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JP2006279740

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DESCRIPTION JP2006279740
PROBLEM TO BE SOLVED: To suppress the generation of a virtual sound source, ensure
resolution, separate a target sound source from a plurality of sound sources, and stably analyze
the amount of generated sound according to the distance and the fluctuation of sound speed An
intensity measurement device and a sound source position analysis method are provided.
According to a sound source position analysis method, a procedure of calculating a delay sum at
a constant angle pitch for each direction assuming a distance to a sound source, extracting a
direction in which the delay sum is maximal, and setting it as a position of a temporary sound
source The procedure of changing the distance to the temporary sound source by a geometric
progression with constant magnification, calculating the delay sum, and setting the position of
the maximum as the position of the temporary sound source 2 again, constant angle pitch and
distance around the temporary sound source 2 Is changed in a geometric progression with a
constant magnification to calculate the delay sum, and the procedure of setting the maximum
position as the sound source position 3 is performed. In addition, the microphone array is
characterized in that the microphones are arranged such that the distance between the
microphones is gradually increased or decreased in order so that the difference becomes a
constant value. [Selected figure] Figure 7
Sound source specific sound intensity measuring device and sound source position analysis
method
[0001]
The present invention relates to a sound source-specific sound intensity measuring device and a
sound source position analysis method for measuring the noise generation amount of each
machine at a construction site or the like where a plurality of machines operate.
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[0002]
Conventionally, as a method of measuring the sound generation amount for each sound source
when sound is generated from a plurality of sound sources, microphones are disposed near each
sound generation source, and the relationship between the sound source-microphone distance
and the measurement value There is known a method of analyzing the amount of generation for
each sound source (see, for example, Non-Patent Document 1).
However, this method is dangerous in the construction site because the microphone is installed
at a position close to the sound source such as a machine, and the distance between the sound
source and the microphone is short. It had the problem that analysis values differed greatly.
[0003]
Moreover, there are the following as another means to be considered. (1) In the method of using
a physically directional microphone, it is necessary to mechanically control the direction of the
microphone so as to always face the direction of the highest sensitivity, and the mechanism
becomes complicated. (2) In the method based on sound intensity measurement, only the vector
sum of the direction of arrival of the sound is measured, and the intensity for each sound source
can not be measured (see, for example, Non-Patent Document 2). (3) In the method by the
microphone array widely used for sound source search, quantitative sound source is assumed on
the assumption that a large number of sound sources are close and not considering the influence
of leakage from nearby sound sources, etc. It was not suitable for the measurement of strength
(see, for example, Patent Document 1). "Power level calculation of multiple sound sources based
on measured data" Japan Society of Mechanical Engineers No. 97-37 VS Tech'97 New Vibration
and Sound Technology, pp. 90-93 "Sound Intensity Probe" Noise Control Vol. 28, No. 2 (2004)
pp. 75-79 JP-A-10-170333
[0004]
The present invention is conventionally made in view of the fact that there was no one suitable
for quantitative measurement of sound source intensity, and in order to measure the intensity of
each sound source at a position about 5 m away from the sound source, (1) Suppress the
generation of virtual sound source, (2) Secure the resolution, (3) Separate the desired sound
source from multiple sound sources, and stabilize the amount of sound generation according to
the distance and the fluctuation of sound velocity It is an object of the present invention to
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provide a sound source-specific sound intensity measurement device and a sound source position
analysis method for analysis.
[0005]
In order to achieve the above object, according to the sound source-specific sound intensity
measuring apparatus of the present invention, sound pressure data of a plurality of microphones
oriented in a sound source direction and arranged in line in a direction perpendicular to the
sound source direction are AD converted -In a device that measures the sound generation
amount for each sound source by analysis, the procedure for calculating the delay sum at a
constant angle pitch for each direction, assuming the distance to the sound source, extracting the
direction in which the delay sum is maximal The procedure for setting the position of 1, the
procedure for changing the distance to the temporary sound source by a geometric progression
with constant magnification to calculate the delay sum, and setting the position that becomes
maximum as the position of the temporary sound source 2 It is characterized in that the direction
is changed by a constant angle pitch and the distance by a geometric progression of constant
magnification to calculate the delay sum, and the procedure of setting the maximum position as
the sound source position 3 is performed.
Further, the program of the sound source position analysis method of the present invention
calculates the delay sum at a constant angle pitch for each direction assuming the distance to the
sound source to the computer, extracts the direction in which the delay sum is maximal,
Procedure of changing the distance to the temporary sound source in a geometric progression
with constant magnification to calculate the delay sum, and setting the maximum position again
as the position of the temporary sound source 2, direction in the vicinity of the temporary sound
source 2 Is performed by changing the constant angle pitch and the distance in a geometric
progression of constant magnification to calculate the delay sum, and performing the procedure
of setting the maximum position as the sound source position 3. Further, a computer readable
recording medium recording the sound source position analysis program of the present invention
is a computer, means for calculating the delay sum at a constant angle pitch for each direction
assuming the distance to the sound source, and the delay sum becomes maximum. A means for
extracting the direction and setting it as the position of the temporary sound source 1; changing
the distance to the temporary sound source by a geometric progression with constant
magnification to calculate the delay sum, and setting the maximum position as the position of the
temporary sound source 2; In the vicinity of the temporary sound source 2, changing the
direction with a constant angle pitch and distance with a geometric progression with a constant
magnification to calculate the delay sum, and acting as a means for setting the maximum position
as the sound source position 3. . In addition, the microphone array is characterized in that the
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microphones are arranged such that the distance between the microphones is gradually
increased or decreased in order so that the difference between the microphones becomes a
constant value.
[0006]
The present invention exhibits the following excellent effects. (1) In the sound source position
analysis method, the procedure of calculating the delay sum at a constant angle pitch for each
direction, assuming the distance to the sound source, extracting the direction in which the delay
sum is maximal, and setting it as the position of the temporary sound source 1 Change the
distance to the sound source by a geometric progression with constant magnification, calculate
the delay sum, and set the position of the maximum as the position of the temporary sound
source 2 again. The target sound source is separated from the plurality of sound sources by
performing the procedure of changing the delay sum with a constant magnification ratio series to
calculate the delay sum, and setting the sound source position 3 as a sound source position. It is
possible to analyze stably in response to the fluctuation of sound velocity. In addition, the
direction of the sound can be analyzed accurately and the distance to the sound source can also
be analyzed. (2) In the sound source-specific sound intensity measurement device, generation of
a virtual sound source can be suppressed by providing a microphone array in which the distance
between the microphones is gradually increased or decreased in order so that the difference
becomes a constant value.
[0007]
The best mode for carrying out the sound source-specific sound intensity measurement
apparatus and the sound source position analysis method according to the present invention will
be described below based on examples with reference to the drawings.
[0008]
FIG. 1 is a plan view for explaining the arrangement of a sound source and a microphone array.
Sound sources such as a backhoe, an engine generator, a wheel loader, etc. that generate sound
are located at three points of the apex of a triangle. Let each sound source be A sound source 1,
B sound source 2 and C sound source 3. A preamplifier is added to the microphones 4, and each
microphone 4 sets the direction of the sound source (in this case, the direction of the sound
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source 2 is the sound source direction. 21) at intervals of 0.1 m, for example, and arranged in the
direction perpendicular to the sound source direction to form the microphone array 5. The sound
sources 1, 2, 3 and the microphone array 5 are located in the same plane.
[0009]
FIG. 2 is an explanatory view showing the details of the microphone array 5. Each microphone 4
with a preamplifier is connected to an AD converter 6, and the AD converter 6 is connected to a
computer 7. The measured values of the sound pressure from each of the microphones 4 are
recorded in the computer 7 by simultaneous sampling at 24 kHz, and the amount of sound
generation for each sound source is analyzed from this data.
[0010]
FIG. 3 shows the directivity of the microphone array 5 disposed in the direction perpendicular to
the sound source direction shown in FIGS. 1 and 2. In the arrangement of the microphones in
FIG. 1, the sound from the direction opposite to the direction in which directivity is given (the
upper direction in the drawing) is affected, but the sound from the rear is made as small as
negligible.
[0011]
Next, a method of stably analyzing the generation amount of sound for each sound source will be
described. As shown in FIG. 4, in order to generate a sound from a certain sound source 8 and
reach each microphone of the microphone array 5, a time difference due to the distance
difference occurs. Under the ideal condition that the sound velocity is constant and the distance
measurement is accurately performed and the error of the microphone can be ignored, the delay
time is calculated from the positions of the sound source 8 and each microphone 4 and the delay
time is corrected and synthesized. By calculating the delay sum, a sound pressure signal
corresponding to a multiple of the microphone can be obtained. However, there is a need for a
robust algorithm that can cope with the movement of the sound source and the fact that the
measured values include various errors. Here, the delay sum will be additionally described. As
shown in FIG. 1, when the microphones 4 are arranged, sound is generated from the A sound
source, and only three microphones are extracted from the sound collected at a sampling rate of
24 kHz, the sound pressure signals 9, 10 and 11 of each microphone are shown in FIG. There is a
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time difference 12 as shown in FIG. This time difference can be set to a calculated value by
assuming the position of the sound source and the speed of sound. In the calculation of the delay
sum, the sound pressure signal is shifted by the delay time calculated assuming the position of
the sound source, the waveforms are matched as shown in FIG. 6, and the sound pressure signals
of the respective microphones are added. If the delay times of the 21 microphones are accurately
calculated, the delay sum is 21 times the signal. Since the sound pressure fluctuates and takes
positive and negative values, in the present case, the delay sum is the root mean square (time
average) of the sound pressure. Also, the magnitude of the sound is a ratio of the mean square
(time average) of the sound pressure per microphone to the reference value in decibel
conversion.
[0012]
FIG. 7 is a diagram for explaining an algorithm for stably analyzing the generation amount of
sound for each sound source in the present invention, and is composed of a flowchart on the left
and a schematic diagram on the right. i) First, the distance to the sound source is appropriately
set, and the delay sum is calculated on the assumption that the sound source is at the position of
one degree pitch (cal. 1). ii) Next, the position of the maximum value of the delay sum output is
determined, and is used as a temporary sound source (1). iii) Next, change the delay sum in a 1.5fold geometric progression from the lower limit to the upper limit assuming the distance to the
sound source (cal. 2), and re-evaluate the location of the temporary sound source (2) Position. iv)
In the vicinity of the temporary sound source (2), change the direction with a pitch of 0.1 ° and
the distance with a geometric progression of 1.1 times and calculate the delay sum (cal. 3).
Position).
[0013]
8 arranges sound sources of 500 Hz and 100 dB for A sound source 1 and 1000 Hz and 100 dB
for B sound source 2 and 2000 Hz and 100 dB for C sound source 3 in the arrangement of FIG. 1
in the anechoic chamber and generates sound simultaneously from three sound sources Of the
algorithm of FIG. This is an example analyzed using up to 1. It can be understood from FIG. 8 that
the maximum point 13 of the delay sum can be separated for each sound source.
[0014]
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FIG. 9 shows cal. The calculated values of the sound power level are shown using up to three. As
a result of calculating the sound power level for each sound source at the set distance, the error
of the B sound source 2 is 1.8 dB, but the other is 0.1 dB, which is a good result. Further, it is
understood from the analysis of the sound that the error in the direction of the sound source is
within the range of 0 to 0.8 degrees, and accurate analysis is possible. Also, it is understood that
the distance to the sound source can be analyzed although there is an error.
[0015]
FIG. 10 shows the arrangement of FIG. 1 in the semi-echo chamber, simultaneously generating
the pre-recorded sound of the wheel loader from the A sound source 1 to the backhoe and the B
sound source 2 to the motor generator and the C sound source 3 respectively. The acoustic
power levels are 100.9 dB, 95.3 dB and 100.2 dB, and cal. This is an example analyzed using up
to three. It can be seen that although the sound from the B sound source 2 is small and affected
by the sound leaking from both sides, the error is large but analysis is possible. Here, the Acharacteristic effective sound power level is an A-characteristic sound power level calculated in
the same manner as in the case of a stationary noise source from the effective noise level
(equivalent noise level) measured for an irregular and greatly fluctuating noise source For the
definition of the A characteristic effective acoustic power level and the effective noise level, see
"Predictive model of construction noise" ASJ CN-Model 2002 ", Journal of the Acoustical Society
of Japan, Vol. 58, No. 11 (2002), pp. 711-731).
[0016]
However, when the sound source contains many high frequency of 4000 Hz, a virtual sound
source becomes a problem. Next, the method of suppressing the virtual sound source will be
described. FIG. 11 shows an example in which the sound source is arranged at the position of the
A sound source 1 in the arrangement of FIG. 1 and analyzed in an anechoic chamber, the left
figure shows a 2000 Hz sound source analysis, and the right figure shows a 4000 Hz sound
source Analysis is shown. The horizontal direction of the figure represents the direction of the
sound source with reference to the microphone array, and the vertical direction of the figure
represents the delay sum of the sound pressure signals of the microphones. The virtual sound
source 14 is generated in the sound source analysis of.
[0017]
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In FIG. 12, in order to suppress a virtual sound source, the intervals of 21 microphones 4 in the
microphone array 5 are unevenly arranged. Assuming that each microphone is expressed as 4-1,
4-2,... 4-20, 4-21 sequentially from the left side, the distance between the leftmost microphone 41 and 4-2 is d, and the microphone 4- The rightmost interval between 20 and 4-21 is 2d, and the
distance from d to 2d is increased by d / (n-2) in order. However, n represents the number of
microphones. The difference d / (n-2) of the gradually increasing or decreasing interval can
suppress the imaginary sound source even if it is a different value of about 1/2 to 2 times, but
about d / (n-2) is the most effective. is there. Of course, the arrangement intervals of the
microphones may be reversed. The directivity of the microphone array thus arranged is as shown
in FIG. In FIG. 3, in the case of 4000 Hz, a virtual sound source appears in the direction of ± 64
° with respect to the frontal direction (upper side of the figure) having directivity, and sound is
generated in the direction in which no sound is generated. Would be judged as Such a virtual
source further increases as the frequency increases. On the other hand, in FIG. 13, in the case of
4000 Hz, the imaginary sound source is suppressed to −20 db or less, and it can be clearly
identified that this suppressed sound is an imaginary sound source. It can be seen that the
imaginary sound source is suppressed also at 8000 Hz and 16,000 Hz.
[0018]
The principle of suppressing the virtual sound source in the microphone array in which the
microphones are arranged as shown in FIG. 12 will be described below. FIG. 14 shows the
relationship between the wave front 15 of the sound wave and the microphone array 5 when the
wavelength 16 is sufficiently long. Assuming that the sound source direction is 17, the arrival
time of the wave front 15 is shifted by the distance difference shown by 18 There is. The
waveform is shifted so as to be equivalent to the case where the microphone is at the position
indicated by 19 by adding the time of this shift. This process is performed in FIG. When the shift
time is matched with the sound source direction, the delay sum is maximized, but when the
frequency becomes high and the wavelength becomes shorter than the microphone interval, a
wave front 15 is also generated between the microphones 4 as shown in FIG. For this reason, a
phenomenon may occur in which the phases are matched even in the direction of the imaginary
sound source indicated by 20 other than the true sound source direction 17. It is because of this
phenomenon that there are more peaks in FIG. 3 at 4000 Hz than at other frequencies. As the
frequency increases in this manner, the number of imaginary sound sources also increases,
which is a phenomenon that occurs because the microphone spacing is equal. Therefore, it is
conceivable that the phases are matched in the true sound source direction 17 and the phases
are not matched in the direction 20 of the virtual sound source by shifting the microphone
interval by 24 as shown by 23 in FIG. However, if the shift is not appropriate, there is a concern
that the phases will not be completely offset even in an unexpected direction. For example, it is
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possible that several microphones may come to the striped position of the wavefront 15 by
chance. The reference numeral 25 indicates the position of a virtual microphone when the
directivity of the shifted microphone array is directed to the true sound source direction, and the
reference 26 indicates when the shifted microphone is directed to the virtual sound source
Shows the position of a virtual microphone. Therefore, in the present invention, as described
above, the microphone interval is regularly extended from d to 2d. In this arrangement, for
example, even if the first and second microphones are in phase, the third and subsequent phases
are shifted by 360 ° / (n−2) each, so that the phases of the respective waveforms in the delay
sum are aligned. There will be no false sound source.
[0019]
On the other hand, when the sound source contains a large amount of low frequency components
of 250 Hz or less, the effect of the reduction in resolution becomes a problem. Next, a method for
securing the resolution will be described. The resolution is improved by increasing the length of
the microphone array, but due to the practical limit of the array length, the frequency below the
limit frequency is to be removed by the band pass filter. Assuming that the low frequency
component is proportional to the high frequency component of the sound power level for each
sound source, all frequency components are estimated. FIG. 17 shows an example where 21
microphones are arranged at intervals of 10 cm, and a sound source is arranged at the position
of A sound source 1 in FIG. 1 and analyzed in an anechoic chamber. The figure on the right
shows a 250 Hz source analysis. Although the delay sum is included in the range 9 of resolution
± 20 ° in the left 500Hz source analysis diagram and 20 ° resolution can be secured, the
delay sum from the range 27 of resolution ± 20 ° in the right 250Hz source analysis diagram
Is sticking out. For this reason, since resolution is poor at 250 Hz or less, the band pass filter is
used in quantitative analysis.
[0020]
It is a top view explaining arrangement of a sound source and a microphone array concerning an
embodiment of the invention. It is an explanatory view showing the details of the microphone
array concerning an embodiment of the invention. FIG. 3 is a diagram showing the directivity of
the microphone array disposed in the direction perpendicular to the sound source direction
shown in FIGS. 1 and 2. It is a figure explaining that the time difference by a distance difference
arises that sound is generated from a sound source and it reaches each microphone of a
microphone array. It is the figure which showed the example of the sound pressure signal of
three microphones. It is the figure which showed the example of the sound pressure signal which
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made the phase match. It is a figure explaining the algorithm which analyzes stably the
generation amount of the sound according to sound source which concerns on embodiment of
this invention. FIG. 2 is a diagram showing an example of analysis where sound sources of three
sound sources are generated at the same time by arranging sound sources of 500 Hz and 100 dB
for A sound source, 1000 Hz and 100 dB for B sound source and 2000 Hz and 100 dB for C
sound source in the arrangement of FIG. It is the figure which showed the calculated value of the
sound power level. It is the figure which showed the calculated value of A characteristic effective
sound power level of construction machine sound. It is the figure which showed the example
which arrange | positioned the sound source in the position of A sound source in the
arrangement | positioning of FIG. 1, and analyzed in an anechoic room. FIG. 7 is a view in which
the distance between microphones in the microphone array according to the embodiment of the
present invention is increased by d / (n−2) in order from d to 2d. It is a figure which shows the
directivity of the microphone array arrange | positioned like FIG. It is a figure explaining the
relationship of the microphone space | interval and wave face in case a wavelength is long
enough. It is a figure explaining the relationship between the microphone space | interval in the
case where a wavelength is shorter than a microphone space | interval, and a wave front. It is a
figure explaining the example which shifted the microphone interval, in order to avoid generation
| occurrence | production of a virtual sound source. It is the figure which arranged the 21
microphones by 10 cm intervals, arranged the sound source in the position of A sound source of
FIG. 1, and analyzed it in the anechoic room.
Explanation of sign
[0021]
DESCRIPTION OF SYMBOLS 1 A sound source 2 B sound source 3 C sound source 4 microphone
5 microphone array 6 AD converter 7 computer 8 sound source 9, 10, 11 sound pressure signal
of each microphone 12 delay time between each microphone 13 delay sum maximum point 14
imaginary sound source 15 wavefront 16 wavelength 17 true sound source direction 18 delay
distance 19 virtual microphone arrangement by giving delay time 20 virtual sound source
direction 21 virtual microphone arrangement facing the real sound source direction 22 sound
source direction of the virtual sound source Placement of virtual microphones 23 Microphones
shifted so as not to be at equal intervals 24 Deviation amount 25 Positions of virtual
microphones when the direction of the true sound source is turned 26 Positions of microphones
shifted 26 as a virtual sound source Virtual microphone position when oriented 27 resolution ±
20 Range of °
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