Virtual receiving array amplitude increase method for distance determination of underwater line array sound sources

Abstract

The present invention relates to the field of underwater acoustic technology, and particularly relates to a method for amplifying a virtual receiving array for determining the distance of a linear array underwater sound source, including determining the positions of each array element on the physical receiving array according to the structure of the physical receiving array; creating a corresponding virtual receiving array located above the water according to the positions of each array element on the physical receiving array located underwater; the physical receiving array collects and receives the target sound source; segmenting the signals received by the physical receiving array to create signals received by the virtual receiving array; performing beamforming processing on the signals received by the physical receiving array and the virtual receiving array; and obtaining the relative position between the target sound source and the receiving array according to the processing results. The present invention utilizes the characteristic that the position of the target sound source and the position of the virtual sound source generated by the reflection of the water surface are symmetric about the water-air interface, and folds the energy of the virtual sound source with the target signal through signal processing to achieve the effect of amplifying the sound source level of the target sound source.

Description

Technical Field

[0001] The present invention relates to the field of underwater acoustic technology, and particularly relates to a method for increasing the amplitude of a virtual receiving array for distance measurement of a linear array underwater sound source. Background Art

[0002] With the progress of sonar technology and its application and popularization in the environment of military-civilian integration, while the detection ability of underwater sonar is improving, its structural composition is becoming increasingly complex. From single transducers to multi-transducers, from omnidirectional transmission to beamforming, the technical composition of the software and hardware of sonar transmission systems is also becoming more diverse. While the function is improved, the performance of sonar in different environments is more likely to change. First of all, the performance of sonar transducers themselves will change over time, and current sonar equipment generally uses multi-element arrays. When sonar works, it is necessary for each element in the array to cooperate with each other to meet the performance requirements. In this case, even if the performance change of each element transducer is very small, the superposition of the performance changes of the element transducers may cause an obvious change in sonar performance. Secondly, after the transducer is installed on the acoustic array frame in the sonar fairing of the submarine, due to the difference between the actual working environment of the sonar and the theoretical design and acceptance measurement environment, the design parameters and acceptance measurement parameters will be different from the performance parameters during actual use. In addition, affected by the harsh marine environment and other structures at the installation position on the submarine, the transducer installed in the sonar fairing of the submarine is affected by the scattered reflection of the array frame and the internal structure of the fairing; at the same time, after a period of time, the acoustic performance of the element will decline. When the underwater acoustic components of the sonar equipment are damaged or fail and are replaced with transducer elements with good performance, the performance parameters of the sonar equipment will also change. The above factors affecting sonar performance all occur after the sonar equipment has been installed and used for a period of time, which has led to...

[0003] Among them, the transmitting source level of sonar is an important technical index for evaluating the performance of active sonar. Accurately calibrating the transmitting source level of sonar is of great significance for China's national defense construction, marine environment development, etc. According to the definition of the source level in acoustics, to accurately calibrate the transmitting source level, it is first necessary to accurately measure the distance between the transmitting sound source and the receiving transducer (a standard hydrophone in underwater acoustics, and the hydrophone will be used hereinafter).

[0004] However, in the actual measurement process, it is often affected by the small transmitting source level of the target sound source and the large background noise in the experimental water area. The signal received by the receiving array often has insufficient signal-to-noise ratio, and the position of the target sound source cannot be accurately located, resulting in a large measurement uncertainty component.

[0005] Based on this, this application is made. Summary of the Invention

[0006] In view of the deficiencies in the prior art, the present invention provides a method for amplifying a virtual receiving array for distance measurement of an underwater sound source in a linear array. By utilizing the characteristic that the virtual sound source position generated by the position of the target sound source and its reflection on the water surface is symmetric about the water-air interface, the energy of the virtual sound source is folded with the target signal through signal processing, so as to achieve the effect of amplifying the sound source level of the target sound source, and effectively solve the problem of "insufficient signal-to-noise ratio of the signal received by the receiving array during actual measurement" in the background art.

[0007] To achieve the above object, the present invention adopts the following technical solutions:

[0008] A method for amplifying a virtual receiving array for distance measurement of an underwater sound source in a linear array, including the following:

[0009] Determine the positions of each array element on the physical receiving array according to the structure of the physical receiving array;

[0010] Create a corresponding virtual receiving array located above the water according to the positions of each array element on the physical receiving array located underwater;

[0011] The physical receiving array collects and receives the target sound source;

[0012] Segment the signal received by the physical receiving array and create a signal for the virtual receiving array to receive;

[0013] Perform beamforming processing on the signals received by the physical receiving array and the virtual receiving array;

[0014] Obtain the relative position between the target sound source and the receiving array according to the processing result.

[0015] Preferably, according to the segmentation of the signal received by the physical receiving array and creating a signal for the virtual receiving array, specifically: under far-field conditions, the water-air interface is regarded as a horizontal plane; the target sound source located underwater forms a virtual sound source through reflection on the water-air interface; the virtual sound source is axisymmetric with the target sound source about the water-air interface, and the virtual receiving array is axisymmetric with the physical receiving array about the water-air interface; the virtual receiving array receives the target sound source signal and the virtual sound source signal, where the target sound source signal refers to the direct wave generated by the target sound source, and the virtual sound source signal refers to the reflected wave generated by the target sound source through the water-air interface; the signal received by the virtual receiving array is opposite in phase to the signal received by the physical receiving array, and the relative positions of the virtual receiving array, the target sound source and the virtual sound source are the same as those of the physical receiving array, the target sound source and the virtual sound source, and the virtual sound source can be folded to the target sound source to improve the measurement signal-to-noise ratio.

[0016] Preferably, the number of array elements of the virtual receiving array is the same as that of the physical linear receiving array.

[0017] Preferably, the signals received by the elements of the virtual receiving array are composed of the reflected wave and the direct wave generated by the target sound source.

[0018] Preferably, for the virtual receiving array, the reflected wave generated by the target sound source arrives at the elements of the virtual receiving array prior to the direct wave.

[0019] Preferably, the time interval between the reflected wave and the direct wave received by the elements of the virtual receiving array is the same as the time interval between the direct wave and the reflected wave received by the physical elements from the target sound source.

[0020] Preferably, the signals received by the virtual receiving array are created by segmenting the signals received by the physical receiving array, specifically as follows:

[0021] First, considering only the water surface reflection, for any element on the physical receiving array M, the signal s(t) it receives is the sum of the sound wave s0(t) generated by the target sound source and the sound wave s1(t) generated by the virtual sound source.

[0022] s(t) = s0(t) + s1(t) (10)

[0023] Wherein, due to the influence of time delay and water surface reflection, s1(t) is expressed as:

[0024] s1(t) = -s0(t) * δ(t - Δt) (11)

[0025] Substituting formula (11) into formula (10), we can obtain:

[0026] s(t) = s0(t) * [δ(t) - δ(t - Δt)] (12);

[0027] If it is assumed at this time that there is an image element at the position symmetric to the water-air interface of this element, which receives a signal s'(t) with the same amplitude but opposite phase.

[0028] s'(t) = -s0(t) * [δ(t - Δt) - δ(t)] = -s(t) (13)

[0029] According to formula (12), the signals received by the virtual receiving array can be obtained.

[0030] Generally speaking, compared with the prior art by the above technical solution conceived by the present invention, the following beneficial effects can be achieved:

[0031] The virtual receiving array amplification method of the present invention utilizes the characteristic that the virtual sound source position generated by the target sound source position and its water surface reflection is symmetric about the water-air interface, and folds the energy of the virtual sound source with the target signal through signal processing to achieve the effect of amplifying the transmitted sound source level of the target sound source. It can effectively solve the problem of "insufficient signal-to-noise ratio of the signal received by the receiving array during actual measurement" in the background technology. Description of the Drawings

[0032] Figure 1 is a schematic diagram of the signal received by the physical receiving array in this embodiment;

[0033] Figure 2 is a schematic diagram of the signals received by the virtual receiving array and the physical receiving array in this embodiment;

[0034] Among them, Figure 1 the S point in represents the sound source position, the S' point represents the virtual sound source position, and R1 to R6 are the element positions of the linear array M, Figure 2 in the virtual receiving array, the signals of the target sound source and the virtual sound source received are exactly opposite in phase to the signals received by the physical linear array, and the relative positions of the virtual receiving array, the target sound source and the virtual sound source are the same as the relative positions of the receiving array, the target sound source and the virtual sound source. The virtual sound source can be folded to the target sound source to improve the measurement signal-to-noise ratio. Detailed Implementation Modes

[0035] The following describes in detail the specific implementation modes of the present invention with reference to the accompanying drawings, so that those skilled in the art can more clearly understand how to practice the present invention. Although the present invention is described in conjunction with its preferred specific embodiments, these embodiments are only illustrative and do not limit the scope of the present invention.

[0036] When the target sound source emits a sound signal and generates a water surface reflection wave, the reflection wave can be regarded as a sound signal emitted by a virtual sound source. When using a linear array to receive signals, the delay times of the signals received by each element are also different, as Figure 1 shown. The S point in the figure represents the sound source position, the S' point represents the virtual sound source position, and R1 to R6 are the element positions of the linear array M.

[0037] Under far-field conditions, the time delay difference between adjacent elements receiving signals is τ = dsinθ / v, and the expression of the output signal of the beamformer is:

[0038]

[0039] Among them, w m is the transmission coefficient for different elements in the beamforming algorithm;

[0040] Let the transmission coefficient be 1, then the received signals p of each element in Equation (1) can be m(t) is expressed as:

[0041] p m (t) = s[t - (m - 1)τ], m = 1, 2,..., M (2)

[0042] Where s is the discrete signal after sampling the open - circuit voltage of the hydrophone. Therefore, equation (1) can be written as:

[0043]

[0044] Performing Fourier transform on both sides of equation (3), we can obtain:

[0045]

[0046] Thus, the transfer function of the beamformer can be expressed as follows:

[0047]

[0048] Taking a(f, θ) = [1, e -j2πfτ ,.., e -j2πf(M-1)τ T as the array steering vector of the array, also known as the direction vector. It can be seen that the array steering vector a(f, θ) of the array is related to the signal frequency f and the time - delay difference τ of the signals received by adjacent array elements. When the parameters of each element of the sonar array are determined, the time - delay difference τ is only affected by the arrival direction angle θ of the signal. Therefore, it can be obtained that when the parameters of each element of the sonar array are determined, the array steering vector a(f, θ) of the array is only related to the signal frequency f and the arrival direction angle θ of the signal, that is

[0049] a(f, θ) = [1, e -j2πfdsinθ / v , e -j2πf2dsinθ / v ,..., e -j2πf(M-1)dsinθ / v T (6)

[0050] The weighting vector W = [w1, w2,..., w m …w M T can be arbitrarily selected. It can be uniform weighting, cosine weighting, Hamming weighting, etc., or can be calculated by various adaptive algorithms.

[0051] In order to study the localization problem of a moving sound source, it is necessary to adopt beam - forming technology for Doppler - effect removal. When the sound source moves in a straight line at a constant speed v0, the sound - pressure signal measured by the receiving array at the measurement point r = [x, y, z] can be expressed as:

[0052]

[0053] ​​​where \(r_0 = [v_0\tau, 0, 0]\) is the sound source position and \(M_0\) is the Mach number.

[0054]

[0055] Assume that \(N\) scanning points move linearly at a constant speed \(v_0\) in a moving scanning frame, and the positions of these scans are expressed as \(r_0(\tau)+\Delta r\). n , \(n = 1, 2, \cdots, N\), then the time-domain beamforming result after Doppler effect removal is expressed as:

[0056]

[0057] where \(r\) m is the position of the \(m\)-th array element and \(M_0\) is the Mach number. is the angle between the velocity vector \(v_0\) and the vector \(r\) m \(-r_0(\tau)-\Delta r\) n . In this way, the energy of the received acoustic wave signal can be concentrated at a point to achieve the purpose of improving the signal-to-noise ratio. However, when the signal source level of the target sound source still cannot meet the signal-to-noise ratio requirement, the virtual sound source folding and amplification method of this embodiment needs to be used:

[0058] First, considering only the water surface reflection, for any array element on the linear array \(M\), the received signal \(s(t)\) is the sum of the acoustic wave \(s_0(t)\) generated by the target sound source and the acoustic wave \(s_1(t)\) generated by the virtual sound source, as shown in formula (10).

[0059] \(s(t)=s_0(t)+s_1(t)\) (10)

[0060] Among them, due to the influence of time delay and water surface reflection, \(s_1(t)\) can be expressed as:

[0061] \(s_1(t)=-s_0(t)*\delta(t - \Delta t)\) (11)

[0062] Substituting formula (11) into formula (10), we can get:

[0063] \(s(t)=s_0(t)*[\delta(t)-\delta(t - \Delta t)]\) (12)

[0064] If it is assumed at this time that there is a mirror image array element at the symmetric position of this array element with respect to the water-air interface, which receives a signal \(s'(t)\) with the same amplitude but opposite phase, according to formula (12), we can get:

[0065] \(s'(t)=-s_0(t)*[\delta(t - \Delta t)-\delta(t)]=-s(t)\) (13)

[0066] If all the array elements in the linear array are assumed to have corresponding virtual array elements R1' to R6', as Figure 2 shown, then substituting formula (13) into formula (3) at this time gives:

[0067]

[0068] When the linear array performs beamforming at this time, the sound pressure focused at the sound source is doubled, proving the effectiveness of the virtual receiving array amplitude increase method.

[0069] In summary: The present invention is carried out through the following steps in the specific implementation process:

[0070] 1. Determine the positions of each array element on the receiving array according to the structure of the receiving array;

[0071] 2. Create a corresponding virtual receiving array according to the positions of each array element on the receiving array;

[0072] 3. After the experiment, segment the collected signals and create the signals of the virtual linear array according to formula (13);

[0073] 4. Perform beamforming processing on the signals received by the receiving array and the virtual receiving array according to formulas (3) to (9);

[0074] 5. Obtain the relative position between the target sound source and the receiving array according to the signal processing results;

[0075] 6. Perform signal processing and calculation on multiple segments of measured signals and draw the running trajectory of the sound source.

[0076] The above is only the preferred implementation manner of the present invention, but the protection scope of the present invention is not limited thereto. Any modification and replacement based on the technical solution and inventive concept of the present invention should be covered by the protection scope of the present invention.

Claims

1. A method for amplifying a virtual receiving array for determining the distance of an underwater sound source in a linear array, characterized in that It includes the following: Determine the positions of each array element on the physical receiving array according to the structure of the physical receiving array; Create a corresponding virtual receiving array on the water surface according to the positions of each array element on the physical receiving array located underwater; The physical receiving array collects and receives the target sound source; Segment the signals received by the physical receiving array and create a virtual receiving array to receive signals. Specifically, under far-field conditions, the water-air interface is regarded as a horizontal plane; the target sound source located underwater forms a virtual sound source on the water surface through reflection at the water-air interface; the virtual sound source is axisymmetric with the target sound source with the water-air interface as the axis, and the virtual receiving array is axisymmetric with the physical receiving array with the water-air interface as the axis; the virtual receiving array receives the target sound source signal and the virtual sound source signal. The target sound source signal refers to the direct wave generated by the target sound source, and the virtual sound source signal refers to the reflected wave generated by the target sound source through the water-air interface; the signals received by the virtual receiving array are opposite in phase to the signals received by the physical receiving array, and the relative positions of the target sound source and the virtual sound source are the same. The virtual sound source can be folded to the target sound source to improve the measurement signal-to-noise ratio; First, considering only the reflection of the water surface, for any array element on the physical receiving array M, the signal s(t) it receives is the sum of the sound wave s0(t) generated by the target sound source and the sound wave s1(t) generated by the virtual sound source, s(t) = s0(t) + s1(t) (10) Among them, due to the influence of delay and water surface reflection, s1(t) is expressed as: s1(t) = -s0(t) * δ(t - Δt) (11) Substituting formula (11) into formula (10) can obtain: s(t) = s0(t) * [δ(t) - δ(t - Δt)] (12); If it is assumed at this time that there is a mirror image array element at the position symmetric to the water-air interface of this array element that receives a signal s'(t) with the same amplitude but opposite phase, s'(t) = -s0(t) * [δ(t - Δt) - δ(t)] = -s(t) (13) The signal received by the virtual receiving array can be obtained according to formula (12); Perform beamforming processing on the signals received by the physical receiving array and the virtual receiving array; Obtain the relative position between the target sound source and the receiving array according to the processing results.

2. The method for amplifying a virtual receiving array for determining the distance of an underwater sound source in a linear array according to claim 1, characterized in that: The number of array elements of the virtual receiving array is the same as that of the physical receiving array.

3. The method for amplifying a virtual receiving array for determining the distance of an underwater sound source in a linear array according to claim 1, characterized in that: The signals received by the array elements of the virtual receiving array are composed of the reflected wave and the direct wave generated by the target sound source.

4. The method for amplifying a virtual receiving array for determining the distance of an underwater sound source in a linear array according to claim 1, characterized in that: For the virtual receiving array, the reflected wave generated by the target sound source arrives at the array elements of the virtual receiving array earlier than the direct wave.

5. The method for amplifying a virtual receiving array for determining the distance of an underwater sound source in a linear array according to claim 1, characterized in that: The time interval between the reflected wave and the direct wave generated by the target sound source received by the array elements of the virtual receiving array is the same as the time interval between the direct wave and the reflected wave generated by the target sound source received by the physical array elements.

6. The method for amplifying a virtual receiving array for determining the distance of an underwater sound source in a linear array according to claim 1, characterized in that: Perform beamforming processing on the signals received by the physical receiving array and the virtual receiving array. Specifically, under far-field conditions, the time delay difference between the signals received by adjacent array elements is τ = dsinθ / v, where d is the distance between adjacent array elements and v is the sound wave propagation speed. Then the expression of the signal output by the beamformer is: where, w m is the transmission coefficient for different array elements in the beamforming algorithm; Now let the transmission coefficient be 1, then the received signal p of the m-th array element in Equation (1) m (t) can be expressed as: p m \(p(t)=s[t-(m - 1)\tau], m = 1, 2, \cdots, M\ (2)\) In the formula, s is the discrete signal after sampling the open-circuit voltage of the hydrophone. Therefore, formula (1) can be written as: Taking the Fourier transform of both sides of equation (3), we get: Thus, the transfer function of the beamformer can be expressed as follows: Let \(a(f,\theta)=[1,e -j2πfτ ,..,e -j2πf(M-1)τ T be the array steering vector of the array. It can be seen that the array steering vector \(a(f,\theta)\) of the array is related to the signal frequency \(f\) and the time delay difference \(\tau\) of the signals received by adjacent array elements. When the parameters of each element of the sonar array are determined, the time delay difference \(\tau\) is only affected by the direction angle \(\theta\) of the incoming signal. Therefore, it can be obtained that when the parameters of each element of the sonar array are determined, the array steering vector \(a(f,\theta)\) of the array is only related to the signal frequency \(f\) and the direction angle \(\theta\) of the incoming signal, that is​ a(f,θ) = [1, e -j2πfdsinθ / v , e -j2πf2dsinθ / v ,..., e -j2πf(M-1)dsinθ / v T (6)​ The weighted vector W = [w1, w2,..., w m … w M T is obtained by calculating with uniform weights, cosine weighting, Hamming weighting or an adaptive algorithm;​ When the sound source moves in a straight line at a constant speed v0, the sound pressure signal measured by the receiving array at the measurement point r = [x, y, z] is expressed as: where r0 = [v0τ, 0, 0] is the sound source position and M0 is the Mach number. Assume that N scanning points move linearly at a constant speed v0 in a moving scanning frame, and the positions of these scans are expressed as r0(τ)+Δr n , where n = 1, 2,..., N, then the time-domain beamforming result after Doppler effect removal is expressed as: where r m is the position of the m-th array element, M0 is the Mach number, is the angle between the velocity vector v0 and the vector r m - r0(τ) - Δr n and

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