Broadband signal response compensation method and device for underwater simulation sound source
By iteratively optimizing filter parameters, the problem of nonlinearity in underwater acoustic systems that cannot be corrected by traditional linear compensation methods was solved, achieving high-precision broadband signal simulation and improving the performance and stability of the acoustic system.
Patent Information
- Application Number
- CN202610141617.8
- Authority / Receiving Office
- CN · China
- Patent Type
- Applications(China)
- Current Assignee / Owner
- Filing Date
- 2026-02-02
- Publication Date
- 2026-05-26
AI Technical Summary
Traditional linear compensation methods are difficult to accurately correct signal distortion caused by nonlinear factors in underwater acoustic systems, affecting compensation accuracy and adaptability, and failing to achieve high-precision broadband signal simulation.
An iterative optimization mechanism is adopted to dynamically adjust the parameters of the compensation filter. The frequency response of the filter is optimized step by step through the iterative compensation method, thereby gradually improving the compensation effect and adapting to the nonlinear characteristics of the system.
It significantly improves the compensation quality of analog sound sources, enhances audio signal quality, strengthens the stability and reliability of the system under different conditions, and avoids sound quality distortion and loss of detail.
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Figure CN122092824A_ABST
Abstract
Description
Technical Field
[0001] This invention belongs to the field of underwater acoustic engineering technology, specifically relating to a broadband signal response compensation method and device for underwater simulated sound sources. Background Technology
[0002] In acoustic experiments, sound source systems are used to simulate acoustic targets. Currently, linear methods are commonly used for acoustic simulation, which involves measuring the frequency response characteristics of the sound source system at a certain amplitude, generating a simulated signal based on the linear relationship, and thus simulating the acoustic target.
[0003] Due to the inherent nonlinear characteristics of the system, traditional linear compensation methods often fail to achieve ideal compensation results. Firstly, in terms of compensation accuracy, traditional methods cannot accurately capture and correct signal distortion caused by nonlinear factors. Because the behavior of nonlinear systems is often complex and variable, it is difficult to accurately describe using simple linear models. Therefore, traditional methods often result in undercompensation or overcompensation. When dealing with nonlinear problems, traditional methods have limitations in compensation accuracy and adaptability, failing to accurately and effectively compensate for analog sound sources, thus affecting the overall system performance.
[0004] Therefore, there is an urgent need for a method that can effectively handle system nonlinearity and achieve wideband accurate compensation in order to improve the performance of underwater acoustic simulation systems. Summary of the Invention
[0005] The technical problem to be solved by the present invention is to provide a broadband signal response compensation method and device for underwater simulated sound sources. The method uses an iterative optimization mechanism to dynamically adjust the parameters of the compensation filter, thereby effectively correcting the frequency response distortion caused by system nonlinearity and realizing high-precision broadband signal simulation.
[0006] The technical solution of this invention is to provide a broadband signal response compensation method for an underwater simulated sound source, comprising the following steps:
[0007] Step 1: Set the basic parameters of the compensation method, such as the number of iterations, and initialize the frequency response of the filter;
[0008] Step 2: Generate the ideal digital output signal and calculate its power spectrum;
[0009] Step 3: Filter the ideal output signal;
[0010] Step 4: Output the filtered signal through the transmission channel and calculate the power spectrum of the actual output signal;
[0011] Step 5: Calculate the difference between the power spectrum of the actual output signal and the power spectrum of the filtered signal, and correct the frequency response of the filter.
[0012] Step 6: Determine whether to end the iteration. If yes, take the corrected filter frequency response as the final result. If not, return to step 3 and continue.
[0013] This invention proposes an innovative compensation method based on the iterative principle. The core of this iterative compensation method lies in fully considering and adapting to the nonlinear characteristics of the system. It abandons the limitations of one-time compensation in traditional methods, instead adopting a gradual strategy. This method sets initial compensation parameters and then continuously fine-tunes these parameters during actual audio processing. Each iteration is based on the result of the previous iteration, further optimizing the compensation parameters. As the number of iterations increases, the accuracy of the compensation parameters continuously improves, thus making the compensation of the analog sound source increasingly precise and effective. Ultimately, this iterative compensation method can improve the compensation quality of the analog sound source.
[0014] As a preferred option, step 1 is performed as follows:
[0015] 2.1 Setting the number of iteration rounds , signal points filter initialization values Set the initial iteration count ;
[0016] 2.2, Let the filter frequency response be... , Initialize the filter frequency response to ,Right now .
[0017] As a preferred option, step 2 is performed as follows:
[0018] 3.1 Generating an ideal digital output signal ;
[0019] 3.2 Calculate the signal power spectrum Power spectrum The calculation method is as follows:
[0020] .
[0021] Preferably, in step 3, the ideal output signal is filtered to generate a filtered signal. The specific method is as follows:
[0022] .
[0023] Preferably, in step 4, the filtered signal is transmitted through a transmission channel to obtain the actual output. Calculate the actual output signal power spectrum :
[0024] .
[0025] As a preferred option, step 5 is performed as follows:
[0026] 5.1 Calculate the power spectrum of the actual output signal For the power spectrum of the filtered signal Gain :
[0027] (4)
[0028] 5.2, Regarding the frequency response of the filter Make corrections:
[0029] (5)
[0030] 5.3, will Updated to and the iteration count Add 1.
[0031] As a preferred method, determine whether the iteration has ended, i.e., check the iteration count. Is it greater than or equal to the number of iteration rounds? or gain If the preset threshold is reached, the corrected filter frequency response will be used as the final result; otherwise, return to step 3 to continue.
[0032] The preset threshold can be determined based on the allowable error range between the ideal output signal and the actual output signal; or based on the system noise level or signal-to-noise ratio requirements.
[0033] The present invention also provides a broadband signal response compensation device for an underwater simulated sound source, comprising:
[0034] The parameter initialization module is used to set the iteration parameters and initialize the filter frequency response;
[0035] The signal generation and spectrum calculation module is used to generate an ideal output signal and calculate its power spectrum.
[0036] The filtering module is used to filter ideal signals using the current filter frequency response;
[0037] The channel transmission and output spectrum calculation module is used to transmit the filtered signal through the channel and calculate the actual output power spectrum.
[0038] The frequency response correction module is used to calculate the gain and correct the filter frequency response based on the difference between the power spectrum of the actual output and the filtered signal.
[0039] The iteration control module is used to determine whether the iteration is complete and to control the flow jump.
[0040] Compared with the prior art, the present invention has the following advantages:
[0041] This invention employs an iterative compensation method, which accurately captures the complex changes caused by the nonlinear characteristics of the system, thereby achieving progressive optimization of the analog sound source during the compensation process. Each iteration is based on the previous compensation result, fine-tuning the compensation parameters to continuously improve the compensation accuracy, ultimately achieving a compensation effect very close to the ideal state. This not only significantly improves the quality of the audio signal but also enhances the stability and reliability of the system under different operating conditions. It effectively avoids the sound quality distortion and detail loss problems common in traditional methods. Attached Figure Description
[0042] Figure 1 This is a schematic diagram of the implementation process of the present invention.
[0043] Figure 2 This is the ideal digital signal power spectrum in the embodiments of the present invention.
[0044] Figure 3 This is the power spectrum of the filtered output signal in this embodiment of the invention. Detailed Implementation
[0045] The present invention will be further described below with reference to the accompanying drawings and specific embodiments:
[0046] A broadband signal response compensation method for an underwater simulated sound source, such as Figure 1 As shown, the simulation method includes the following steps:
[0047] Step 1: Set the basic parameters of the compensation method, such as the number of iterations, and initialize the frequency response of the filter. The specific method is as follows:
[0048] Set the number of iteration rounds , signal points filter initialization values Set the initial iteration count .
[0049] Let the filter frequency response be , Initialize the filter frequency response to ,Right now .
[0050] Step 2: Generate the ideal digital output signal and calculate its power spectrum. The specific method is as follows:
[0051] Generate an ideal digital output signal .
[0052] Calculate signal power spectrum Power spectrum The calculation method is as follows:
[0053]
[0054] Step 3: Filter the ideal output signal to generate the filtered signal. The specific method is as follows:
[0055]
[0056] Step 4: Obtain the actual output by transmitting the filtered signal through the transmission channel. Calculate the actual output signal power spectrum :
[0057]
[0058] Step 5: Calculate the difference between the power spectrum of the actual output signal and the power spectrum of the filtered signal, and correct the frequency response of the filter. This includes the following steps:
[0059] Calculate the power spectrum of the actual output signal For the power spectrum of the filtered signal Gain :
[0060]
[0061] For filter frequency response Make corrections:
[0062]
[0063] Will Updated to and the iteration count Add 1.
[0064] Step 6: Determine whether to end the iteration, i.e., check the iteration count. Is it greater than or equal to the number of iteration rounds? If yes, then the corrected filter frequency response is taken as the final result; otherwise, return to step three and continue.
[0065] In this embodiment, the number of iteration rounds is set. =8, signal points =5000, filter initialization value = Set the initial iteration count .
[0066] Let the filter frequency response be , Initialize the filter frequency response to ,Right now .
[0067] Generate an ideal digital output signal and calculate its power spectrum, such as Figure 2 As shown.
[0068] The ideal output signal is filtered using equation (2) to generate the filtered signal. .
[0069] The filtered signal is transmitted through the transmission channel to obtain the actual output using equation (3). Calculate the actual output signal power spectrum .
[0070] The difference between the power spectrum of the actual output signal and the power spectrum of the filtered signal is calculated, and the frequency response of the filter is corrected accordingly.
[0071] Will Updated to and the iteration count Add 1.
[0072] To determine whether the iteration has ended, check the iteration count. Is it greater than or equal to the number of iteration rounds? If yes, then the corrected filter frequency response is taken as the final result; otherwise, return to step three and continue.
[0073] The final filtering result is as follows Figure 3 As shown.
[0074] As can be seen, by employing a compensation method based on the iterative principle, this invention can accurately capture the complex changes brought about by the nonlinear characteristics of the system, thereby achieving step-by-step optimization of the analog sound source during the compensation process. Each iteration is based on the compensation result of the previous iteration, fine-tuning the compensation parameters to continuously improve the compensation accuracy, ultimately achieving a compensation effect that is very close to the ideal state. This not only significantly improves the quality of the audio signal but also enhances the stability and reliability of the system under different operating conditions. It effectively avoids the sound quality distortion and detail loss problems common in traditional methods.
[0075] On the other hand, the above-described method of the present invention can be implemented by an apparatus comprising:
[0076] Parameter initialization modules, such as DSPs or CPUs, are used to set iteration parameters and initialize the filter frequency response;
[0077] The signal generation and spectrum calculation module is used to generate an ideal output signal and calculate its power spectrum.
[0078] The filtering module is used to filter ideal signals using the current filter frequency response;
[0079] The channel transmission and output spectrum calculation module is used to transmit the filtered signal through the channel and calculate the actual output power spectrum.
[0080] The frequency response correction module is used to calculate the gain and correct the filter frequency response based on the difference between the power spectrum of the actual output and the filtered signal.
[0081] The iteration control module is used to determine whether the iteration is complete and to control the flow jump.
[0082] The above description only illustrates preferred embodiments of the present invention and should not be construed as limiting the scope of the claims. Any equivalent procedural modifications made using this specification are included within the patent protection scope of this invention.
Claims
1. A broadband signal response compensation method for an underwater simulated sound source, characterized in that: Includes the following steps, Step 1: Set the basic parameters of the compensation method, such as the number of iterations, and initialize the frequency response of the filter; Step 2: Generate the ideal digital output signal and calculate its power spectrum; Step 3: Filter the ideal output signal; Step 4: Output the filtered signal through the transmission channel and calculate the power spectrum of the actual output signal; Step 5: Calculate the difference between the power spectrum of the actual output signal and the power spectrum of the filtered signal, and correct the frequency response of the filter. Step 6: Determine whether to end the iteration. If yes, take the corrected filter frequency response as the final result. If not, return to step 3 and continue.
2. The broadband signal response compensation method for underwater simulated sound sources according to claim 1, characterized in that: Step 1 is performed as follows: 2.1 Setting the number of iteration rounds , signal points filter initialization values Set the initial iteration count ; 2.2, Let the filter frequency response be... , Initialize the filter frequency response to ,Right now .
3. The broadband signal response compensation method for underwater simulated sound sources according to claim 2, characterized in that: Step 2 is performed as follows: 3.1 Generating an ideal digital output signal ; 3.2 Calculate the signal power spectrum Power spectrum The calculation method is as follows: 。 4. The broadband signal response compensation method for underwater simulated sound sources according to claim 3, characterized in that: In step 3, the ideal output signal is filtered to generate the filtered signal. The specific method is as follows: 。 5. The broadband signal response compensation method for underwater simulated sound sources according to claim 4, characterized in that: In step 4, the filtered signal is transmitted through the transmission channel to obtain the actual output. Calculate the actual output signal power spectrum : 。 6. The broadband signal response compensation method for underwater simulated sound sources according to claim 5, characterized in that: Step 5 is performed as follows: 5.1 Calculate the power spectrum of the actual output signal For the power spectrum of the filtered signal Gain : (4) 5.2, Regarding the frequency response of the filter Make corrections: (5) 5.3, will Updated to and the iteration count Add 1.
7. The broadband signal response compensation method for underwater simulated sound sources according to claim 6, characterized in that: In step 6, determine whether to end the iteration, i.e., check the iteration count. Is it greater than or equal to the number of iteration rounds? or gain If the preset threshold is reached, the corrected filter frequency response will be used as the final result; otherwise, return to step 3 to continue.
8. A broadband signal response compensation device for an underwater simulated sound source, characterized in that, include: The parameter initialization module is used to set the iteration parameters and initialize the filter frequency response; The signal generation and spectrum calculation module is used to generate an ideal output signal and calculate its power spectrum. The filtering module is used to filter ideal signals using the current filter frequency response; The channel transmission and output spectrum calculation module is used to transmit the filtered signal through the channel and calculate the actual output power spectrum. The frequency response correction module is used to calculate the gain and correct the filter frequency response based on the difference between the power spectrum of the actual output and the filtered signal. The iteration control module is used to determine whether the iteration is complete and to control the flow jump.
