'close direction,' are shown in FIG. 16 and unwanted signals
coming from direction ', direction' are removed The directional characteristics of the output of
the device of FIG. 14 when in motion are shown in FIG.
Further, FIG. 18 shows the directivity characteristic of the output of the apparatus of FIG. 14
when the unnecessary signal coming from the direction is removed.
[0057]
Next, as shown in FIG. 20A by the control signal, when the coefficient of the weighting circuit 41
of the first stage tap is made zero, the reduction amount or removal amount of the unnecessary
signal coming from the direction shown in FIG. Will be reduced. That is, as described above,
when the unnecessary signal arrives from the direction ', the time when the unnecessary signal
reaches the two microphones 11 and 21 is almost simultaneous, but the coefficient of the tap in
the first stage is zero Therefore, a fixed delay due to the delay element 31 is inserted into the
reference input, and the input point in the subtraction circuit 14 differs between the main side
and the reference side with respect to the unnecessary signal coming from the direction '. . For
this reason, the reduction amount of the unnecessary signal arriving from this direction, 'is
reduced. However, the unnecessary signal from the direction closer to the 'side than this
direction,' ', can be timed by the adaptive filter circuit 30, so it can be favorably reduced or
eliminated. This state is a state showing the directivity characteristic shown in FIG.
[0058]
Next, as the number of taps for fixing the weighting factor to zero is increased sequentially from
the first stage as shown in FIG. 20B by the control signal, the reduction or removal amount
decreases in the direction 'from side to side'. It is limited and substantially equivalent to moving
from the directivity characteristic of FIG. 16 to the directivity characteristic of FIG.
03-05-2019
16
[0059]
Then, as shown in FIG. 20C, when only the coefficients of the taps of the final stage are updated,
and all others are made zero, only the unnecessary voice from the direction of FIG. The amount
removed will be reduced.
That is, it becomes the same as the directivity characteristic of FIG.
[0060]
Therefore, according to the embodiment of FIG. 14, when the weighting factor is made zero
sequentially from the tap of the first stage by the control signal, the arrival direction of the
unnecessary signal which can be removed gradually becomes limited. Then, in a state where the
coefficients of only the final stage taps are updated, only the sound coming from the direction is
removed. Thus, according to this example, the zoom effect of the sound is obtained.
[0061]
Therefore, for example, a signal corresponding to the zoom operation of the lens of the camera
integrated VTR can be supplied to the control terminal 51. That is, for example, when the zoom
side is set to the telephoto side, sound that comes from a wide range on the rear side of the
subject is adaptively removed without using a tap that sets the coefficient to zero by the control
signal. Make it possible to pick up only voice. From this state, when the zoom lens is gradually
shifted to the wide angle side, the number of taps for which the coefficient is zero is sequentially
increased from the first stage side by the control signal to narrow the range of arrival directions
of removable sounds. By doing this, it is possible to obtain an input voice according to the
shooting state.
[0062]
In addition, a manual operation knob may be provided, and the number of taps from the first
stage which makes the coefficient zero may be selected by the operation knob.
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[0063]
In the example of FIG. 14, a fixed delay circuit may be inserted before and / or after the adaptive
filter circuit 30.
[0064]
The calculation method for updating the weighting coefficients in the adaptive filter circuit is not
limited to the LMS method, and a learning identification method or another algorithm can be
used.
[0065]
As described above, according to the present invention, even if the input direction of the
unnecessary speech changes, the unnecessary speech can be always eliminated or reduced by
using the adaptive filter circuit, and It can be eliminated or reduced regardless of the size of the
input level.
[0066]
Also, by providing means for forcing the weight coefficient from the first stage of the adaptive
filter circuit to a predetermined stage to be zero in order, the arrival direction of the unwanted
signal to be eliminated or reduced can be limited, and the zoom effect of the sound can be
reduced. You can get it.
[0067]
Brief description of the drawings
[0068]
1 is a block diagram of an embodiment of the voice input device according to the present
invention.
[0069]
2 is a diagram for explaining the arrangement and directivity of the two microphones used in the
example of FIG.
[0070]
03-05-2019
18
3 is a block diagram for explaining the adaptive noise reduction system.
[0071]
4 is a block diagram of an example of an adaptive filter circuit used in the adaptive noise
reduction system.
[0072]
5 is a diagram for explaining the operation of an embodiment of the present invention.
[0073]
6 is a simplified principle configuration diagram of an embodiment of the present invention.
[0074]
7 is a directional characteristic diagram at a certain moment of the output of the device of the
example of FIG.
[0075]
8 is a directional characteristic diagram at an instant of the output of the device of the example
of FIG.
[0076]
9 is a directional characteristic diagram at an instant of the output of the device of the example
of FIG.
[0077]
10 is a characteristic diagram of an example of voice input to an example of the present
invention.
[0078]
11 is a characteristic diagram of the output signal of one embodiment of the present invention to
which the voice of the characteristic of FIG. 10 is inputted.
03-05-2019
19
[0079]
12 is a block diagram of another embodiment of the present invention.
[0080]
13 is a diagram for explaining another example of two microphones used in the present
invention.
[0081]
FIG. 14 is a block diagram of still another embodiment of the present invention.
[0082]
15 is a diagram for explaining an example of two microphones used in the example of FIG.
[0083]
16 is a diagram showing directivity characteristics for explaining the embodiment of FIG.
[0084]
17 is a diagram showing directivity characteristics for explaining the embodiment of FIG.
[0085]
18 is a diagram showing the directivity characteristics for explaining the embodiment of FIG.
[0086]
19 is a diagram showing directivity characteristics for explaining the embodiment of FIG.
[0087]
20 is a diagram for explaining the operation of the embodiment of FIG.
[0088]
Explanation of sign
03-05-2019
20
[0089]
11 microphone for main input 14 subtraction circuit 21 microphone for reference input 24 delay
circuit 25 adaptive filter circuit 30 adaptive filter circuit 50 LMS arithmetic circuit 51 control
signal input terminal
03-05-2019
21
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