JPH06292291

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DESCRIPTION JPH06292291
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
so-called superdirective microphone device using an adaptive noise canceller.
[0002]
2. Description of the Related Art For example, in a camera integrated VTR, while photographing
an object, the sound around the object is simultaneously recorded. In collecting the sound, it is
generally considered that only the sound from the direction of the subject is collected. That is, for
example, a microphone device having a directional characteristic that picks up only the sound
from the front of the camera is used.
[0003]
As an example of this type of microphone device, for example, a so-called gun microphone is
known. It comprises a pipe section which extends to the front of the diaphragm. Then, a large
number of through holes are provided on the side wall of this pipe portion, and it has directivity
having high sensitivity to the sound from the front (the opposite direction to the diaphragm) of
the pipe portion in the centerline direction. Is configured as.
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[0004]
However, this microphone requires a long pipe portion and has the drawback of becoming large.
Moreover, it is uni-directional with high sensitivity only in front of the microphone, and only
fixed directivity can be obtained. Therefore, it is difficult to cope with the case where voices from
not only the voice from the desired voice arrival direction but also voices from the side around
the camera are to be picked up, for example, and there is no freedom in the direction of
directivity.
[0005]
Therefore, the applicant has proposed a microphone device which can be made compact and
have superdirective characteristics by applying an adaptive noise canceller as Japanese Patent
Application No. 4-143209.
[0006]
FIG. 4 shows a basic configuration of the adaptive noise canceller. First, the adaptive noise
canceller will be described.
[0007]
In FIG. 4, 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 supplied to the synthesis circuit 4 through the delay circuit
3.
Further, the reference input signal input through the reference input terminal 2 is supplied to the
synthesis circuit 4 through the adaptive filter circuit 5 and is subtracted from the signal from the
delay circuit 3.
The output of the synthesis circuit 4 is fed back to the adaptive filter circuit 5 and is led to the
output terminal 6.
[0008]
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In this adaptive noise canceller, the main input signal is the sum of the desired signal s and the
noise signal n0 uncorrelated with it. On the other hand, the noise signal n1 is input as the
reference input signal. The noise signal n1 of the reference input is uncorrelated with the desired
signal s, but is correlated with the noise signal n0.
[0009]
The adaptive filter circuit 5 filters the reference input noise signal n1 and outputs a signal y
which approximates the noise signal n0. In this case, the adaptive filter circuit 5 updates the
filtering coefficient of the reference input noise signal n1 by the predetermined adaptive
algorithm so that the subtraction output (residual output) e of the combining circuit 4 is
minimized. Go on.
[0010]
As the output signal y of the adaptive filter circuit 5, it is also possible to obtain a signal having
the same amplitude as that of the noise signal n0. The delay circuit 3 compensates for the time
delay required for the arithmetic processing in the adaptive filter circuit 5, the propagation time
in the adaptive filter, and the like, and is used to time align with the signal to be subtracted.
[0011]
The principle of the adaptive noise canceller will be described below.
[0012]
Now, assuming that the desired signal s, the noise n0, the noise n1, and the output signal y are
statistically stationary and the average value is zero, the residual output e becomes e = s + n0−y.
The expected value of this squared is E [e2] = E [s2] + E [(n0-y) 2] + 2E [s (n0-y) because the
desired signal s has no correlation with the noise n0 and the output y. ]] = E [s2] + E [(n0-y) 2].
Assuming that the adaptive filter circuit 5 converges, the adaptive filter circuit 5 updates the
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adaptive filter coefficients such that E [e2] is minimized. At this time, E [s2] is not affected, so
Emin [e2] = E [s2] + Emin [(n0-y) 2].
[0013]
That is, E [(n0-y) 2] is minimized by minimizing E [e2], and the output y of the adaptive filter
circuit 5 becomes an estimate of the noise signal 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.
[0014]
Although the output e generally has some noise remaining in the signal s, since the output noise
is given by (n 0 −y), minimizing E [(n 0 −y) 2] is an output It is equivalent to maximizing the
signal to noise ratio.
[0015]
In some cases, the synthesis circuit 4 may be acoustic synthesis means.
That is, the adaptive filter circuit 5 forms a noise cancellation voice signal -y having the same
phase as noise and the opposite phase of noise and supplies this to a speaker or the like to
acoustically add to the main voice to reduce noise. Do. The residual e in this case is picked up by
the residual detection microphone.
[0016]
The adaptive filter circuit 5 can be realized either by an analog signal processing circuit or a
digital signal processing circuit, but in general, a digital processing circuit using a DSP (digital
signal processor) It is made up of An example of the configuration of the adaptive filter circuit 5
in the case of the configuration of the digital processing circuit is shown in FIG.
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[0017]
In this example, the adaptive filter circuit 5 comprises an FIR filter type adaptive linear combiner
100 and a filter coefficient update computing means 110. The adaptive filter circuit 5 can be
configured as software by a DSP on which a microcomputer is mounted. In this example, the
algorithm for updating the filter coefficient will be described as using LMS (Least Mean Squares),
which is frequently used because the amount of calculation is small and practical.
[0018]
The LMS method will be described with reference to FIG. As shown in FIG. 6, in this case, an
adaptive linear combiner 100 of FIR filter type is used. The adaptive linear coupler 100 includes
a plurality of delay circuits DL1, DL2,... DLm (m is a positive integer) each having a delay time Z-1
of unit sampling time, an input noise n1, and each delay circuit DL1. , DL2,..., A weighting circuit
MX0, MX1, MX2,... MXm for multiplying the output signal of the DLm and the weighting
coefficient (filter coefficient), and an adding circuit 101 for adding the outputs of the weighting
circuits MX0 to MXm. The output of the adder circuit 101 is the signal y described in FIG.
[0019]
The weighting coefficients to be supplied to the weighting circuits MX0 to MXm are formed by
the filter coefficient arithmetic circuit 110 based on the residual signal e from the synthesizing
circuit 4 and the reference input n1 by the LMS algorithm. The algorithm executed by the filter
coefficient calculation circuit 110 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. 6,
output yk, weighting coefficient wjk (j = 0, 1, 2, ... Assuming that m), the input / output
relationship is as shown in the following equation 1:
[0022]
Then, if the weight vector Wk at time k is defined as Wk = [w0k w1k w2k... Wmk] T, the input /
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output relation is given by yk = Xk T · Wk (1).
Here, assuming that the desired response is dk, the residual ek is ek = dk −yk = dk −Xk T · Wk
(2)
[0023]
In the LMS method, updating of the weight vector is sequentially performed according to the
equation (3) as Wk + 1 = Wk + 2 μ · ek · Xk (3). The initial value of the weighting factor is set to
a constant value or a random value. Here, μ is a gain factor (step gain) that determines the speed
and stability of adaptation.
[0024]
The microphone device proposed above is configured using the adaptive noise canceller as
described above. That is, as shown in FIG. 4, the first microphone 7 for main input sound
collection and the second microphone 8 for reference input sound collection are provided, and
the output signal of the first microphone 7 is input to the main input terminal 1. And the output
signal of the second microphone 8 is input to the reference input terminal 2.
[0025]
Then, in this case, when the desired voice arrival direction is AR, the main input microphone 7
uses a nondirectional microphone as shown in FIG. Alternatively, a unidirectional microphone is
placed with its main pointing axis in the desired voice incoming direction.
[0026]
In addition, as shown in FIG. 5, the microphone for reference input 8 has sensitivity to the voice
input direction to be excluded, and the desired voice arrival direction AR is a null direction of
directivity, as shown in FIG. Arrange and use. If the voice arrival direction to be excluded is 90
degrees to the desired voice arrival direction, use a bi-directional microphone so that the desired
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voice arrival direction AR has a directivity null direction You may
[0027]
With such a configuration, the reference input signal hardly contains the desired voice signal, so
the reference input signal becomes a signal uncorrelated with the desired voice, and an
unnecessary signal (noise component) in the main input It becomes a signal having a correlation.
Therefore, if the second microphone 8 for collecting the reference input is configured to have a
predetermined sensitivity in the arrival direction of the unnecessary signal to be reduced, the
unnecessary signal component included in the main input is adaptively canceled. And only the
desired voice signal can be obtained at the output terminal.
[0028]
As described above, by using the adaptive noise canceller, it is possible to realize a superdirective
microphone device having a sharp directivity in the direction of the sensitivity minimum of the
unidirectional microphone 8 for reference input.
[0029]
However, as a practical matter, it is difficult to completely change the directivity of the reference
input sound pickup microphone so as not to pick up the desired sound signal (make the
sensitivity zero).
For this reason, it is inevitable that the desired audio signal is mixed into the reference input
signal at a certain level.
[0030]
This state deviates from the precondition of adaptive processing that both input signals of
desired speech and reference input speech are uncorrelated, especially when the level of
unnecessary signal (noise) of reference input is considerably low. In the conventional adaptive
processing such as the above, there is a problem that the desired voice signal itself is a target of
reduction and the desired voice signal is degraded.
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[0031]
In view of the above, it is an object of the present invention to provide a microphone device using
an adaptive noise canceller and capable of reducing deterioration of a desired audio signal.
[0032]
SUMMARY OF THE INVENTION In order to solve the above problems, the microphone device
according to the present invention corresponds to the first and second microphones and the first
microphone when the reference numerals of the embodiments described later correspond to
each other. Filter means to which an output signal is supplied, subtraction means for subtracting
the output signal of the filter means from the output signal of the second microphone, and noise
components included in the main input signal adaptively based on the reference input signal And
the characteristic of the filter means is that when the identification signal is incident from the
directivity direction to be created and the sound is collected by the first and second microphones,
And the output signal of the first microphone is used as the main input signal for the adaptive
noise cancellation. Is supplied to the over, characterized in that the output signal of said
subtracting means is supplied to the adaptive noise canceler as reference input.
[0033]
In the present invention of the above construction, the characteristic of the output signal of the
subtraction means is such that it is the lowest sensitivity (zero) to the signal from the directional
direction to be produced, ie, the signal from the desired voice direction. It becomes.
That is, from the subtracting means, a signal equivalent to that picked up by the microphone with
zero sensitivity in the desired voice direction is obtained.
Therefore, it is possible to avoid that the desired voice is mixed in the output of the subtracting
means which is the reference input of the adaptive noise canceller.
[0034]
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT An embodiment of the microphone
device according to the present invention will be described below with reference to FIG.
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This example is an example where the adaptive filter circuit of the adaptive noise canceller has a
digital configuration.
[0035]
In FIG. 1, reference numerals 11 and 12 denote first and second microphones, which in this
example are both nondirectional microphones. Reference numeral 10 denotes an adaptive noise
canceller, and the adaptive filter circuit 5 is composed of the adaptive linear combiner 100 and
the filter coefficient calculation circuit 110 as described above.
[0036]
An audio signal collected and converted into an electrical signal by the first microphone 11 is
converted into a digital signal in the A / D converter 13 and supplied to the main input terminal
1 of the adaptive noise canceller 10 as a main input signal. Be done.
[0037]
On the other hand, an A / D converter 14 converts an audio signal obtained by being picked up
by the second microphone 12 and converted into an electric signal into a digital signal and
supplied to a subtraction circuit 15.
A digital signal of the output of the first microphone 11 from the A / D converter 13 is supplied
to the subtraction circuit 15 through an FIR filter circuit 21 having a configuration similar to that
of the adaptive linear coupler 100.
[0038]
Then, the subtraction circuit 15 subtracts the signal from the FIR filter circuit 21 from the digital
signal from the A / D converter 14, and the subtraction output is supplied to the reference input
terminal 2 of the adaptive noise canceller 10 as a reference input. The output signal of the
adaptive noise canceller 10 is converted back to an analog signal by the D / A converter 16 and
is derived at the output terminal 17.
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[0039]
The filter coefficients of the FIR filter circuit 21 are supplied from the filter coefficient calculation
supply circuit 22. Similar to the filter coefficient calculation circuit 110 of the adaptive noise
canceller 10, the filter coefficient calculation supply circuit 22 can be configured by a DSP. In
addition, the filter coefficient calculation circuit 110 of the adaptive noise canceller 10 can also
be used in common.
[0040]
The value of the filter coefficient is determined as described later, but when the identification
signal is incident from the directivity direction to be created and the first and second
microphones 11 and 12 collect the sound, It is the value of the filter coefficient of the result
identified so that an output may become zero.
[0041]
The digital signal of the output of the first microphone 11 from the A / D converter 13 is
supplied to the filter coefficient calculation supply circuit 22 via the switch SW1, and the output
signal of the subtraction circuit 15 is supplied via the switch SW2. Be done.
The switches SW1 and SW2 are turned on when the filter coefficient to be supplied to the FIR
filter circuit 21 is obtained. Then, after the filter coefficients are obtained, these switches SW1
and SW2 are turned off, and the obtained filter coefficients are fixed as the filter coefficient value
of the FIR filter circuit 21.
[0042]
The operation of the filter coefficient calculation supply circuit 22 and the method of
determining the filter coefficient of the FIR filter circuit 21 will be described. First, the switches
SW1 and SW2 are turned on. Then, the identification signal is made to enter from the directivity
direction to be created as the microphone device of this example, that is, from the desired voice
arrival direction to obtain the highest sensitivity. There is no or negligible incident of other
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signals. In this state, as shown in FIG. 2, the configuration of the front part of the adaptive noise
canceller 10 becomes an adaptive noise canceller 20 that works to cancel the identification
signal as noise.
[0043]
That is, this adaptive noise canceller 20 is composed of a subtraction circuit 15, an FIR filter
circuit 21 and a filter coefficient calculation supply circuit 22, and the main input is the output of
the A / D converter 14 and the reference input is the A / D converter 13. The output of the
subtraction circuit 15 is the output of the subtraction circuit 15. When an adaptive operation is
performed by the adaptive noise canceller 20 in this state, the filter coefficient calculation supply
circuit 22 updates the filter coefficient of the FIR filter circuit 21 so that the output power of the
subtraction circuit 15 becomes zero.
[0044]
Then, the adaptive noise canceller 20 continues the adaptive processing operation for a
predetermined time such that the adaptive processing converges, and turns off the switches SW1
and SW2 when the predetermined time has elapsed. This switching off can be performed
automatically using, for example, a timer. The filter coefficient calculation supply circuit 22
stores the filter coefficient at the time of convergence in the memory. As this memory, a nonvolatile memory including a battery backup RAM can be used.
[0045]
Therefore, after that, the filter coefficients of the FIR filter circuit 21 are fixed to filter coefficients
such that the output of the subtraction circuit 15 becomes zero with respect to the incidence of
the identification signal.
[0046]
This indicates that the directional characteristic of the output of the subtraction circuit 15 is
superdirective such that the sensitivity of the direction of arrival of the identification signal, that
is, the desired voice incoming direction becomes zero.
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That is, the sensitivity of the desired voice arrival direction is equal to that collected by the
superdirective microphone with the minimum sensitivity.
[0047]
In the configuration of FIG. 1, the output of the subtraction circuit 15 is supplied as a reference
input to the reference input terminal 2 of the adaptive noise canceller 10, and the digital signal
of the output of the first microphone 11 from the A / D converter 13 is used as the main input.
Since the signal is supplied to the main input terminal 1 of the adaptive noise canceller 10, the
amount of leakage of the desired voice to the reference input is reduced, the deterioration of the
desired voice is prevented, and unnecessary signals from directions other than the desired voice
arrival direction are obtained. It can be canceled adaptively.
[0048]
Furthermore, in the example of FIG. 1, the identification signal is input from the desired voice
direction, and the switches SW1 and SW2 are turned on and off to perform adaptive noise
cancellation with the desired voice direction as the desired voice direction. It is possible to realize
a superdirective microphone device in which the desired voice direction is the directivity
direction.
[0049]
Also, in the example of FIG. 1, two omnidirectional microphones are used to create
omnidirectionality as the main input and unidirectionality as the reference input. The variation in
characteristics, the difference in sensitivity, and the cost are lower than in the case of using a sex
microphone.
[0050]
However, although using the same nondirectional microphone, there is always some variation in
characteristics or sensitivity due to the influence of a housing or the like.
However, in the above example, since the characteristic difference and the sensitivity difference
of the two microphones 11 and 12 are also collectively corrected, it is not necessary to consider
the variation in the characteristic of the microphone unit.
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[0051]
By the way, in the example of FIG. 1, the output of the subtraction circuit 15 has a characteristic
difference with respect to the output of the single microphone 11 in terms of frequency and
phase.
The adaptive noise canceller 10 compensates for this characteristic difference of the reference
input to form a signal approximating an unnecessary signal (noise signal) in the main input and
works to remove the unnecessary signal, but the reference input In the adaptive filter circuit 5,
the burden is increased by the compensation operation of the characteristic difference as
compared with the case where the signal from the unidirectional microphone is used.
[0052]
The example of FIG. 3 takes this point into consideration, and the output signal of the subtraction
circuit 15 is supplied to the reference input terminal 2 of the adaptive noise canceller 10 through
the equalizer circuit 23 for compensating for the characteristic difference.
The other configuration is completely the same as that of FIG. The equalizer circuit 23 can be
configured by a digital filter.
[0053]
According to this example, since the characteristic differences such as frequency, phase, and
amplitude are compensated by the equalizer circuit 23, the load on the adaptive filter circuit 5 of
the adaptive noise canceller 10 is lightened by that amount, and adaptive noise reduction is
satisfactorily performed. An operation can be performed to realize a highly accurate
superdirective microphone device.
[0054]
For example, when the desired voice direction is determined as in the camera integrated type
VTR, the FIR filter circuit 21 is configured to be supplied with a filter coefficient from, for
example, a non-volatile memory, and it is As shown in FIG. 2, the identification signal is made to
enter from the desired voice direction, and the adaptive operation is performed to obtain the
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optimum filter coefficient, and the obtained optimum filter coefficient is stored in the non-volatile
memory. Can be realized.
[0055]
Further, when the filter coefficient calculation supply circuit 22 is separated into a filter
coefficient register and a filter coefficient calculation circuit, the filter coefficient calculation
circuit part of the filter coefficient calculation circuit 22 can be configured by a microcomputer
as described above. The microcomputer constituting the adaptive noise canceller 10 can have the
role of the filter coefficient calculation circuit.
As described above, the filter coefficient calculation circuit of the filter coefficient calculation
supply circuit 22 operates prior to the use of the microphone device or operates only in the
manufacturing process, so any adaptive noise reduction processing of the microphone device is
performed. Not burdened.
[0056]
The algorithm for updating the filter coefficients is not limited to the above-described LMS
method, and it is needless to say that, for example, a learning algorithm or another algorithm can
be used.
[0057]
As described above, according to the present invention, the outputs of the two microphones are
synthesized through the filter circuit, and the filter coefficient of the filter circuit is made to have
the desired voice direction as the minimum sensitivity. By determining in advance, it is possible
to obtain an output signal of a single directivity that is optimal as a reference input.
Therefore, it is possible to suppress leakage of the desired voice to the reference input, and it is
possible to realize a super-directional microphone device with higher accuracy, with less
deterioration of the desired voice signal.
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