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JPH07274280

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DESCRIPTION JPH07274280
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
sound pickup apparatus suitable for use in, for example, a VTR integrated video camera.
[0002]
2. Description of the Related Art The applicant of the present application has previously filed
Japanese Patent Application No. Hei 4-27298 as a sound collection device suitable for use in, for
example, a VTR-integrated video camera etc., for example, collecting stereo sound. See the official
gazette). That is, in this apparatus, good stereo sound collection can be performed by performing
spatial and electrical processing using two microphones provided relatively close to each other.
[0003]
By the way, in such a VCR integrated type video camera etc., when this is used outdoors, for
example, the problem of the peculiar wind noise which generate | occur | produces when a wind
hits a microphone arises. Therefore, for the purpose of reducing such wind noise, the applicant of
the present application has previously proposed Japanese Patent Application No. 3-153341
(refer to Japanese Patent Application Laid-Open No. 5-7392).
03-05-2019
1
[0004]
That is, in this device, in the case of signals collected by two microphones provided relatively
close to each other, it is considered that the correlation of the low range audio signal is high,
while the wind noise is low. it is conceivable that. Therefore, the difference component of the
signal collected by the two microphones is extracted, the peak value of this difference component
is detected, and particularly the low frequency component of the signal collected according to
this peak value is attenuated. It is.
[0005]
Therefore, in this device, wind noise with low correlation is mainly extracted as the difference
component, and the amount of attenuation of the low-frequency component of the collected
signal is controlled according to the peak value of the difference component to obtain wind. The
low frequency component of the picked up signal is attenuated only when there is much noise,
and wind noise is reduced.
[0006]
However, in this device, when there is a lot of wind noise, all low frequency components are
attenuated, so that the original sound signal to be picked up is attenuated.
Also, the fluctuation of the peak value of the wind noise changes sharply, and when the amount
of attenuation is controlled according to this peak value, the amount of attenuation is frequently
controlled, which may cause an auditory sense of unnatural feeling. is there.
[0007]
This application is made in view of such a point, and the problem to be solved is attenuated to
the voice signal which should be originally collected in the conventional device. Also, the amount
of attenuation is frequently controlled, which may cause an auditory sense of unnatural feeling.
[0008]
A first means according to the present invention has two or more microphones 1A and 1B, and
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2
extracts low-frequency difference components of signals collected by these microphones
(computing unit 5 Low-pass filter 6) and subtract this difference component from the signal
collected by each of the microphones (computers 4A and 4B), and the level of the difference
component to be subtracted is envelope detected This is a sound collection device configured to
control (amplifier 7) a signal in which predetermined attack characteristics (characteristic setting
circuit 10) and recovery characteristics (characteristic setting circuit 11) are set to the detected
signal.
[0009]
A second means according to the present invention is the sound collection device according to
the first means, wherein the signal subjected to subtraction of the difference component is
subjected to predetermined adaptive processing to further reduce the wind noise. It is an
apparatus.
[0010]
A third means according to the present invention is the sound pickup apparatus according to the
second means, wherein the adaptive processing supplies the difference component to the
adaptive filters 17A and 17B, and the obtained signal is subtracted from the difference
component. It is a sound collection device which is made to subtract from the low frequency
component of the signal (computers 15A, 15B).
[0011]
A fourth means according to the present invention is the sound collection device according to the
second means, wherein the adaptive processing is performed using a control signal for
controlling the level of the difference component.
[0012]
According to this, according to this, two or more microphones are extracted, and the lowfrequency difference component of the signals collected by these microphones is extracted, and
this difference component is subtracted from the signals collected by the respective microphones
At the same time, by controlling the level of the difference component to be subtracted by a
signal in which predetermined attack characteristics and recovery characteristics are set to the
envelope detection signal of the difference component, the influence on the collected voice signal
is minimized. Only the wind noise can be reduced, and a good sound signal can be collected.
[0013]
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3
Further, in addition to the above-described processing, wind noise can be further reduced by
performing predetermined adaptive processing on the signal from which the difference
component has been subtracted, and in this case, it is picked up by the respective microphones
described above. Since the process of subtracting the difference component from the signal is
performed first, the load of the adaptive process can be reduced, and the sound signal can be
picked up with a better configuration with a simple configuration.
[0014]
DESCRIPTION OF THE PREFERRED EMBODIMENTS In FIG. 1, 1A and 1B show two microphones
provided close to each other, such as a stereo microphone of a video camera integrated with a
VTR.
The signals collected by the microphones 1A and 1B are supplied to the A / D converters 3A and
3B through the amplifiers 2A and 2B, respectively.
Output signals of the A / D converters 3A and 3B are supplied to computing units (Σ) 4A and 4B,
respectively.
[0015]
Further, output signals of the A / D converters 3A and 3B are supplied to the computing unit (Σ)
5, and an output signal of the A / D converter 3B is subtracted from an output signal of the A / D
converter 3A.
The subtraction output of the computing unit 5 is supplied to a low pass filter (LPF) 6.
As a result, in the low pass filter 6, the low-pass difference component of the signal collected by
the two microphones 1A and 1B is extracted.
[0016]
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4
An output signal of the low pass filter 6 is supplied to an amplifier 7.
Further, the output signal of the low pass filter 6 is supplied to the envelope detection circuit
(DET) 9 through the amplifier 8, and the detection signal is supplied to the attack characteristic
setting circuit 10 and the recovery characteristic setting circuit 11.
Then, the output signals of these characteristic setting circuits 10 and 11 are supplied to a
computing unit (Σ) 12 and added to each other.
[0017]
The signal added by the computing unit 12 is supplied to the amplifier 7 and the gain is
controlled by the added signal. Thus, the level of the output signal of the low pass filter 6 passed
through the amplifier 7 is controlled. The output signal of the amplifier 7 is supplied to the
computing unit 4A and subtracted from the output signal of the A / D converter 3A. The output
signal of the amplifier 7 is supplied to the computing unit 4B and added to the output signal of
the A / D converter 3B.
[0018]
Further, the output signals of the arithmetic units 4A and 4B are respectively supplied to low
pass filters (LPFs) 13A and 13B and high pass filters (HPFs) 14A and 14B. The output signals of
these low pass filters 13A and 13B are supplied to computing units (Σ) 15A and 15B,
respectively. Further, the output signals of the high pass filters 14A and 14B are supplied to
computing units (器) 16A and 16B, respectively.
[0019]
Further, the output signal of the low pass filter 6 is supplied to adaptive filters 17A and 17B
described later. Further, subtraction signals to be described later from the computing units 15A
and 15B are supplied to the adaptive filters 17A and 17B, respectively. Further, the addition
signal from the computing unit 12 is supplied to the adaptive filters 17A and 17B. The
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5
coefficients of the adaptive filters 17A and 17B are controlled in accordance with the addition
signal and the subtraction signal.
[0020]
Further, the output signals of these adaptive filters 17A and 17B are respectively supplied to the
computing units 15A and 15B, and are subtracted from the output signals of the low pass filters
13A and 13B. The signals subtracted by these computing units 15A and 15B are respectively
supplied to the computing units 16A and 16B, and are added to the output signals of the high
pass filters 14A and 14B. And this addition signal is taken out to output terminal 18A, 18B,
respectively.
[0021]
Therefore, in this device, the low-pass filter 6 outputs a low-frequency difference component of
the signal collected by the two microphones 1A and 1B provided close to each other, that is, a
low-correlation low-frequency component, ie, wind noise. Taken out. The wind noise is levelcontrolled by the amplifier 7 and is subtracted from the output signals of the original A / D
converters 3A and 3B by the computing units 4A and 4B. Note that, since the output signal of the
A / D converter 3B is subtracted by the computing unit 5, the subtraction is executed by addition
in the computing unit 4B.
[0022]
As a result, in these computing units 4A and 4B, only the component of wind noise among the
signals collected by the microphones 1A and 1B is subtracted, so that the wind can be generated
without attenuating the original sound signal to be collected. Noise can be removed.
[0023]
Further, in this device, the component of the wind noise from the low pass filter 6 is envelopedetected by the detection circuit 9, and the level of the wind noise subtracted by the arithmetic
units 4A and 4B is controlled by this detection signal.
03-05-2019
6
Thus, only when wind noise occurs, wind noise is removed according to the level.
[0024]
Further, attack characteristics and recovery characteristics when controlling the level of wind
noise are set by the setting circuits 10 and 11. That is, for example, at the beginning of wind
noise, in order to improve the response to a gust or the like, the setting circuit 10 is set with an
attack characteristic having a quick response so that the control characteristic follows this rise.
Further, when the level of wind noise becomes small, if the response of control is made fast, the
change point becomes easy to be perceived by audibility, so that the setting characteristic of the
setting circuit 11 is set with the recovery characteristic to make the response slow.
[0025]
Thus, according to this apparatus, two or more microphones 1A and 1B are provided, and the
low-frequency difference component of the signal collected by these microphones is extracted
(computer 5, low-pass filter 6). The level of the difference component to be subtracted is
subtracted from the signals collected by the respective microphones (operators 4A and 4B), and
the attack and recovery characteristics of the difference component envelope detection signal
(detection circuit 9) are calculated. By controlling (amplifier 7) with the signal set (characteristic
setting circuits 10 and 11), the influence on the collected audio signal is minimized and only the
wind noise is reduced, and a good audio signal is collected. Can do.
[0026]
In this apparatus, the attack characteristic and recovery characteristic setting circuits 10 and 11
are, for example, the primary time constant processing method, but in order to further improve
the audibility characteristic, the double time constant processing method is added to this. It can
also be adopted.
This is the above-mentioned response, in particular, changing the time constant corresponding to
the recovery characteristic according to the situation.
[0027]
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7
That is, for example, for a large noise input such as a plosive sound, since the noise input ends in
an instant, it is not preferable for the sense of hearing to extend the recovery time. Therefore, the
recovery time in such a case is shortened, and the response time is changed in accordance with
the polarity of the change of the signal so as to lengthen the recovery time when it enters stably
for a predetermined time or more. By devising the signal processing in this way, it is possible to
obtain an aurally natural response characteristic to a complex level change of wind noise.
[0028]
Furthermore, in this device, it is difficult to completely remove wind noise from the output
signals of the operators 4A and 4B, and if the level of the component of wind noise to be
subtracted by the operators 4A and 4B is increased, the opposite side An influence may be
generated by the output signal of the microphone. Therefore, in this device, the wind noise is
further reduced by further applying an adaptive process to the output signals of the computing
units 4A and 4B.
[0029]
That is, in this device, the low pass filters 13A and 13B have the same characteristics as the low
pass filter 6 described above. On the other hand, the characteristics of the high pass filters 14A
and 14B are opposite to those of the low pass filters 13A and 13B, and when the signals
separated by these filters are added as they are, they are returned to the original signal.
[0030]
Further, the component of wind noise from the low pass filter 6 described above is supplied as a
reference signal to the adaptive filters 17A and 17B, and the output signals of these adaptive
filters 17A and 17B are low pass filters 13A and 13B in the computing units 15A and 15B,
respectively. Is subtracted from the output signal of Further, these subtraction signals are fed
back to the adaptive filters 17A and 17B, respectively. Further, a signal for controlling the level
of wind noise added by the computing unit 12 is supplied to the adaptive filters 17A and 17B.
[0031]
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8
Therefore, these adaptive filters 17A and 17B use the above-mentioned signals to minimize wind
noise components that could not be removed by the processing up to the operators 4A and 4B at
the output of the operators 15A and 15B. Configured As an example of the configuration of such
adaptive filters 17A, 17B, for example, an LMS (Least Mean Square) algorithm as shown in FIG. 2
can be applied.
[0032]
That is, in FIG. 2, the component of the wind noise from the above-mentioned low-pass filter 6 is
supplied to the series circuit of arbitrary delay elements 71, 72. The input and output signals of
these delay elements 71, 72... 7n are supplied to computing units 80, 81. The coefficients W0k,
W1k... Wnk are respectively multiplied by these arithmetic units 80, 81. Further, the output
signals of these computing units 80, 81... 8n are added by the computing unit 90.
[0033]
Further, in this circuit, the coefficients W0k, W1k... Wnk of the arithmetic units 80, 81. The
arithmetic circuit 100 is supplied with the subtraction signal from the arithmetic unit 15 A or 15
B described above and a signal for controlling the level of wind noise added by the arithmetic
unit 12.
[0034]
In this arithmetic circuit 100, for example, Wi (k + 1) = Wik + 2μεk Xk, where k: an absolute
time μ for performing processing: a gain factor that determines the speed and stability of the
adaptation, of the wind noise added by the arithmetic unit 12 It corresponds to a signal that
controls the level. ε k: an error at one clock cycle, which corresponds to a subtraction signal
from the arithmetic unit 15A or 15B. The coefficients W0k, W1k... Wnk of the arithmetic units
80, 81.
[0035]
03-05-2019
9
As a result, from the adaptive filters 17A and 17B, output signals that minimize the components
of wind noise that could not be removed by the processing to the operators 4A and 4B are taken
out at the outputs of the operators 15A and 15B, respectively. The output signals of these
adaptive filters 17A and 17B are supplied to the computing units 15A and 15B, respectively, and
subtracted from the output signals of the low pass filters 13A and 13B. The processing effect can
be enhanced as the number of steps of the delay element increases.
[0036]
Therefore, in this device, the output signals of the adaptive filters 17A and 17B are supplied to
the computing units 15A and 15B, respectively, to be subtracted from the output signals of the
low pass filters 13A and 13B, and the computing unit 4A in the subtracted signals. The adaptive
filters 17A and 17B are controlled to minimize components of wind noise that could not be
removed by the processing up to 4B.
[0037]
In this case, in these adaptive filters 17A and 17B, most of the wind noise is removed by the
processing up to the computing units 4A and 4B, so only the remaining small components need
be removed. Compared with a device in which wind noise removal is performed only by an
adaptive filter, it is possible to perform better wind noise removal in a small-scale circuit.
[0038]
Thus, according to this apparatus, in addition to the process of subtracting the difference
component from the signals collected by the respective microphones 1A and 1B described above,
predetermined adaptive processing (adaptive filters 17A and 17B) is performed on the signal
from which the difference component is subtracted. Can further reduce wind noise, and in this
case, the processing of subtracting the above-mentioned difference components can be
performed first to lighten the load of the adaptive processing, and the configuration is simple.
The sound signal can be picked up more favorably.
[0039]
Furthermore, the signals from which the low-range wind noise components have been removed
by these adaptive processes are supplied to the computing units 16A and 16B, respectively, and
are added to the high-frequency signals from the high pass filters 14A and 14B.
03-05-2019
10
As a result, an audio signal from which wind noise has been removed is taken out from the
signals collected by the respective microphones 1A, 1B at the output terminals 18A, 18B.
[0040]
In the above-mentioned apparatus, the cutoff frequency of the low pass filters 6, 13A, 13B and
the high pass filters 14A, 14B changes its optimum value depending on the microphones 1A, 1B
and the structure around them, but generally 200 to 500 Hz It is effective to select a value
between.
[0041]
Further, in the above-described apparatus, the algorithm applied to the adaptive filters 17A and
17B is not limited to the LMS (Least Mean Square) algorithm described above, and learning
identification method, affine projection algorithm, least squares method, RLS (Recursive Least
Square) Various algorithms such as fast Kalman can be applied.
[0042]
Furthermore, in the above-described embodiment, the signals collected by the microphones 1A
and 1B are digitized by the A / D converters 3A and 3B and processed. However, the invention of
the present application is similarly implemented by analog processing. is there.
However, digital processing is advantageous, particularly when it includes adaptive processing,
and its implementation potential is high.
When digital processing is performed, so-called digital signal processor (DSP) can be used.
[0043]
Further, in the above embodiment, the case of two microphones has been described, but in the
case of using three microphones, it can be performed as shown in FIG.
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11
That is, in FIG. 3, the signals collected by the three microphones 1A, 1B and 1C are supplied to
the A / D converters 3A, 3B and 3C through the amplifiers 2A, 2B and 2C, respectively.
[0044]
Further, for example, the signal collected by the central microphone 1C is supplied to the
computing unit 50 as it is.
The signals collected by the left and right microphones 1A and 1B are attenuated (bit-shifted) by
the 1/2 calculators 51A and 51B and supplied to the calculator 50. Then, these attenuated
signals are subtracted from the signals collected by the central microphone 1C, and the
subtracted signals are supplied to the low pass filter 6 to perform the same processing as
described above. Thus, processing in the case of using three microphones is performed.
[0045]
The signal processing shown in the above-mentioned Japanese Patent Application No. 2-132051
(refer to Japanese Patent Application Laid-Open No. 4-27298) can be carried out, for example, at
any position of the circuit of the embodiment of the present application shown in FIG.
[0046]
According to the present invention, it has two or more microphones, the low-frequency
difference component of the signal collected by these microphones is extracted, and this
difference component is collected by each microphone By subtracting from the signal and
controlling the level of the subtracted difference component with a signal in which
predetermined attack characteristics and recovery characteristics are set in the envelope
detection signal of the difference component, the influence on the collected voice is minimized. It
was possible to reduce the wind noise and reduce the noise to a good level to pick up the audio
signal.
[0047]
Further, in addition to the above-described processing, wind noise can be further reduced by
performing predetermined adaptive processing on the signal from which the difference
component has been subtracted, and in this case, it is picked up by the respective microphones
described above. Since the process of subtracting the difference component from the signal is
performed first, the load of the adaptation process can be lightened, and it is possible to perform
03-05-2019
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better sound signal collection with a simple configuration.
[0048]
Brief description of the drawings
[0049]
1 is a block diagram of an example of a sound collection device according to the present
invention.
[0050]
2 is a block diagram of an example of an adaptive filter applied to the present invention.
[0051]
3 is a block diagram of another example of the sound collection device according to the present
invention.
[0052]
Explanation of sign
[0053]
1A, 1B Microphone 2A, 2B, 7, 8 Amplifier 3A, 3B A / D Converter 4A, 4B, 5, 12 Arithmetic Unit 6,
13A, 13B, 15A, 15B, 16A, 16B Low Pass Filter 9 Envelope Detection Circuit 10 Attack
characteristic setting circuit 11 Recovery characteristic setting circuit 14A, 14B High pass filter
17A, 17B Adaptive filter 18A, 18B Output terminal
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