A method and product for estimating the spatio-temporal distribution of an interface reverberation signal

Abstract

The application discloses a kind of interface reverberation signal space-time distribution estimation method and product, the method includes: constructing and by transmitting transducer emits including multiple sub-pulses acoustic signal, each the sub-pulse time length is same;By hydrophone array with symmetrical form receives echo signal;The echo signal received by two half arrays mutually symmetrical is respectively subjected to conventional beam forming, then to the output of and processing and difference processing is respectively subjected to time-frequency analysis to obtain respective time-frequency spectrum;According to the time and frequency characteristics of the sub-pulse constructed, two-way time-frequency spectrum is respectively subjected to time-frequency coding amplitude formation processing, then difference processing is carried out, and the distribution estimation of interface reverberation is obtained.The method has better effect while also easy to implement, using the method makes underwater sound detector simultaneously detect interface reverberation and target, with in the same time in one fell swoop advantage.

Description

Technical Field

[0001] This invention belongs to the field of marine acoustic detection technology, and particularly relates to a method and product for estimating the spatiotemporal distribution of interface reverberation signals. Background Technology

[0002] Interface reverberation is an important phenomenon in the field of marine acoustic detection. It refers to the reverberation caused by natural objects such as the sea surface and seabed during active acoustic detection. Interface reverberation often exhibits characteristics of random phase and amplitude. Because detection typically uses long-pulse, fixed-frequency signals, and the scale of these natural objects is infinitely large, hydrophones will receive echoes from different directions. These echoes superimpose, making it difficult for existing methods to accurately detect interface reverberation.

[0003] Existing technology one sharpens the beam pattern by dividing the receiving array into two subarrays and forming beam pairs in the same direction, then summing and differentiating the beams. This method is used for target detection, improving target detection performance and azimuth estimation capabilities. However, this method detects point targets and cannot detect infinitely large targets such as interfaces because the echoes from infinitely large targets have random phase and amplitude characteristics. Furthermore, the commonly used matched filtering method cannot be used to detect reverberation with random phase and amplitude characteristics, making it impossible to obtain high-resolution spatiotemporal results through matched filtering. Existing technology two uses multiple measurement points to measure reverberation, but the transmitting and receiving points are not at the same location.

[0004] In summary, from the perspective of reverberation measurement, current technologies lack methods for single-station measurement of marine reverberation, and multi-station methods have stringent application conditions. From the perspective of acoustic measurement technology, due to the random phase and amplitude characteristics of reverberation, it cannot be measured using conventional methods (including the commonly used matched filtering technique). Therefore, existing technologies are used to measure targets, not reverberation. When using active sonar to detect targets, interface reverberation will be a long-term, wide-space echo signal with random phase and amplitude, in the same frequency domain as the target echo. Since the generation mechanism of reverberation and the generation mechanism of target echo are the same, the two are strongly correlated, hindering target detection. To distinguish between target and interface reverberation, there is an urgent need to propose a more accurate estimation method for interface reverberation signals. Summary of the Invention

[0005] The purpose of this invention is to overcome the shortcomings of the prior art and to propose a method and product for estimating the spatiotemporal distribution of interface reverberation signals.

[0006] In view of this, the present invention proposes a method for estimating the spatiotemporal distribution of interface reverberation signals, comprising:

[0007] Step 1: Construct and transmit an acoustic signal comprising multiple sub-pulses, each sub-pulse having the same duration, using a transmitting transducer;

[0008] Step 2: The echo signal is received by a hydrophone array with a symmetrical shape;

[0009] Step 3: After performing conventional beamforming on the echo signals received by the two symmetrical half-arrays, perform time-frequency analysis on the outputs after sum and difference processing to obtain their respective time spectra;

[0010] Step 4: Perform time-frequency encoding amplitude shaping processing on the two time spectra according to the time and frequency characteristics of the constructed acoustic signal, and then perform difference processing to obtain the distribution estimate of the interface reverberation.

[0011] Preferably, the construction of an acoustic signal comprising multiple sub-pulses in step 1 includes:

[0012] When the acoustic signal is a PTCW waveform, the duration T of the sub-pulse is determined. p The number of sub-pulses N, and the total pulse duration T = NT. p Where T conforms to the limitations of the transmitting transducer; the carrier frequency f corresponding to each sub-pulse is determined one by one. n Where n = 1, ..., N, f n Simultaneously satisfying the transmission range of the transmitting transducer and the reception range of the hydrophone; the PTCW waveform s(t) satisfies the following equation:

[0013]

[0014] s p (t)=sin(2πf n t), t∈[0,T p ]

[0015] When the acoustic signal is a PTFM waveform, the duration T of the sub-pulse is determined. p Given the number of sub-pulses N, the starting frequency f and bandwidth B of the pulses are determined. The PTFM waveform s(t) satisfies the following equation:

[0016]

[0017] Preferably, the hydrophone array with a symmetrical shape in step 2 includes two symmetrical half-arrays AL and AR, with the corresponding array elements having an equal number, symmetrical array element positions, and identical properties of mutually symmetrical array elements.

[0018] Preferably, step 3 includes:

[0019] Step 3-1: Determine the detection angle θ based on the detection direction;

[0020] Step 3-2: Perform conventional beamforming on the array elements within the half-arrays AL and AR at angle θ to obtain the corresponding output x. CBL (t) and x CBR (t);

[0021] Step 3-3: For x CBL (t) and x CBR (t) The output of the operation and processing is the sum of the half-array beamforming results x. SUM (t), for x CBL (t) and x CBR (t) Subtraction processing, outputting the difference x between the half-array beamforming results. DIF (t);

[0022] Steps 3-4: For x SUM (t), x DIF (t) Perform time-frequency analysis and output the corresponding time-frequency spectrum x. TFS (t,f) and x TFD (t,f).

[0023] Preferably, steps 3-4 include: when using a Hamming window STFT to x SUM (t), x DIF (t) Perform time-frequency analysis, including:

[0024] Step S3-4-1: Calculate the sub-pulse frequency difference Δf n The greatest common divisor f GCD , where n represents the sub-pulse number, n≠1;

[0025] Step S3-4-2: Take f GCD An integer factor, denoted as Δf, is used as the target for the STFT output frequency spacing. STFTOut ;

[0026] Step S3-4-3: Take the number of points for the window length. Where f s The sampling rate for the signal received by the hydrophone;

[0027] Step S3-4-4: Select sub-pulse duration T p One factor is used as the time interval of the STFT output, denoted as Δt. STFTOut ;

[0028] Step S3-4-5: Calculate the overlap N overlap :N overlap =N window -f s ·Δt STFTOut ;

[0029] Step S3-4-6: Use the calculated window length N windowOverlap N overlap and FFT points N fft As parameters of the STFT, the output x that meets the time-frequency condition is obtained. TFS (t,f) and x TFD (t,f), where N fft =N window .

[0030] Preferably, steps 3-4 include: when using an FIR filter bank to filter x SUM (t), x DIF (t) Perform time-frequency analysis, including:

[0031] Step S3-4-1: Select the center frequency and bandwidth of the FIR filter bank, where the number of filters in the filter bank is N, which is the number of frequency points of the transmitted waveform, and the center frequency of each filter is f, which is the sub-pulse carrier frequency of the transmitted waveform. n n = 1, ..., N;

[0032] Step S3-4-2: Use a standard linear-phase FIR filter;

[0033] Step S3-4-3: Convert the input signal x SUM (t) and x DIF (t) are passed through the filter bank respectively, and N filtered output signals are obtained;

[0034] Steps S3-4-5: Remove the group delay generated by each filter to obtain the output x that meets the time-frequency conditions. TFS (t,f) and x TFD (t,f).

[0035] Preferably, the time-frequency conditions include:

[0036] Results are obtained at all frequency points of the transmitted waveform, and the duration T of the transmitted waveform sub-pulse is... p It can be divided by the time interval of the time-frequency output.

[0037] Preferably, step 4 includes:

[0038] Use a time-frequency encoded amplitude shaper to process the time spectrum x TFS (t,f), x TFD Accumulate (t,f) and calculate the accumulation result x. CMS (t), x CMD (t):

[0039]

[0040] Where f0 is the cumulative reference frequency, taken as the carrier frequency of the first sub-pulse, and Δf nΔt is the difference between the carrier frequency and the reference frequency of the nth sub-pulse. n Let the difference between the start time of the nth sub-pulse and the start time of the 1st sub-pulse satisfy the following formulas:

[0041] Δf n =f n -f0=f n -f1

[0042] Δt n =t n -t1=(n-1)T p

[0043] Use the subtractor on x CMS (t), x CMD Subtracting (t) gives x R (t), that is:

[0044] x R (t)=x CMS (t)-x CMD (t)

[0045] Where, x R (t) The horizontal axis represents time, and the value is the estimated value of the interface reverberation echo in a fixed direction θ.

[0046] Preferably, the method further includes step 5: estimating the interface reverberation echo signal for a set of angles, and plotting the reverberation curve based on the maximum value at different angles, specifically including:

[0047] Step S5-1: Estimate the value x of the interface reverberation echo signal obtained by calculating direction θ1. R Find the maximum value of (t), called RL(t1,θ1), and record the time t1 where the maximum value is located and the maximum value RL(t1,θ1);

[0048] Step S5-2: For each direction m=1,…,M, calculate RL(t) according to step S5-1. m ,θ m Record the time t when the maximum value is reached. m and the maximum value RL(t) m ,θ m );

[0049] Step S5-3: Plot the maximum value and the time of the maximum value on a two-dimensional graph with the horizontal axis representing direction θ and the vertical axis representing t, or plot RL(t)... m ,θ m () is used as the amplitude to draw a three-dimensional image.

[0050] On the other hand, the present invention also provides a computer program product, including a computer program that, when executed by a processor, implements the above-described method.

[0051] Compared with the prior art, the advantages of the present invention are:

[0052] 1. This method utilizes the symmetry of the hydrophone array and creates accumulative conditions in the time-frequency domain through waveform settings, thereby estimating the spatiotemporal distribution of reverberation more accurately.

[0053] 2. Since hydrophone arrays are mostly symmetrical, and the waveform required by this method is easy to transmit, and the computational load of signal processing is not significantly increased, this method is easy to implement while achieving better results.

[0054] 3. This method accumulates data based on hydrophone array elements and echo waveforms, thus exhibiting better stability in the face of changing and unreliable marine environments.

[0055] 4. The hydrophone array and detection waveform used in this method can be used not only for detecting interface reverberation but also for target detection. Therefore, using this method to enable the underwater acoustic detector to simultaneously detect interface reverberation and targets has the advantage of achieving multiple objectives in one time.

[0056] 5. This invention can be used to enhance the estimation accuracy of interface reverberation signals by marine acoustic detection equipment. Attached Figure Description

[0057] Figure 1 This is a flowchart of the spatiotemporal distribution estimation method for interface reverberation according to the present invention;

[0058] Figure 2 The spectrum of an example PTCW signal;

[0059] Figure 3 This is a schematic diagram of the symmetrical hydrophone array required for this method;

[0060] Figure 4 A schematic diagram showing the positional relationship between the detector and the interface;

[0061] Figure 5 A signal processing method for reverberant echo signals at ocean interfaces;

[0062] Figure 6 The measurement results are used to apply the present invention to the reverberation measurement of the sea surface. Detailed Implementation

[0063] Considering that although interface reverberation has random phase and amplitude characteristics, its frequency is still fixed, this invention takes advantage of its constant frequency characteristic and incorporates the time-frequency characteristics of frequency hopping signals. By sending frequency hopping signals, the time spectrum of the beam is extracted and accumulated at specific frequencies, making the reverberation signal in the corresponding direction more significant, thus forming a better interface reverberation estimation method.

[0064] This method is a spatiotemporal distribution estimation method for interface reverberation echo signals. This method offers advantages in signal processing, but also places demands on the preceding sound generation and reception steps. The preceding steps employ conventional procedures, but only by using them in accordance with the manner described in this invention can the signal processing steps of this invention produce their beneficial effects.

[0065] Step S1: Preliminary steps. The preliminary steps include sending and receiving sound signals.

[0066] Step S1.1: Transmitting the acoustic signal. This method relies on an existing method for transmitting acoustic signals. This method must be used to design and transmit the acoustic signal for the innovative signal processing method S2 of this invention to take effect. The transmitted underwater acoustic signal waveform should be a combination of multiple continuous waves (CW), called a continuous wave sequence (Pulse Train of Continuous Wave). The PTCW waveform is a continuous transmission of N CW waveforms at different frequencies. The PTCW waveform can be constructed by determining the number of sub-pulses, the duration of each sub-pulse, and the frequency of each sub-pulse. The constructed PTCW waveform is recorded. The constructed PTCW signal is transmitted using an underwater acoustic transducer. Alternatively, a PTFM (Pulse Train of Frequency Modulation) waveform composed of a set of linear frequency modulated (LFM) wave pulses can be transmitted.

[0067] Step S1.2: Acquiring acoustic signals. This method relies on a universally applicable hydrophone array and requires a specific signal acquisition duration. A hydrophone meeting this requirement and the specified duration are necessary for the innovative signal processing method S2 of this invention to take effect. The hydrophone must have a symmetrical physical form. That is, regardless of the arrangement of the array elements, it should be divided into two symmetrical parts, each containing an equal number of elements. Based on relative position, the two parts are referred to as the left half (AL, Array on the Left) and the right half (AR, Array on the Right). Here, AL and AR are not required to be in an absolute left-right relationship in the horizontal direction; they can be rotated arbitrarily in space. Left and right only indicate relative position. Hydrophones configured according to this symmetrical characteristic meet the requirements of this method.

[0068] Subsequently, based on the qualified array, a hydrophone array is used to collect echoes for a certain duration. The echo collection duration should be sufficiently long, greater than T. max :

[0069]

[0070] Where T max For the total duration of echo reception, h is the distance from the detector (underwater acoustic transducer and hydrophone array) to the target interface (sea surface or seabed), L is the farthest distance (in horizontal distance) from which the reverberation of the desired interface is detected, and c is the speed of sound in water.

[0071] Step S2: Signal processing is performed on the echo data received by the hydrophone array. Step S2 is an innovative aspect of this invention, but it relies on step S1 to produce good results. The signal processing step S2 consists of the following steps:

[0072] Step S2.1: Select and determine the direction θ of the reverberation echo signal of the interface to be detected.

[0073] Step S2.2: For the two subarrays AL and AR of the hydrophone array, conventional beamforming (CBF) is performed on the direction θ according to their element positions. The data acquired by AL and AR, after passing through CBF, will output x respectively. CBL (t) and x CBR (t).

[0074] Step S2.3: Use the adder (ADD, Adder) and subtractor (SUB, Subtractor) to process x. CBL (t), x CBR (t) Sum and difference of the two. The adder output is x. SUM (t), the subtractor output is x DIF (t), that is:

[0075] x SUM (t)=ADD(x CBL (t),x CBR (t))=x CBL (t)+x CBR (t)

[0076] x DIF (t)=SUB(x CBL (t),x CBR (t))=x CBL (t)-x CBR (t)

[0077] Step S2.4: For xSUM (t), x DIF (t) are input to a time-frequency analyzer (TFA) for time-frequency analysis, and the output is its time spectrum, referred to as x. TFS (t,f) and x TFD (t,f). The time-frequency analyzer can use a Short-Time Fourier Transform (STFT) or a set of Finite Impulse Response (FIR) filters to generate the time spectrum, i.e.:

[0078] x TFS (t,f)=TFA(x SUM (t))

[0079] x TFD (t,f)=TFA(x DIF (t)) Step S2.5: Using the Time-Frequency Code Magnitude Former (TFCMF), process the time spectrum x TFS (t,f) and x TFD (t,f) are used for time-frequency encoding amplitude formation respectively. The output of this step is x. CMS (t) and x CMD (t). The time-frequency encoded amplitude shaper used in this step processes the echo signal according to the time and frequency characteristics of the PTCW constructed in step S1.1 of the preceding step S1:

[0080]

[0081] Wherein, the reference frequency f0 is taken as equal to the carrier frequency f1 of the first sub-pulse, Δf n Δt is the difference between the carrier frequency and the reference frequency of the nth sub-pulse. n This is the difference between the start time of the nth sub-pulse and the start time of the 1st sub-pulse. That is:

[0082] Δf n =f n -f0=f n -f1

[0083] Δt n =t n -t1=(n-1)T p

[0084] Step S2.6: Use the subtractor to subtract x CMS (t), x CMD (t) is the difference, i.e.:

[0085] x R (t)=SUB(x CMS (t),x CMD (t))=x CMS (t)-x CMD (t)

[0086] x R (t) is the estimated value of the interface reverberation. This value is for the direction θ selected in step S2.1.

[0087] (Optional) Step S3: If it is necessary to estimate the interface reverberation signal in multiple directions, the hydrophone array needs to have good receiving directivity for the desired multiple directions. For a selected set of directions θ1,…,θ M Proceed one by one:

[0088] Step S3.1: Calculate the estimated value x of the interface reverberation for direction θ1 according to step S2. R Find the maximum value of (t), which is called RL(t1,θ1).

[0089] Step S3.2: For m = 1, ..., M, calculate RL(t) according to step S3.1. m ,θ m ).

[0090] Step S3.3: Transfer M RL(t) values ​​to... m ,θ m The data is plotted on an image with the horizontal axis θ and the vertical axis t. This image reflects the echo delay at which the estimated maximum value of the interface reverberation echo signal is located in each of a set of directions. This is called the reverberation curve. The reverberation curve describes the arrival time properties of interface reverberation in different directions.

[0091] The technical solution of the present invention will be described in detail below with reference to the accompanying drawings and embodiments.

[0092] Example 1

[0093] Embodiment 1 of the present invention proposes a method for estimating the spatiotemporal distribution of interface reverberation signals, such as... Figure 1 As shown. The specific steps are as follows:

[0094] Step S1: Preliminary steps, including sending and receiving acoustic signals. Step S2: Signal processing step, which involves directional signal processing of the echo data received by the hydrophone array to obtain a directional estimation result of the reverberation. Step S3: Multi-directional signal processing step. Step S2 is the innovation of this patent. While the sub-steps within step S1 are general steps, their selection is necessary to provide a reasonable prerequisite for the innovative step S2 of this patent.

[0095] The specific implementation methods for the important steps are given below:

[0096] (1) Step S1.1: Construct a PTCW waveform or other waveforms with accumulative properties.

[0097] This method requires the transmission of a PTCW waveform. A PTCW waveform is a continuous combination of multiple CWs, where each sub-pulse has a different frequency but the same duration. Such a waveform can be constructed using the following method:

[0098] 1. First, determine the duration T of the sub-pulse. p The number of sub-pulses, N. Note that the total pulse duration is T = NT. p The duration of subpulses and the total duration may be limited by the transmitting transducer; if the duration is too long, the energy consumption may also be too high. Various actual conditions of the detector need to be considered when setting the duration and number of subpulses.

[0099] 2. Determine the carrier frequency f corresponding to each sub-pulse. n Where n = 1, ..., N. The frequency f of the transmitted waveform. n It should be within the range where the transmitting transducer can emit sound, and also within the range where the hydrophone can receive it.

[0100] 3. Substitute into the formula to generate the PTCW waveform:

[0101]

[0102] in

[0103] s p (t)=sin(2πf n t), t∈[0,T p ]

[0104] Each sub-pulse of the PTCW waveform is a sine wave. However, after performing a time-frequency transformation using STFT, it becomes clear that its energy in the time-frequency domain is concentrated at multiple different frequencies, with each frequency having an equal duration. Figure 2 As shown.

[0105] This step can also send a PTFM waveform, which is an LFM pulse train obtained by connecting a set of LFM waves end to end.

[0106] Constructed using the following method:

[0107] 1. Determine the duration T of the sub-pulse. p The number of sub-pulses, N.

[0108] 2. Determine the starting frequency f of the pulse. l And bandwidth B.

[0109] 3. Substitute into the formula to generate the PTFM waveform:

[0110]

[0111] in

[0112]

[0113] Similar to PTCW and PTFM, this method can be used when the waveform has multiple sub-pulses and exhibits repeatable and accumulative properties in the time and frequency domain.

[0114] (2) Step S1.2: Select a symmetrical hydrophone array to collect echo signals.

[0115] This method assumes that the hydrophone array used in the detector can be divided into two symmetrical parts, denoted as AL and AR. This means that there should be an even number of elements. After dividing the array, AL and AR should contain an equal number of elements, with symmetrical element positions and consistent element properties. A schematic diagram of a symmetrical hydrophone array is shown below. Figure 3 As shown.

[0116] Although referred to as the left half array and the right half array, in actual use they are not limited to an absolute left-right relationship, but only to refer to two mutually symmetrical sides. In actual use, hydrophone arrays can be symmetrical vertically, horizontally, or at an angle, as long as both sides are symmetrical.

[0117] Additionally, a certain duration is required to utilize the echo from the hydrophone array. When measuring reverberation, assume the distance between the detection device and the interface is h, and the distance between the transmitting transducer and the hydrophone in the detection device is negligible (much less than h). If the desired maximum horizontal distance to the detection interface is L, such as... Figure 4 As shown. Then, according to the Pythagorean theorem, the desired distance from the farthest point of the detection interface to the detection device is:

[0118]

[0119] Let the speed of sound in water be c. Considering that the time for the sound signal to travel from the transmitting transducer to the interface and back to the hydrophone is a two-way journey, the total time required to receive the echo is T. max for:

[0120]

[0121] (3) Step S2: Signal processing.

[0122] Upon receiving the echo, to obtain a better estimation of the interface reverberation signal, this method employs the following approach: the receiving array elements are divided into two parts for processing, based on conventional beamforming technology, and accumulation processing is performed in the time-frequency domain according to signal characteristics. The overall process is as follows: Figure 5 As shown.

[0123] Step S2.1: Determine a direction θ to be detected. Generally, θ is composed of the elevation angle and the horizontal azimuth angle. θ is determined by the detection objective, and both the transmitting transducer and the hydrophone array need to be able to transmit and receive signals in that direction.

[0124] Step S2.2: Perform conventional beamforming on the array elements within the half-arrays AL and AR at angle θ. The purpose of conventional beamforming is to phase-shift and accumulate the signals based on the relative positions of the array elements, combining the waveforms received by multiple elements to achieve better detection results. The outputs after CBF for the left and right half-arrays are x... CBL (t) and x CBR (t).

[0125] Step S2.3: Use adder ADD and subtractor SUB to process x CBL (t), x CBR (t) Perform summation and difference calculations. The output is x. SUM (t), x DIF (t). That is:

[0126] x SUM (t)=ADD(x CBL (t),x CBR (t))=x CBL (t)+x CBR (t)

[0127] x DIF (t)=SUB(x CBL (t),x CBR (t))=x CBL (t)-x CBR (t)

[0128] x SUM (t) represents the sum of the results of half-array beamforming, x DIF (t) represents the difference in the beamforming results of the half-array.

[0129] Step S2.4: Use the time-frequency spectrum analyzer (TFA) to analyze x SUM (t), x DIF (t) Perform time-frequency analysis. The output is the time spectrum x. TFS (t,f), x TFD(t,f). Various time-frequency analyzers can be used as the TFA, such as STFT, or a set of FIR filters. However, for the subsequent step S2.5 to be operable, the output of the time-frequency analyzer should have the following characteristics:

[0130] 1. Results should be available at all frequency points of the transmitted waveform so that the results at these frequency points can be extracted;

[0131] 2. Transmitted waveform sub-pulse duration T p It should be divisible by the time interval of the TFA output.

[0132] The output x of a TFA that meets the above two conditions TFS (t,f), x TFD (t,f) creates the conditions for the accumulation in the subsequent step S2.5; otherwise, the accumulation would not be possible because the sub-pulse frequency was not calculated or the time could not be aligned.

[0133] Taking an STFT using a Hamming window as an example, to achieve the above two conditions, the following steps should be followed: S2.4.1-S2.4.5 Selecting time-frequency transformation parameters:

[0134] Step S2.4.1: Calculate the sub-pulse frequency difference Δf n The greatest common divisor (GCD) of n (n ≠ 1) is denoted as f. GCD .

[0135] Step S2.4.2: Take f GCD An integer factor, denoted as Δf, is used as the target for the STFT output frequency spacing. STFTOut .

[0136] Step S2.4.3: Calculate the number of points for the window length. Where f s The sampling rate for the signal received by the hydrophone.

[0137] Step S2.4.4: Select the sub-pulse duration T p One factor is used as the time interval of the STFT output, denoted as Δt. STFTOut .

[0138] Step S2.4.5: Calculate the overlap N overlap =N window -f s ·Δt STFTOut .

[0139] Use the calculated window length N window Overlap N overlap , FFT points N fft =N windowAs parameters of the STFT, the output x that meets the above two conditions can be calculated. TFS (t,f), x TFD (t,f).

[0140] If an FIR filter bank is used as the time-spectrum analyzer, the following steps S2.4.6-S2.4.9 must be followed:

[0141] Step S2.4.6: Select the center frequency and bandwidth of the FIR filter bank. The number of filters in the filter bank is equal to the number of frequency points N of the transmitted waveform, and the center frequency of each filter should be equal to the sub-pulse carrier frequency f of the transmitted waveform. n , n=1,…,N.

[0142] Step S2.4.7: Determine the filter order and window function, and design each FIR filter separately. To ensure that the phase remains unchanged relative to the original data, a standard linear-phase FIR filter should be used.

[0143] Step S2.4.8: Convert the input signal x SUM (t), x DIF (t) are passed through the filter bank respectively, and N filtered signals are obtained from each output.

[0144] Step S2.4.9: Remove the group delay generated by each filter.

[0145] Input signal x SUM (t), x DIF (t) After steps S2.4.6-S2.4.9, the output is obtained as N filtered time-domain signals with corrected delays. The sampling rate is the same as the original signal, and results are available at all frequency points of the transmitted waveform, thus satisfying two conditions.

[0146] Step S2.5: Use the Time-Frequency Encoded Amplitude Former (TFCMF) to process the time spectrum x TFS (t,f), x TFD Accumulate (t,f) and calculate the accumulation result x. CMS (t), x CMD (t). That is, calculate:

[0147]

[0148] Where f0 is the cumulative reference frequency, which is taken as the carrier frequency of the first sub-pulse, i.e., f0 = f1. Δf n Δt is the difference between the carrier frequency and the reference frequency of the nth sub-pulse. n This is the difference between the start time of the nth sub-pulse and the start time of the 1st sub-pulse. That is:

[0149] Δf n =fn -f0=f n -f1

[0150] Δt n =t n -t1=(n-1)T p

[0151] Due to the generated time spectrum x TFS (t,f), x TFD (t,f) satisfies condition 1 proposed in step S2.4: it has results at all frequency points of the transmitted waveform, therefore from the time spectrum x TFS (t,f), x TFD Extracting a time-domain signal with frequency f = f0 + Δf from (t,f) is feasible. This is because condition 2 is satisfied: the sub-pulse duration T... p The time interval Δt between the output time points of TFS and TFD should be usable. STFTOut Divisible, therefore x is taken from the time spectrum TFS (t,f0+Δf n ), x TFD (t,f0+Δf n x can be obtained by time-shifting by an integer number of sampling points. TFS (t+Δt n ,f0+Δf n ), x TFD (t+Δt n ,f0+Δf n This makes the cumulative summation after the shift feasible.

[0152] After extraction and summation in this step, the data corresponding to the frequency and time delay in the echo will be accumulated. The accumulated signal contains the power corresponding to all sub-pulses of the transmitted waveform, making the reverberation more prominent.

[0153] Step S2.6: Use the subtractor to subtract x CMS (t), x CMD Subtracting (t) gives x R (t), that is:

[0154] x R (t)=SUB(x CMS (t),x CMD (t))=x CMS (t)-x CMD (t)

[0155] x R (t) The x-axis represents time, and the values ​​are estimated interface reverberation echoes in a fixed direction θ. For x RThe estimation of (t) utilizes the sidelobe suppression effect obtained by processing the two half-arrays of the symmetrical array separately, and the accumulation effect of the echo in the time spectrum according to the emission waveform pattern, which has better reliability.

[0156] During a sea test, a deep-sea exploration platform used this method to process the echo signal of a PTCW signal. When the horizontal azimuth angle was 0 degrees and the elevation angle was 50 degrees, the resulting reverberation x... R (t) such as Figure 6 As shown. Near 0 seconds is the sound signal received by the hydrophone directly transmitted from the transducer; at 4.06 seconds is the reverberation of the signal transmitted vertically from the detection platform to the sea surface, reflected by the sea surface, and then reaching the platform; at 6.75 seconds is the reverberation of the sea surface in the selected direction.

[0157] (4) Step S3: Estimate the interface reverberation echo signal for a set of angles to obtain the reverberation curve.

[0158] If necessary, further estimates of the interface reverberation echo signal can be obtained for a set of angles, and reverberation curves can be plotted based on the maximum values ​​at different angles. The reverberation curves can be used to compare with target characteristics to derive other necessary conclusions.

[0159] The determination of the interface reverberation curve requires the selection of a set of angles θ1,…,θ M For example, if the detection device aims to detect differences in interface reverberation at different horizontal azimuths, the angle should be set as follows: the pitch angle remains constant, while the horizontal azimuth angle is sampled at certain intervals from 0 to 360 degrees. If the purpose of the detection device is to detect changes in interface reverberation at different pitch angles, the angle can be set as follows: the pitch angle varies at intervals from 0 to 90 degrees, while the horizontal azimuth angle remains constant.

[0160] The following method was used to compare the time delays at which the maximum values ​​of the interface reverberation echo signals were located in different directions:

[0161] Step S3.1: Calculate the estimated value x of the interface reverberation echo signal for direction θ1 according to step S2. R Find the maximum value of (t), called RL(t1,θ1), and record the time t1 where the maximum value is located and the maximum value RL(t1,θ1).

[0162] Step S3.2: For each direction m = 1, ..., M, calculate RL(t) according to step S3.1. m ,θ m Record the time t when the maximum value is reached. m and the maximum value RL(t) m ,θ m ).

[0163] Step S3.3: Plot the maximum value and the time at which the maximum value occurs on a graph with the x-axis representing the direction θ and the y-axis representing t. A two-dimensional graph can be plotted, or RL(t) can be used. m ,θ m () is used as the amplitude to draw a three-dimensional image.

[0164] Example 2

[0165] Embodiments of the present invention may also provide a computer program product, including a computer program / instructions. When the computer program / instructions are executed by a processor, the various steps in the above method embodiments can be implemented.

[0166] In summary, the purpose of this method is to measure reverberation, not to measure any specific target. It has the following two main innovations: ① The timing and method of converting the signal into energy are different from existing technologies; ② This method additionally requires the time-frequency characteristics of the transmitted waveform and uses these time-frequency characteristics to specifically extract and accumulate the signal energy.

[0167] Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention and are not intended to limit it. Although the present invention has been described in detail with reference to the embodiments, those skilled in the art should understand that modifications or equivalent substitutions to the technical solutions of the present invention do not depart from the spirit and scope of the technical solutions of the present invention, and all such modifications or substitutions should be covered within the scope of the claims of the present invention.

Claims

1. A method for estimating the spatiotemporal distribution of an interface reverberation signal, comprising: Step 1: Construct and transmit an acoustic signal comprising multiple sub-pulses, each sub-pulse having the same duration, using a transmitting transducer; Step 2: The echo signal is received by a hydrophone array with a symmetrical shape; Step 3: After performing conventional beamforming on the echo signals received by the two symmetrical half-arrays, perform time-frequency analysis on the outputs after sum and difference processing to obtain their respective time spectra; Step 4: Perform time-frequency encoding amplitude shaping processing on the two time spectra according to the time and frequency characteristics of the constructed acoustic signal, and then perform difference processing to obtain the distribution estimate of the interface reverberation.

2. The spatiotemporal distribution estimation method for interface reverberation signals according to claim 1, characterized in that, Step 1 involves constructing an acoustic signal comprising multiple sub-pulses, including: When the acoustic signal is a PTCW waveform, the duration T of the sub-pulse is determined. p The number of sub-pulses N, and the total pulse duration T = NT. p Where T conforms to the limitations of the transmitting transducer; the carrier frequency f corresponding to each sub-pulse is determined one by one. n Where n = 1, ..., N, f n Simultaneously satisfying the transmission range of the transmitting transducer and the reception range of the hydrophone; the PTCW waveform s(t) satisfies the following equation: s p (t)=sin(2πf n t),t∈[0,T p ] When the acoustic signal is a PTFM waveform, the duration T of the sub-pulse is determined. p The number of sub-pulses N determines the starting frequency f of the pulse. l With bandwidth B, the PTFM waveform s(t) satisfies the following equation:

3. The spatiotemporal distribution estimation method for interface reverberation signals according to claim 1, characterized in that, The hydrophone array with a symmetrical shape in step 2 includes two symmetrical half arrays AL and AR. The corresponding array elements should have an equal number, symmetrical positions, and consistent properties.

4. The spatiotemporal distribution estimation method for interface reverberation signals according to claim 3, characterized in that, Step 3 includes: Step 3-1: Determine the detection angle θ based on the detection direction; Step 3-2: Perform conventional beamforming on the array elements within the half-arrays AL and AR at angle θ to obtain the corresponding output x. CBL (t) and x CBR (t); Step 3-3: For x CBL (t) and x CBR (t) The output of the operation and processing is the sum of the half-array beamforming results x. SUM (t), for x CBL (t) and x CBR (t) Subtraction processing, outputting the difference x between the half-array beamforming results. DIF (t); Steps 3-4: For x SUM (t), x DIF (t) Perform time-frequency analysis and output the corresponding time-frequency spectrum x. TFS (t,f) and x TFD (t,f).

5. The spatiotemporal distribution estimation method for interface reverberation signals according to claim 4, characterized in that, Steps 3-4 include: when using a Hamming window STFT to x SUM (t), x DIF (t) Perform time-frequency analysis, including: Step S3-4-1: Calculate the sub-pulse frequency difference Δf n The greatest common divisor f GCD , where n represents the sub-pulse number, n≠1; Step S3-4-2: Take f GCD An integer factor, denoted as Δf, is used as the target for the STFT output frequency spacing. STFTOut ; Step S3-4-3: Take the number of points for the window length. Where f s The sampling rate for the signal received by the hydrophone; Step S3-4-4: Select sub-pulse duration T p One factor is used as the time interval of the STFT output, denoted as Δt. STFTOut ; Step S3-4-5: Calculate the overlap N overlap :N overlap =N window -f s ·Δt STFTOut ; Step S3-4-6: Use the calculated window length N window Overlap N overlap and FFT points N fft As parameters of the STFT, the output x that meets the time-frequency condition is obtained. TFS (t,f) and x TFD (t,f), where N fft =N window .

6. The spatiotemporal distribution estimation method for interface reverberation signals according to claim 4, characterized in that, Steps 3-4 include: when using an FIR filter bank to pair x SUM (t), x DIF (t) Perform time-frequency analysis, including: Step S3-4-1: Select the center frequency and bandwidth of the FIR filter bank, where the number of filters in the filter bank is N, which is the number of frequency points of the transmitted waveform, and the center frequency of each filter is f, which is the sub-pulse carrier frequency of the transmitted waveform. n n = 1, ..., N; Step S3-4-2: Use a standard linear-phase FIR filter; Step S3-4-3: Convert the input signal x SUM (t) and x DIF (t) are passed through the filter bank respectively, and N filtered output signals are obtained; Steps S3-4-5: Remove the group delay generated by each filter to obtain the output x that meets the time-frequency conditions. TFS (t,f) and x TFD (t,f).

7. The method for estimating the spatiotemporal distribution of interface reverberation signals according to claim 5 or 6, characterized in that, The time-frequency conditions include: Results are obtained at all frequency points of the transmitted waveform, and the duration T of the transmitted waveform sub-pulse is... p It can be divided by the time interval of the time-frequency output.

8. The spatiotemporal distribution estimation method for interface reverberation signals according to claim 4, characterized in that, Step 4 includes: Use a time-frequency encoded amplitude shaper to process the time spectrum x TFS (t,f), x TFD Accumulate (t,f) and calculate the accumulation result x. CMS (t), x CMD (t): Where f0 is the cumulative reference frequency, taken as the carrier frequency of the first sub-pulse, and Δf n Δt is the difference between the carrier frequency and the reference frequency of the nth sub-pulse. n Let the difference between the start time of the nth sub-pulse and the start time of the 1st sub-pulse satisfy the following formulas: Δf n =f n -f0=f n -f1 Δt n =t n -t1=(n-1)T p Use the subtractor on x CMS (t), x CMD Subtracting (t) gives x R (t), that is: x R (t)=x CMS (t)-x CMD (t) Where, x R (t) The horizontal axis represents time, and the value is the estimated value of the interface reverberation echo in a fixed direction θ.

9. The spatiotemporal distribution estimation method for interface reverberation signals according to claim 8, characterized in that, The method further includes step 5: estimating the interface reverberation echo signal for a set of angles, and plotting the reverberation curve based on the maximum value at different angles, specifically including: Step S5-1: Estimate the value x of the interface reverberation echo signal obtained by calculating direction θ1. R Find the maximum value of (t), called RL(t1,θ1), and record the time t1 where the maximum value is located and the maximum value RL(t1,θ1); Step S5-2: For each direction m=1,…,M, calculate RL(t) according to step S5-1. m ,θ m Record the time t when the maximum value is reached. m and the maximum value RL(t) m ,θ m ); Step S5-3: Plot the maximum value and the time of the maximum value on a two-dimensional graph with the horizontal axis representing direction θ and the vertical axis representing t, or plot RL(t)... m ,θ m () is used as the amplitude to draw a three-dimensional image.

10. A computer program product, comprising a computer program, characterized in that, When the computer program is executed by a processor, it implements the method as described in any one of claims 1-9.

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