JP3208424B2 - Surface array multi-channel adaptive phasing method and apparatus - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a group of sensors for transmitting sound waves or vibrations and receiving reverberation sounds or vibrations from a target or receiving radiation sounds or vibrations from a target in water or in the air. The present invention relates to a surface array multi-channel adaptive phasing method and apparatus capable of reducing noise arriving from a side beam direction by being applied to a device such as a sonar having a phase difference.

[0002]

2. Description of the Related Art Underwater or in the air, the output of a receiver for transmitting sound waves and receiving reverberation sound from a target or the output of a receiver for receiving sound radiated from a target is covered with noise existing in the surroundings. . Therefore, in order to find a received signal from a target, it is necessary to perform a phasing process on the output of the receiver to reduce the noise level.

A conventional phasing method and apparatus for a receiver output for transmitting a sound wave and receiving a reverberation sound from a target or a receiver output for receiving a radiation sound from a target in water or in the air is described below. The output of the container was multiplied by a fixed value called a shading coefficient and added.

On the other hand, the LMS (Least Mean Square) algorithm calculates the output by the following calculation.

Assuming that the input vector is X _k and the tap weight is W _k , the output y _k is y _k = W _k ^T X _k (where k: sampling time). Instantaneous error signal epsilon _k expressed by taking the difference between the output _{y k} and the desired response _{d k ε k = d k -y} k = d k -W k T X k. The tap weight is calculated by W _{k + 1} = W _k +2 με _k X _k . μ is a step parameter.

However, since the LMS (Least Mean Square) algorithm reduces the signal output when there is a correlation between the signal output and the noise output, the LMS algorithm is currently used for phasing of sonar in Japan. Not been.

[0007] Frost adaptation processing {O. L. Frost,
“An Algorithm for LinearyConstraind Adaptive Arra
y Processing ”, Proc. IEEE, vol 60, No8, August, 92
In 6-935 (1972) Ｘ, X_kIs the k-th sample,
W_kIs the tap weight for the k-th sample, and k +
The first tap weight is obtained by the following equation and sequentially output y _kCalculate
I do. y_k= W_kX_k W_{k + 1/2}= W_k+ 2μy_kX_k [E_{1 k + 1/2}… E_{L k + 1/2}] = [F− [1
1 ... 1] W_{k + 1/2}] / D

(Equation 1) W _{k + 1} = W _k +2 μy _k X _k + E _{k + 1/2} y _{k + 1} = W _{k + 1} X _{k + 1} where μ is a step parameter, D is the number of elements, L is the number of taps, and f is a filter.

[0008]

There are various methods for selecting a shading coefficient, such as a method for narrowing the width of the main beam and a method for decreasing the sensitivity of the side beam. Is selected. However, these methods all assume that the noise source is located at a sufficiently long distance. For this reason, the optimal shading coefficient has not been obtained for a noise source located at a short distance and a noise source whose existence azimuth and position fluctuates with time, so that the optimal phasing process has not been achieved.

In view of the above, the present invention provides a plurality of sets each including a central receiver and two receivers at both ends, so that the center receiver and the receivers at both ends in each set are arranged. Using the directivity difference and phase difference with the detector, and applying the adaptive processing calculation to this, if the noise source is in the side beam direction, it exists in any direction regardless of the distance. Another object of the present invention is to provide a surface array multi-channel adaptive phasing method and apparatus capable of always selecting an optimal shading coefficient so as to minimize the noise level.

[0010] Other objects and novel features of the present invention will be clarified in embodiments described later.

[0011]

In order to achieve the above object, the invention according to claim 1 of the present invention transmits a sound wave or a vibration and receives a reverberation sound or a vibration from a target, or emits a sound or a vibration from a target. In the phasing method of the sound wave or vibration sensor output that receives a plurality of sets of three sensors arranged at equal intervals on a straight line, a plurality of sets of sensors are arranged.
Pairs of shared sensors share a central sensor
In addition, the adaptive processing calculation is applied to the difference between the reception directivity and the phase difference between the sum output of the two sensors at both ends and the center sensor output in each set to reduce the noise coming from the sidelobe direction. Features.

[0012]

[0013] The present application claims 2 invention, our in claim 1
And a plurality of sets of the sensors are arranged on M (1 ≦ M) straight lines.

According to a third aspect of the present invention, in the second aspect , the M straight lines have a planar arrangement, and angles formed by the straight lines are equal to each other.

According to a fourth aspect of the present invention, in the second aspect , the M straight lines have a planar arrangement, and some or all of the angles formed by the straight lines have different angles.

According to a fifth aspect of the present invention, in the second aspect , N _i (1 ≦ N _i ) sets of the sensors are arranged on the _i- th (1 ≦ i ≦ M) straight line. .

[0017] the invention of claim 6 is in the claim _5, in which the _{N 1 = N 2 = ... =} N i = ... = N M.

According to a seventh aspect of the present invention, in the fifth aspect , some or all of the _Nis are different.

According to an eighth aspect of the present invention, in the fifth aspect , the sensor interval between both ends of the first set on the i-th straight line is _di1 , the sensor interval between both ends of the second set is _di2 , and the jth set. sensor interval _d ij of both ends of, when the sensor distance across the _{N i} th set was _{d iNi, d i2 = 2 ×} d i1,
d _i3 = 3 × d _i1 ,..., d _ij = j × d _{i1,.
It is} an integral multiple of the sensor interval between both ends of the first set, such as _iNi = N _i × d _i1 .

According to a ninth aspect of the present invention, in the fifth aspect , when the sensor interval between the two ends of the j-th set on the i-th straight line is d _ij (1 ≦ j ≦ N _i ), the sensor interval d A part or all of _ij is different from an integral multiple of the sensor interval at both ends of the first set.

According to a tenth aspect of the present invention, in the fifth aspect , the distance between the sensors at both ends of the first straight line is d ₁₁ ,
If the sensor interval at both ends of the second straight line is d ₂₁ , the sensor interval at both ends of the i-th straight line is d _i1 , and the sensor interval at both ends of the M-th straight line is d _M1 , d ₁₁ = d ₂₁
=... = D _i1 =... = D _M1 .

According to an eleventh aspect of the present invention, in the fifth aspect , the distance between the sensors at both ends of the i-th straight line is set to d.
_i1 (1 ≦ i ≦ M), the sensor interval d
A part or all of _i1 is provided at different intervals.

The twelfth aspect of the present invention is directed to the first aspect.
11. In any one of 11., the adaptive processing calculation is performed so that the sum of the value of 0.5 times the sum of the outputs of the two sensors at both ends in the pair of sensors and the output of the central sensor is minimized. .

The invention of claim 13 of the present application is the invention of claim 12
In the above, the calculation of the adaptation processing is performed using an LMS algorithm.

The invention of claim 14 of the present application is the invention of claim 13
In the above, the calculation of the adaptation processing is performed by using the frost adaptation processing.

The invention of claim 15 of the present application is the invention of claim 13
In the above, when the adaptation processing calculation is performed using the LMS algorithm, two ends at both ends in the sensor set are used.
By keeping the sum of the tap weights for the added outputs of the two sensors and the tap weights for the central sensor output at one, the gain for the received signal from the target is always kept at one.

[0027] The invention of claim 16 of the present application is the invention of claim 15.
In order to always keep the sum of the tap weight for the added output of the two sensors at both ends in the set of sensors and the tap weight for the central sensor output at 1, the sum of the tap weights is calculated for each adaptation process calculation, The tap weights are divided by this sum to keep the total tap weights at one.

[0028] The invention of claim 17 of the present application is the invention of claim 1.
In 3, 15 or 16 , the tap weights in the calculation of tap weights of the LMS algorithm are both positive and negative, and when the tap weights become negative, the upper limit value of the absolute value of the tap weights is determined. It is.

[0029] The invention of claim 18 of the present application is the above-described claim 1.
In 3, 15, or 16 , when the tap weight in the calculation of the tap weight of the LMS algorithm becomes negative, its absolute value is taken and only a positive numerical value is selected.

According to a nineteenth aspect of the present invention, there is provided a phasing device for transmitting a sound wave or vibration and receiving a reverberation sound or vibration from a target or a sound wave or vibration sensor output for receiving a radiation sound or vibration from a target. , Placed at equal intervals on a straight line
Multiple sensor sets consisting of three sensors
And the set of sensors is a central sensor
And a delay unit that delays the output of each sensor, an adder that adds the delay outputs of the two sensors at both ends of each sensor set, and multiplies each adder output by 0.5. A multiplier and a matching processor that receives the delay output of the central receiver and the output of each multiplier in each sensor set and performs a matching process calculation,
It is characterized in that the reception sensitivity of the side lobe is reduced by the adaptive processing calculation of the adaptive processor to reduce noise.

[0031]

BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a block diagram showing an embodiment of a method and an apparatus for phasing a surface array multi-channel adaptation according to the present invention.

FIG. 1 is an overall configuration diagram of an embodiment of the present invention.
FIG. 2 is a perspective view showing an arrangement of the receivers in the embodiment, FIG. 3 is a perspective view showing an example of the arrangement of the sound sources, and FIG. 4 is a characteristic diagram of a calculation result of the receiving sensitivity by the adaptation processing calculation in the embodiment. .

In FIG. 1 and FIG. 2, a receiver 1 as a sound wave sensor for transmitting a sound wave underwater or in the air and receiving a reverberation sound from a target or receiving a radiation sound from a target.
To 13 are all on the XY plane (that is, in a planar arrangement), as shown in FIG. 2, receivers 2, 1, and 3 are a first set, and receivers 4, 1, and 5 are a second set. , Receivers 6, 1, 7 are a third set, receivers 8, 1, 9 are a fourth set, and receivers 10, 1, 1,
Reference numeral 1 denotes a fifth set, and receivers 12, 1 and 13 denote a sixth set of receivers (a set of three receivers that perform three-channel adaptive phasing). . The receiver 1 is a receiver common to each set, and is located at the center of the receiver at both ends of each set.

FIG. 2 shows the first set and the first set for the sake of simplicity.
4 sets, 2nd set and 5th set, 3rd set and 6th set of receivers
Although the case where they are on three straight lines is illustrated,
Typically, there are M (1 ≦ M) straight lines, and i (1 ≦ i ≦ M)
N on the straight line_i(1 ≦ N _i) When there are receivers
think of.

Line 14 is the distance between receiver 2 and receiver 3, line 1
5 is the distance between the receiver 4 and the receiver 5, line 16 is the distance between the receiver 6 and the receiver 7, line 17 is the distance between the receiver 8 and the receiver 9, and line 18 is the receiver A line 19 indicates an interval between the receiver 10 and the receiver 11, and a line 19 indicates an interval between the receiver 12 and the receiver 13.

Normally, the angles 21 formed by the straight lines are all equal. However, even if the angles cannot be made equal to each other, for example, an array arranged in a lattice having different vertical and horizontal intervals, and even if there is a particular purpose, they are not equal. No problem. When the angles formed by the M straight lines are all equal, the angle formed by the straight lines is π / M.

[0037] If the i-th set of straight line on the wave receiver is a N _i number, the receivers spacing at both ends of the first set on the straight line of the i-th across the d _{i1, 2} pair in receiving if the duplexer spacing _{d i2,} j-th set receivers interval _d ij of both ends of the receivers spacing across the _{N i} th set as _{d iNi, d i2 = 2 ×} d i1, d
_{i3 = 3 × d i1, ...} , d ij = j × d i1, ..., d i
_Although it is set to an integral multiple of the receiver interval of the first set such as _Ni = N _i × d _i1 , there is no problem even when the integral multiple cannot be used and when there is a special purpose and the integral multiple is not used.

In the set of receivers on M straight lines,
The receiver interval at both ends of the first straight line is d ₁₁ , the receiver interval at both ends of the second straight line is d ₂₁ , the receiver interval at both ends of the i-th straight line is di ₁ , and the M-th straight line is Assuming that the receiver spacing at both ends is d _M1 , usually d ₁₁ = d ₂₁ =... = D
_{i1 =} ... ₌ but the d _M1, trouble even if you do not equally if there is a case and, in particular, the purpose can not be equal not.

The number N _i of sets of the receivers on M straight lines
Is usually _{N 1 = N 2 = ... =} N i = ... = but the _{N M,}
There is no problem even if they cannot be made equal, or even if there is a particular purpose and they are not made equal.

Each receiver may be arranged on a curved surface. In this case, the delay is delayed by a delay unit, and each receiver is replaced with an arrangement on a plane.

In the overall configuration diagram of FIG. 1, the sound waves received by the receivers 1 to 13 are delayed by the delay units 31 to 43 and the main lobe (main beam) is directed in a required direction. For the sake of simplicity, all the receivers are on a plane, and the delay amount of each delay unit is assumed to be zero. As a result, the apparatus of this embodiment has a main lobe in the Z-axis direction in FIG. Has side lobes (side beams). First
The outputs of the delay units 32 and 33 corresponding to the receivers 2 and 3 at both ends of the second set of receivers are added by the adder 44 to the receivers 4 at both ends of the second set of receivers. The outputs of the delay units 34 and 35 corresponding to the receivers 5 and 5 are added by an adder 45 to the delay units 3 corresponding to the receivers 6 and 7 at both ends in the third set of receivers.
The outputs of 6 and 37 are added by an adder 46 to delayers 38 and 3 corresponding to the receivers 8 and 9 at both ends in the fourth set of receivers.
The output of 9 is added by an adder 47 to delay units 40 and 41 corresponding to the receivers 10 and 11 at both ends in the fifth set of receivers.
The outputs of the delay units 42 and 43 corresponding to the receivers 12 and 13 at both ends of the sixth set of receivers are added by an adder 49, respectively. Multiply 0.5 by 50, 51, 52, 53, 54, 55. Then, the output of the delay unit 31 corresponding to the center receiver 1 of each set of receivers and the multipliers 50, 51, 52, 53,
The outputs of 54 and 55 are input to the adaptation processor 56, and an adaptation processing calculation is performed. (Reception of the sum output of the two receivers at both ends and the output of the center receiver in each set having a 3-channel receiver) The adaptation calculation is applied to the difference in directivity and the difference in phase).

In FIG. 3, balls 61, 62 and 63 are XY
The sound source is on a plane, YZ plane, or ZX plane. Angles 67, 68 and 69 are incident angles.

FIG. 4 shows that when M is 2 and the angle 21 formed by the straight line is π /
2, N = 1, the distance 22 between the sound source and the receiver 1 is about 1.3 times the receiver distance d at both ends, and the frequency is [sound speed / (2 × d)] and [sound speed / d]. 9 is a calculation result of reception sensitivity for each incident angle of the device of the present embodiment. Graphs 71 and 72 show the case where the sound source is at 61, graphs 73 and 74 show the case where the sound source is at 62, and graphs 75 and 76 show the case where the sound source is at 63. It can be seen that the apparatus of the present embodiment reduces the reception sensitivity in the side lobe direction.

Note that M is 2, N is 1, and the receivers 1, 2,.
Delay devices 31, 3 in the device composed of 3, 4 and 5
₂ , 33, 34 and 35 outputs x _1k , x _2k ,
x _3k , x _4k , and x _5k are represented by the following equations (1) to (5). _x1k = _a1kcos (ωt + _ψ1k ) (1) _x2k = _a2kcos (ωt + _ψ2k ) (2) _x3k = _a3kcos (ωt + _ψ3k ) ... (3) _x4k = _a4kcos (ωt) _4k ) ... (4) _x5k = _a5kcos (ωt + _ψ5k ) ... (5) (where k: k-th sampling value, _a1k to
a _5k : amplitude component, ω: angular frequency of sound wave, ψ _{1k to} ψ
_5k : phase component) The outputs of the delay units 32 and 33 are added to an adder 44.
The value obtained by multiplying by 0.5 and multiplying by 0.5 by the multiplier 50 is expressed by Expression (6). _x23k = 0.5 × ( _x2k + _x3k ) (6) The value obtained by adding the outputs of the delay units 34 and 35 by the adder 45 and multiplying by 0.5 by the multiplier 51 is represented by the following equation (7). . x _45k = 0.5 × (x _4k + x _5k ) (7) Then, the output values of the delay unit 31 and the multipliers 50 and 51 are input to the adaptation processing unit 56 and the adaptation processing calculation is performed to _{execute the} processing shown in FIG. The result can be obtained.

The adaptation processing calculation can be performed using a calculation method according to claims 16, 17 and 18 or 19 or a frost adaptation processing calculation.

The case where the adaptation processing calculation is performed by using the calculation method according to claim 16, 17 and 18 or 19 will be described below. 3 channels of the result of multiplying 0.5 to the sum output of the two receivers at both ends in the receivers of the sets of wave receiver and y _1, the central receivers output y _2. The adaptation processing output z for each set of receivers in the adaptation processor is z = W ₁ y ₁ + W ₂ y ₂ (8) where W ₁ and W ₂ are adaptation processing algorithms (using the LMS algorithm) , And W ₁
The sum of the W ₂ is selected to be 1. That is, the result of multiplying the added output of the two receivers at both ends by 0.5 (= y
Tap weight _{W 1} and the center of the receiving transducer output to ₁₎ (= y
By always keeping the 1 the sum of tap weights W ₂ for _2), to keep the gain for the received signal from the target always 1.

The error signal ε in the adaptation processing algorithm
Is set so that the output z of the adaptive processing calculation is minimized, and the equation
Assuming (9), ε = −z (9) The tap weights W ₁ and W ₂ can be calculated by the following equation (10a) or (10b). _{W k + 1 j = | W} k j + 2με k y k j | ... (10a) W k + 1 j = W k j + 2με k y k j ... (10b) where the formula (10a) adopts the number of all the tap weights positive If you like to, if equation (10b) is employed in any case the tap weights positive and negative, when determining the tap weights W ₁ j
= 1, and j = 2 when determining the tap weights _{W 2.} In addition, when the tap weight of the equation (10b) is adopted in both positive and negative cases, when the tap weight becomes negative, the upper limit of the absolute value of the tap weight is determined, and the tap weight is set so as not to diverge. Good to do.

The tap weights W ₁ and W ₂ thus calculated
With respect to (the respective convergence values obtained by the LMS algorithm), the received signal from the target arriving with the sound wave incident direction being the main lobe direction is not reduced.
The tap weight is calculated again using the following equation (11) as a constraint condition for always making the sum of W ₁ and W ₂ equal to 1. That is,
The sum of the tap weights is calculated for each adaptation process calculation, and each tap weight is divided by the sum to keep the total of the tap weights at 1 by using equation (11). _{W 'k + 1 j = W} k + 1 j / (W k + 1 1 + W k + 1 2) ... (11) Here, W' _{k + 1 j} are values after rebate, W
_{k + 1 j} is before the rebate (where j = 1,
2).

As described above, by using the tap weights W ₁ and W ₂ set to the optimal values so as to reduce the noise in the side lobe direction, the adaptive processor 56 adaptively phasing each receiver set. Thus, side lobes can be reduced as compared with the conventional addition phasing.

If the sound wave from the side lobe direction is stationary, the tap weight becomes constant after being set to the optimum value, but if it is not stationary, the tap weight changes by finding the optimum value.

According to this embodiment, the following effects can be obtained.

(1) Underwater or in the air, when sound waves are transmitted and a reverberation sound from a target is received or when a receiver output for receiving a radiation sound from a target is phased, the receivers are arranged at equal intervals. A plurality of sets each having a total of three receivers at both ends and a center are provided, and a difference in reception directivity between the sum output of the two receivers at both ends and the output of the center receiver in each set of receivers; By applying the adaptation processing calculation to the phase difference, it is possible to reduce noise coming from the side lobe direction.

(2) By arranging the sets of the receivers in a plane on a plurality of straight lines, a function of suppressing side lobes in various directions can be provided.

(3) Adaptation processing calculation so that the sum of the value of 0.5 times the sum of the outputs of the two receivers at both ends in the set of receivers and the output of the center receiver is minimized. Should be done,
The configuration is not complicated.

(4) The adaptation processing calculation is performed according to claim 16,
The calculation can be performed by using the calculation method 17 or 18 or 19 or the frost adaptation processing.

(5) The adaptation processing calculation is performed according to claim 16,
When using the calculation method of 17 and 18 or 19, the sum of the tap weights for the added outputs of the two receivers at both ends of the set of receivers and the tap weights for the center receiver output is always 1 , The gain for the received signal from the target can always be kept at 1. Specifically, in order to always keep the sum of the tap weight for the added output of the two receivers at both ends in the set of receivers and the tap weight for the center receiver output at 1, The sum of the tap weights is obtained, and each tap weight is divided by the sum to keep the total of the tap weights at one.

The present invention is useful for a phasing process of a sonar device or the like that receives reflection or radiation of a sound wave, and is applicable to a fish finder.

It is also possible to replace a part or the whole of the receiver with a vibration sensor, perform a process using a transfer function (for converting vibration into sound), and perform an adaptive process calculation. .

It is also possible to replace all of the receivers with a vibration sensor, and to perform an adaptive processing calculation on the vibration.

Although the embodiments of the present invention have been described above, it is obvious to those skilled in the art that the present invention is not limited to the embodiments and various modifications and changes can be made within the scope of the claims. There will be.

[0061]

As described above, according to the surface array multi-channel adaptive phasing method and apparatus according to the present invention,
A plurality of sets of receivers including a center receiver and two receivers at both ends are arranged, and a plurality of sets of the sensors are arranged.
Using a difference in directivity and a phase difference between the center receiver and the receivers at both ends in the set of receivers, and applying an adaptive processing calculation to this. Therefore, if the noise source is in the side beam direction, it is possible to always select the optimal shading coefficient that minimizes the noise level regardless of the distance or the direction in which the noise source exists. . As a result, it is possible to improve the noise source suppression effect.

[Brief description of the drawings]

FIG. 1 is an overall configuration diagram of an embodiment of a surface array multi-channel adaptive phasing method and apparatus according to the present invention (where M =
3, N = 2).

FIG. 2 is a perspective view for explaining a receiver arrangement in the embodiment (provided that M = 3 and N = 2).

FIG. 3 is a perspective view for explaining a position of a sound source of a sound wave incident on the receiver in the embodiment.

FIG. 4 is a characteristic diagram illustrating an example of a calculation result of reception sensitivity when a sound wave enters from the sound source illustrated in FIG. 3 when M = 2 and N = 1 in the embodiment.

[Explanation of symbols]

1-13 Receiver 14 Spacing between receivers at both ends of first set 15 Spacing between receivers at both ends of second set 16 Spacing between receivers at both ends of third set 17 Receiver at both ends of fourth set Spacing between receivers 18 Spacing between receivers at both ends of fifth set 19 Spacing between receivers at both ends of sixth set 21 Angle between first straight line and second straight line 22 Center receiver and sound source 23 Angle between the projection position of the sound source on the XY plane and the straight line formed by the center receiver and the X axis 24 The angle formed by the sound source and the straight line formed by the center receiver and the Z axis 31 to 43 Delay device 44 to Reference Signs List 49 adder 50-55 multiplier 56 adaptation processor 61 sound source on XY plane 62 sound source on YZ plane 63 sound source on ZX plane 64 distance between receiver 1 and sound source 61 65 receiver 1 66 The distance between the receiver 1 and the sound source 63 67 The angle between the X axis and the line 64 68 The Z axis Angle formed by 65 69 Angle formed between Z axis and line 66 71 Reception sensitivity by incident angle when sound source is on XY plane 72 Reception sensitivity by incidence angle when sound source is on XY plane 73 When reception source is on YZ plane Receiving sensitivity by incident angle 74 Receiving sensitivity by incident angle when the sound source is on the YZ plane 75 Receiving sensitivity by incident angle when the sound source is on the ZX plane 76 Receiving sensitivity by incident angle when the sound source is on the ZX plane

──────────────────────────────────────────────────続 き Continuation of the front page (56) References JP-A-6-18663 (JP, A) JP-A-10-207490 (JP, A) Patent 2723865 (JP, B2) Patent 30000568 (JP, B2) Patent 2720845 (JP, B2) (58) Fields surveyed (Int. Cl. ⁷ , DB name) G01S 3/80-3/86 G01S 5/18-5/30 G01S ⁷ /52-7/64 G01S 15/00- 15/96

Claims (19)

(57) [Claims] 1. A method for phasing a sensor output of a sound wave or a vibration for transmitting a sound wave or a vibration and receiving a reverberation sound or a vibration from a target or receiving a sound or a vibration radiated from a target, comprising the steps of: A plurality of sets of sensors consisting of three sensors arranged in a plurality of sensors, wherein the plurality of sensors are arranged
All of the sets share the central sensor, apply the adaptive processing calculation to the difference between the reception directivity and the phase difference of the two sensors at both ends and the center sensor output in each set, and apply the sidelobe direction. Surface array multi-channel adaptive phasing method, characterized in that noise coming from the array is reduced. 2. The method according to claim 1, wherein a set of the plurality of sensors is M (1 ≦ M).
2. The surface array multi-channel adaptive phasing method according to claim 1, wherein the phasing method is arranged on a straight line. 3. The surface array multi-channel adaptive phasing method according to claim 2, wherein the M straight lines are arranged in a plane, and angles formed by the straight lines are equal to each other. 4. The surface array multi-channel adaptive phasing method according to claim 2, wherein the M straight lines have a planar arrangement, and some or all of the angles formed by the straight lines are different. 5. The surface array multi-channel adaptive phasing method according to claim 2 , wherein N _i (1 ≦ N _i ) sets of sensors are arranged on the i-th (1 ≦ i ≦ M) straight line. 6. _{N 1 = N 2 = ... =} N i = ... = N M a surface array multichannel adapted phasing method according to claim 5, wherein the. 7. The surface array multi-channel adaptive phasing method according to claim 5, wherein some or all of the _Nis are different numbers. 8. i-th straight line of the first set of sensor spacing d _{i1 ends,} the _second set of sensors intervals across the sensor spacing d _{i2, j-th} group at both ends d _ij, N _i sets when the sensor interval of both ends of the eye was _{d iNi, d i2}
= 2 × d _i1 , d _i3 = 3 × d _i1 ,..., D _ij = j ×
_6. The surface array multi-channel adaptive phasing method according to claim 5 , wherein _{di 1} ,..., _{di Ni} = N _i × d _i1 is an integral multiple of the sensor interval between both ends of the first set. 9. When a sensor interval between both ends of a j-th set on an i-th straight line is d _ij (1 ≦ j ≦ N _i ), a part or all of the sensor interval d _ij is set to both ends of the first set. 6. The surface array multi-channel adaptive phasing method according to claim 5, wherein the interval is different from an integral multiple of the sensor interval. 10. The sensor interval at both ends of the first straight line is d ₁₁ , the sensor interval at both ends of the second straight line is d ₂₁ ,
When the sensor interval at both ends of the i-th straight line is d _i1 and the sensor interval at both ends of the M-th straight line is d _M1 , d _{11
= D 21 = ... = d i1} = ... = d M1 to the surface array multichannel adapted phasing method according to claim 5, wherein the. 11. When the sensor interval at both ends of the i-th straight line is d _i1 (1 ≦ i ≦ M), the sensor interval d
_6. The surface array multi-channel adaptive phasing method according to claim 5 , wherein a part or the whole of _i1 is set at different intervals. 12. As the sum of 0.5 times the value and the center of the sensor outputs of the two sum of the sensor output at both ends in the set of the sensor is minimized, claim 1 for adaptation process calculation 11 The phase-matching method for three-channel array according to any one of the above. 13. The surface array multi-channel adaptive phasing method according to claim 12 , wherein the adaptive processing calculation is performed using an LMS algorithm. 14. The surface array multi-channel adaptive phasing method according to claim 13 , wherein the adaptive processing calculation is performed using frost adaptive processing. 15. When the adaptation processing calculation is performed using an LMS algorithm, the sum of the tap weight for the added output of the two sensors at both ends in the set of sensors and the tap weight for the central sensor output is always set to 1. 14. The surface array multi-channel adaptive phasing method according to claim 13 , wherein the gain for the received signal from the target is always kept at 1 by keeping. 16. In order to always keep the sum of the tap weights for the added outputs of the two sensors at both ends in the set of sensors and the tap weights for the central sensor output at 1, the sum of the tap weights is calculated for each adaptation process calculation. 16. The surface array multi-channel adaptive phasing method according to claim 15 , wherein each tap weight is divided by the sum to keep a total of tap weights at one. 17. The calculation of tap weights of the LMS algorithm, wherein both positive and negative numerical values are selected as the tap weights, and when the tap weights become negative, an upper limit value of the absolute value of the tap weights is determined. 13, 15 or
Is a surface array multi-channel adaptive phasing method according to item 16 . 18. The calculation of the tap weights of the LMS algorithm, if the tap weight is negative, claim that takes the absolute value, selects only positive numbers 13, 1
17. The surface array multi-channel adaptive phasing method according to 5 or 16 . 19. The phasing device sensor output of acoustic or vibration receiving the radiated sound or vibration from or targeted to receive the reflected sound or vibration from the ultrasonic or outgoing Goal vibration, equally spaced on a straight line Consisting of three sensors
A plurality of sensor sets and a plurality of the sensor sets
Share the central sensor, delay the outputs of each sensor, add the delay outputs of the two sensors at each end of each sensor set, and add 0 to each adder output. .5, and an adaptation processor that receives the delayed output of the central receiver and the output of each multiplier in each set of sensors and performs an adaptation processing calculation. A plane array multi-channel adaptive phasing apparatus characterized in that the reception sensitivity of side lobes is reduced by calculation to reduce noise.