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JPH05111091

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DESCRIPTION JPH05111091
[0001]
TECHNICAL FIELD The present invention relates to a sound collector such as a microphone.
[0002]
BACKGROUND OF THE INVENTION Microphones are used to collect acoustic signals. In general,
it is said that the frequency characteristic of the microphone is flat, and it is better to be able to
record sound as it is. However, such an ideal characteristic is not necessarily obtained, and the
characteristic is changed and corrected as necessary. In order to change the characteristics of the
microphone, as shown in FIG. 13, an electric circuit such as a filter 33 is inserted in the
subsequent stage of the microphone 31 via the amplifier 32 so that the filter corrects the
characteristics of the microphone. Was common. When it is sufficient to correct only the
characteristics of the microphone, the characteristics of the filter may be fixed according to the
microphone, but in the correction including the measurement environment, the characteristics
must be freely changed. However, for analog signals, it is necessary to change the constants of
capacitors and coils in order to arbitrarily change the filter characteristics, which is troublesome,
and even if you want complex characteristics, design it easily I could not.
[0003]
In particular, it has been difficult for general users of such products to produce low tones, high
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tones, and free adjustment. There is also an equalizer-like device to simplify this. However, since
it is a set of several filters, it becomes large-scale. For example, when recording in the field using
a handy type recording device, it will be understood that it is a large-scale, considering that an
equalizer is provided after the microphone.
[0004]
It is known that if the same is processed with a digital signal, desired characteristic conversion
can be easily performed as compared to an analog signal. As shown in FIG. 14, the output of the
microphone 41 is converted to a discrete signal by the A / D converter 43 through the amplifier
42, and the digital filter is operated by a digital signal processor (DSP) 44 or the like, or It is not
very difficult to obtain the characteristics of an arbitrary filter if the operation of convolution
with an arbitrary impulse response waveform is performed. However, if the band frequency
becomes wide, the number of filters increases, or the number of elements of the microphone
increases, A / D conversion may not be in time, and a plurality of DSPs may be required. As a
result, the peripheral device of the microphone becomes large, and as a result, the cost is also
very high. In addition, there is also a problem that in order to be usable for any audio equipment,
the result calculated on the digital signal has to be restored to the analog signal again.
[0005]
[Purpose] The present invention has been made in view of the above-mentioned circumstances,
and it is possible to convert an output signal of a microphone into a free characteristic with an
analog operation as it is and by simple calculation, and a special digital signal. It is an object of
the present invention to provide a microphone adapted to be converted into free characteristics
without using parts.
[0006]
According to the present invention, in order to achieve the above object, (1) a plurality of sound
receiving elements arranged in a line, and a multiplication unit which multiplies an output of
each of the sound receiving elements by a predetermined constant. And an integration unit for
integrating the outputs of the multiplication units, wherein the output of the integration unit
stage is used as an output of a microphone for detecting an acoustic signal, and (2) an amplifier
for performing the operation of applying the constant (3) that the constant can be varied from
the outside, and (4) that, in (2), the amplification factor of the amplifier can be varied from the
outside, and (5) And (6) in (1) or (2), the constant or amplification factor multiplied to each
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element of the sound receiving element is a part of an impulse response waveform of a desired
system, and further, (6) in (1) , Hold some set of constants (A) and (b) having a portion for
selecting one of them and selecting one of them, and the selected pair is multiplied by the output
of each sound receiving element as a constant and summed up, or (7) arranged in a row A
plurality of sound receiving elements, a multiplication unit that multiplies the output of each of
the sound receiving elements by a predetermined constant, an integration unit that integrates the
outputs of the multiplication units, a waveform input from the microphone A storage unit for
storing a fixed amount and a calculation unit for Fourier transform provided in a part of the
storage unit, the data stored in the storage unit is subjected to an operation including the Fourier
transform, and the result is Further, (8) in (7), it comprises: an input unit for inputting a
frequency characteristic; and a Fourier transform unit for Fourier transforming a frequency
characteristic inputted to the input unit; Based on the numerical value obtained as a result of
conversion by the conversion unit (9) A plurality of sound receiving elements arranged in a line,
and (9) a multiplication unit which multiplies an output of each of the sound receiving elements
by a predetermined constant An integration unit that integrates outputs of the multiplication
units; a first storage unit that stores a waveform input from a microphone; a second storage unit
that stores a preset waveform; And a calculation unit for performing Fourier transform of the
waveform stored in the storage unit, and using the result of calculation using the waveform
stored in the second storage unit as the coefficient, and (10) In the above (9), the number of the
sound receiving elements is determined to be the n-th power of 2 (n is an integer), or (11) a
plurality of sound receiving elements arranged in a line; A multiplication unit for multiplying the
output of each of the sound receiving elements by a predetermined constant; The integration unit
that integrates the outputs of each calculation unit, the storage unit that stores the waveform
input from the microphone, the input unit that inputs the frequency characteristic, and Fourier
transform of the frequency characteristic input to the input unit It is characterized by comprising
a Fourier transform operation unit, performing Fourier transform on the waveform stored in the
storage unit, and using the result of operation using the waveform obtained by Fourier
transforming the input characteristic as the coefficient It is.
Hereinafter, the present invention will be described based on an embodiment of the present
invention.
[0007]
FIG. 1 is a block diagram for explaining an embodiment of a microphone according to the present
invention, in which 11 to 1n are multipliers, m1 to mn are microphones, a1 to an are constants,
and 2 is an integrator. A plurality of sound receiving elements are arranged in a line, and the
outputs of the respective elements are added to each other by multiplying the outputs of the
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elements by a fixed constant. First, the n sound receiving elements m1, m2,..., Mn are arranged in
a line, and the interval between them is d. Assuming that the sound velocity is c, the sound wave
from the front is propagated between the elements with a time delay of d / c. Assuming that the
waveform input from the front of this microphone array is f (t), the waveform received by each
microphone is f (t- (i-1) d / c). However, if i = 1, 2,..., An is expressed as a (i), the signal g (t) finally
obtained from this microphone is
[0009]
It turns out that this is a convolution of discrete data. Therefore, the characteristics can be
changed without converting the sound wave into a digital signal according to the setting method
of a (i). By the way, as in the present invention, it is already known that the sound receiving
elements are arranged in a line to form a microphone. However, all of them aim to sharpen the
directional characteristics of the microphone, and none of them aim to change the acoustic
characteristics as described here. Japanese Patent Application Laid-Open No. 59-70097, for
example, is similar to the present invention in construction. This is as shown in FIG. 2, in which
the time delays of the delay elements D1, D2,... Are given to the respective sound receiving
elements arranged in a line, and then the respective outputs are summed. At this time, the delay
elements D1, D2,... Are propagation times of the sound waves from the respective elements to the
element farthest from the sound source. By this, all the signals are summed in the same phase as
the element farthest from the sound source, and the directivity of the front of the microphone
becomes strong. As will be apparent from the following, the invention described herein is
different in configuration and purpose.
[0010]
FIG. 3 shows another embodiment (claim 2) of the microphone according to the present
invention. The same effect as multiplication can be achieved using an amplifier. The outputs of
the microphones are respectively input to an amplifier, and the gain of this amplifier is set in the
same manner as a in FIG. At this time, it is clear that the obtained result is equal to the equation
(1). This is because, since the time delay between the sound receiving elements corresponds to
the −1 power of z in digital signal processing, for example, if two elements are used to set a1 =
1.0 and a2 = −0.95, plane waves A high-frequency-enhanced microphone for As apparent from
equation (1), when a (i) is the impulse response of the system, the characteristics of the system
can be added to the propagated sound wave. Therefore, in the microphone of the present
invention, the constant or amplification factor in each element is made to be a part of the impulse
response waveform of the desired system. Next, calculation results when impulse response
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waveforms are assigned to the coefficients in the embodiment shown in FIG. 1 and FIG. 3 are
shown. This example is to obtain a microphone of a characteristic that is not susceptible to
environmental noise. In general, environmental noise is often less than 500 Hz. Therefore,
consider a microphone whose characteristic is to reduce 500 Hz or less. FIG. 4 is an impulse
response waveform of a system that reduces 500 Hz or less. The first 32 points were sampled to
obtain the coefficients of the respective sound receiving elements. The frequency characteristic
of the result collected by such a microphone is shown in FIG. This is obtained by arranging 32
sound receiving elements at 1.7 cm intervals, and using the coefficient of a shown in Table 1.
However, the frequency characteristic of each element is flat. As expected, the sensitivity of the
low region is lowered, demonstrating the effectiveness of this method.
[0012]
Next, in order to allow the user to easily set the characteristics of the microphone, several types
of impulse response waveforms, such as low-pass emphasis and high-pass emphasis, are stored,
and any one of them is stored. I chose to choose. FIG. 6 is a view showing still another
embodiment (claim 6) of the microphone according to the present invention, in which 41 to 4 n
are A / D converters. The memory 3 stores sampled data of several types of impulse response
waveforms, and the selector selects one of them. It is loaded into a1-an and becomes a
microphone of specified characteristics. However, according to the present invention, if it is
possible for the user to freely set each constant, it is not only possible to select from given ones,
but not only any characteristics can be obtained, but various characteristics can be recorded. You
can also aim for playful effects such as
[0013]
FIG. 7 is a view showing still another embodiment (claim 4) of the microphone according to the
present invention. The amplification factor was made to be able to differ from the outside. In the
configuration shown in FIG. 3, a coefficient setter 5 is provided so that the amplification factor of
each ampule can be changed externally with a volume or the like. If the user turns the respective
knobs to set the impulse response waveform, the microphone will have characteristics according
to that. For visual effects, the volume is better to slide.
[0014]
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Although the foregoing has been described based on analog signals, it goes without saying that
similar effects can be expected with digital signals. FIG. 8 is a view showing still another
embodiment (claim 3) of the microphone according to the present invention. Input each
coefficient with ten keys 6 and return key. The other operations are the same as in FIG.
Furthermore, for example, when it is necessary to correct the acoustic characteristics of a certain
room for sound recording, first determine the room characteristic, calculate the impulse response
of the inverse characteristic thereof, and give it to this microphone as a constant. It does not. In
order to avoid such an inconvenience, the microphone is provided with a waveform storage unit
for holding an impulse response and an operation unit of Fourier transform.
[0015]
FIG. 9 is a view showing still another embodiment (claim 7) of the microphone according to the
present invention, in which 7 is a memory, 8 is a Fourier transform operation unit, and 9 is cpu.
First, a pulse sound is generated where there is a sound source in the room, and the sound is
collected by the sound receiving element at the head of the microphone placed at the sound
receiving position. This does not need to be at the beginning, but it is sufficient to output one
element instead of using all the elements. It is taken into the memory 7, and about 64 points
from the head of the impulse response waveform are taken out, and Fourier transform is
performed by the Fourier transform operation unit 8 to obtain the inverse characteristic thereof.
This is further subjected to inverse Fourier transform to return to the impulse response
waveform, and the obtained data is given as a1, a2,. As a result, it can be used as a collector of
the ideal characteristic which corrected the characteristic of the room.
[0016]
Further, in the embodiment shown in FIG. 9, when the inverse characteristic is obtained, it is
possible to obtain an effect as if the sound was recorded in a concert hall or a church other than
the characteristic. FIG. 10 shows still another embodiment (claim 9) of the microphone according
to the present invention, in which 10 is a multiplier and 11 is a second memory. Parts different
from FIG. 9 will be described. The second memory 11 stores the frequency characteristics of the
hall, the church and the like. As described above, the inverse characteristic of the room obtained
by the Fourier transform is selected from the second memory 11 and the characteristic is
multiplied by the multiplier 10 to obtain a new characteristic. This is returned to the Fourier
transform operation unit 8 and inverse transformed to obtain an impulse response waveform.
The way of giving it as the coefficient of each sound receiving element thereafter is the same as
that shown in FIG. In this embodiment, the second memory 11 is shown to store frequency
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characteristics. However, even if the impulse response waveform is stored here, conversion to the
frequency characteristics can be simplified.
[0017]
As mentioned above, if the method is selected by the user from among the given constants, there
are limits to the characteristics that can be made, and some users may need to obtain arbitrary
characteristics. Since it is necessary to set each element as a constant in the form of impulse
response, not everyone can determine the constant. Therefore, it is a microphone, in which a
plurality of sound receiving elements are arranged in a line, and outputs of the respective
elements are multiplied by a fixed constant and added together, and means for inputting desired
frequency characteristics and Fourier transform Each of the above constants was determined
based on the resulting numerical values.
[0018]
FIG. 11 is a view showing still another embodiment (claim 8) of the microphone according to the
present invention, in which 12 is a characteristic input device and 13 is a Fourier transform unit.
The user moves the sliding volume of the characteristic input device 12 to create the frequency
characteristic that he / she desires. Thereafter, the Fourier transform unit 13 calculates an
impulse response waveform for obtaining this frequency characteristic, and sets it as a coefficient
of each element. Thus, the user can change the microphone to the specified characteristics
without the need for extra knowledge. However, although the characteristics of the microphone
can be changed, if the room where the sound is collected is narrow or the reflection is large, the
influence of the characteristics of the room may be significant and the microphone may not have
the desired characteristics. Therefore, a microphone for detecting an acoustic signal, in which a
plurality of sound receiving elements are arranged in a line, and the outputs of the respective
elements are multiplied by a constant and added together, and the waveform input from the
microphone Has a storage portion for storing the data, a means for inputting a frequency
characteristic, and a function for performing a Fourier transform, and a waveform obtained by
Fourier transforming a waveform stored in the storage portion and Fourier transforming the
input characteristic The result calculated using is used as the coefficient.
[0019]
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FIG. 12 shows yet another embodiment (claim 11) of the microphone according to the present
invention. In the example shown in FIG. 10, the characteristic input unit 12 shown in FIG. 11 is
added in place of the second memory 11, and the operation is the same as that in FIG. That is,
since the desired characteristic and the inverse characteristic of the room are multiplied and the
impulse response is obtained from the result, the room characteristic can also be corrected to
realize the desired microphone. In general, Fourier transform is generally used widely because it
is less computationally efficient to obtain by FFT (Fast Fourier Transforom). The feature of this
method is that the number of data must be made to be n to 2 (n is an integer). In addition, since
the result calculated by this method is also 2 n pieces of data, if the number of sound receiving
elements is not matched with this, there is a risk that data may be excessive or insufficient.
Therefore, in the microphone of the present invention, the number of sound receiving elements is
determined to be 2 to the nth power (n is an integer). As a result, the calculation result using the
FFT is not wasted, and conversely, the amount of calculation can be reduced.
[0020]
As is apparent from the above description, according to the present invention, it is possible to
change the frequency characteristic with only the microphone, and to realize a sound collecting
apparatus which does not have large parts and high price.
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