A method and system for directional sound generation of loudspeakers based on offline simulation and feedforward control

Abstract

This invention provides a method and system for directional sound generation from a loudspeaker based on offline simulation and feedforward control, relating to the field of electroacoustic technology. The method includes: based on the feedforward compensation database, in the online implementation phase, performing frequency decomposition on the input audio signal using a digital signal processor and querying the database to obtain compensation parameters corresponding to each frequency component, thereby obtaining a real-time inverse compensation signal; using the real-time inverse compensation signal, synchronously driving the main loudspeaker and auxiliary loudspeaker, causing the compensated sound wave radiated by the auxiliary loudspeaker to destructively interfere with the sound wave behind the main loudspeaker, forming an acoustic quiet zone. This invention combines offline simulation to construct a precise database with online feedforward query for real-time compensation, ensuring accuracy and efficiency throughout the entire process from data foundation to acoustic quiet zone formation.

Description

Technical Field

[0001] This invention relates to the field of electroacoustic technology, and in particular to a method and system for directional sound generation of loudspeakers based on offline simulation and feedforward control. Background Technology

[0002] Directional speaker technology aims to control the direction of sound wave propagation, concentrating sound energy in a specific area while minimizing sound wave interference from other areas, creating a so-called acoustic quiet zone or private sound field. This technology has broad application prospects in scenarios such as public information dissemination, personal audio spaces, and in-vehicle sound field zoning.

[0003] Existing directional sound generation technologies for loudspeakers mainly rely on the following approaches: First, beamforming technology based on loudspeaker arrays, which synthesizes directional sound beams in space by controlling the phase and amplitude of multiple loudspeaker units in the array; second, the principle of parametric arrays, which utilizes the nonlinear effects of ultrasound in air to generate audible sound. However, these technologies have significant limitations. Beamforming technology typically requires a large number of loudspeaker units and complex real-time signal processing algorithms, resulting in high system costs and computational burdens, and may struggle to achieve effective directional control in the low-frequency range. Parametric array technology, on the other hand, may suffer from low conversion efficiency, limited output sound pressure levels, and nonlinear distortion.

[0004] More importantly, the aforementioned technologies primarily focus on enhancing sound pressure in the target area, with insufficient attention paid to how to actively and effectively suppress sound pressure in non-target areas (especially behind the loudspeaker) to create a high-quality acoustic quiet zone. Some attempts have attempted to use feedback control methods, which involve placing an error microphone in the quiet zone to collect sound pressure signals in real time and adjusting the control signal accordingly. However, this method has high real-time requirements for the feedback system, may be prone to stability issues, and the system deployment is limited by the physical location of the error microphone. Summary of the Invention

[0005] The technical problem to be solved by the present invention is to provide a method and system for directional sound generation of loudspeakers based on offline simulation and feedforward control. By combining offline simulation to build an accurate database with online feedforward query and real-time compensation, the entire process from data foundation to acoustic quiet zone formation is accurate and efficient.

[0006] To solve the above-mentioned technical problems, the technical solution of the present invention is as follows:

[0007] In a first aspect, a method for directional sound emission from a loudspeaker based on offline simulation and feedforward control is provided, the method comprising:

[0008] A high-precision three-dimensional acoustic simulation model was established to simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment;

[0009] Finite element simulation was performed using the high-precision three-dimensional acoustic simulation model to calculate the sound field distribution of the main loudspeaker in the target frequency range and to obtain the sound pressure level, phase and time delay data of the area behind it.

[0010] Based on the sound pressure level, phase, and time delay data, a feedforward compensation database is constructed. The database stores the amplitude and phase of the rearward sound wave from the main loudspeaker, as well as the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference.

[0011] Based on the feedforward compensation database, during the online implementation phase, the input audio signal is frequency decomposed by a digital signal processor, and the database is queried to obtain the compensation parameters corresponding to each frequency component in order to obtain a real-time inverse compensation signal.

[0012] Using the real-time phase-inverse compensation signal, the main loudspeaker and the auxiliary loudspeaker are driven synchronously, so that the compensation sound wave radiated by the auxiliary loudspeaker and the sound wave behind the main loudspeaker undergo destructive interference, forming an acoustic quiet zone.

[0013] Furthermore, a high-precision three-dimensional acoustic simulation model is established to simulate the sound wave propagation behavior of the main speaker and auxiliary speaker in a specific environment, including:

[0014] Based on the physical parameters of the main speaker and the auxiliary speaker and the geometric structure of their environment, a three-dimensional geometric model is established that includes the speakers and their spatial positional relationships.

[0015] Based on the environment type defined by the three-dimensional geometric model, material properties and physical parameters are set for the loudspeaker diaphragm, enclosure and air medium, and corresponding acoustic boundary conditions are configured for the boundaries in the environment. The environment type includes free field, room environment or sound-absorbing environment.

[0016] Based on the aforementioned material properties and acoustic boundary conditions, the diaphragms of the main loudspeaker and the auxiliary loudspeaker are defined as sound source boundaries, and their excitation conditions are set to simulate the sound generation process.

[0017] The model with the established sound source boundary and excitation conditions is meshed, and acoustic simulation is performed within the target frequency range to obtain a high-precision three-dimensional acoustic simulation model required to simulate the propagation behavior of sound waves.

[0018] Furthermore, finite element simulation is performed using the high-precision three-dimensional acoustic simulation model to calculate the sound field distribution of the main loudspeaker within the target frequency range, and to obtain the sound pressure level, phase, and time delay data of the region behind it, including:

[0019] In the high-precision three-dimensional acoustic simulation model, a target area is set behind the main speaker, and multiple frequency points within the target frequency range are selected.

[0020] Based on the frequency point and target area, finite element simulation is performed to calculate the sound field distribution generated in the target area when only the main loudspeaker is excited, and the sound pressure level, phase and time delay data of each point in the target area are extracted as the sound field data of the main loudspeaker.

[0021] Based on the sound field data of the main speaker, an auxiliary speaker is introduced into the simulation and a sound field superposition simulation is performed. By iteratively adjusting the amplitude and phase compensation parameters of the auxiliary speaker, the sound field interference pattern after the main speaker and the auxiliary speaker are superimposed is simulated, and the simulation results of the optimized sound field are obtained.

[0022] Based on the simulation results of the optimized sound field and the sound field interference pattern, the formation effect and coverage of the acoustic quiet zone are verified, and the sound pressure level, phase and time delay data required to achieve the acoustic quiet zone are determined.

[0023] Furthermore, based on the sound pressure level, phase, and time delay data, a feedforward compensation database is constructed. This database stores the amplitude and phase of the rearward sound wave from the main loudspeaker, as well as the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference, including:

[0024] Based on the sound pressure level, phase, and time delay data, the amplitude and phase parameters of the rear sound wave of the main loudspeaker at each frequency are recorded.

[0025] Based on the amplitude and phase parameters of the rear acoustic wave from the main loudspeaker, the compensation signal parameters for the auxiliary loudspeaker used for destructive interference are calculated, including amplitude, phase, and time delay.

[0026] Based on the compensation signal parameters, a data storage structure indexed by frequency is constructed to form a feedforward compensation database containing the rear acoustic wave parameters of the main loudspeaker and the compensation parameters of the auxiliary loudspeaker.

[0027] The feedforward compensation database is optimized by integrating the interpolation algorithm and noise tolerance parameters to obtain a feedforward compensation database that can be queried by a digital signal processor.

[0028] Furthermore, based on the aforementioned feedforward compensation database, during the online implementation phase, a digital signal processor performs frequency decomposition on the input audio signal and queries the database to obtain the compensation parameters corresponding to each frequency component, thereby obtaining a real-time inverse compensation signal, including:

[0029] The input audio signal is decomposed into multiple frequency components.

[0030] For each frequency component, the feedforward compensation database is queried to obtain the corresponding compensation signal parameters;

[0031] Based on the corresponding compensation signal parameters, an initial compensation signal corresponding to each frequency component is obtained;

[0032] The initial compensation signals of all frequency components are synthesized to obtain the final real-time inverted compensation signal.

[0033] Furthermore, using the aforementioned real-time phase-inverting compensation signal, the main loudspeaker and auxiliary loudspeaker are synchronously driven, causing the compensated sound wave radiated by the auxiliary loudspeaker to destructively interfere with the sound wave behind the main loudspeaker, forming an acoustically quiet zone, including:

[0034] The main speaker is driven by the original audio signal to radiate directional sound waves forward and simultaneously generate rearward sound waves.

[0035] Based on the real-time anti-phase compensation signal and the backward acoustic wave characteristics generated by the main loudspeaker, the auxiliary loudspeaker is driven to radiate anti-phase compensation sound waves with approximately equal amplitude and opposite phase.

[0036] By utilizing the sound waves radiated by the main speaker and the auxiliary speaker, and through synchronous control, the two are precisely spatially superimposed in a specific area behind the main speaker, resulting in a destructive interference effect after spatial superposition.

[0037] Based on the destructive interference effect generated by the spatial superposition, a corresponding acoustic quiet zone is formed behind the main loudspeaker.

[0038] Secondly, a loudspeaker directional sound generation system based on offline simulation and feedforward control includes:

[0039] A module is built to create a high-precision three-dimensional acoustic simulation model to simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment;

[0040] The simulation module is used to perform finite element simulation using the high-precision three-dimensional acoustic simulation model, calculate the sound field distribution of the main loudspeaker in the target frequency range, and obtain the sound pressure level, phase, and time delay data of the area behind it.

[0041] The module is used to construct a feedforward compensation database based on the sound pressure level, phase and time delay data. The database stores the amplitude and phase of the rear sound wave from the main loudspeaker and the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference.

[0042] The real-time signal module is used to perform frequency decomposition on the input audio signal through a digital signal processor during the online implementation phase based on the feedforward compensation database, and query the database to obtain the compensation parameters corresponding to each frequency component, so as to obtain a real-time inverse compensation signal.

[0043] The synchronous drive module is used to synchronously drive the main speaker and the auxiliary speaker using the real-time phase-inverse compensation signal, so that the compensation sound wave radiated by the auxiliary speaker and the sound wave behind the main speaker will cause destructive interference, forming an acoustic quiet zone.

[0044] Thirdly, a computing device includes:

[0045] One or more processors;

[0046] A storage device for storing one or more programs that, when executed by one or more processors, cause the one or more processors to implement the method.

[0047] Fourthly, a computer-readable storage medium storing a program that, when executed by a processor, implements the method.

[0048] The above-described solution of the present invention has at least the following beneficial effects:

[0049] This system comprehensively integrates the physical parameters of the main and auxiliary loudspeakers with specific environmental characteristics, meticulously simulating radiation, diffraction, reflection, and interference behaviors in sound wave propagation. By precisely setting model parameters and adapting to multiple scenarios, it provides a realistic framework for subsequent data acquisition. Simultaneously, it presents sound wave propagation behavior in a model-based manner, reducing subsequent data deviations caused by missing key factors and providing complete behavioral data for finite element simulation. Finite element simulations can be performed within the target frequency range at preset step sizes, refining frequency-dimensional data acquisition and accurately capturing sound pressure level, phase, and time delay data in the area behind the main loudspeaker. By arranging a grid of measurement points in the rear area, it collects acoustic data in the spatial dimension, improving data representativeness. It can also simulate the sound field interference after the main and auxiliary loudspeakers are superimposed, obtaining interference-related data in advance, providing a comprehensive reference for subsequent compensation parameter calculations, and extracting simulation results in a structured manner for easy database construction. The system can store the corresponding rearward sound wave parameters of the main loudspeaker and the compensation parameters of the auxiliary loudspeaker. The system stores and forms a structured data set associated with frequencies, facilitating rapid subsequent queries. Frequency decomposition of the input audio signal avoids interference between different frequency components, ensuring each component can be individually matched with compensation parameters. Parameters are obtained by querying the feedforward compensation database, eliminating the need for complex online acoustic calculations and reducing the computational load on the digital signal processor. Simultaneously, it can quickly match appropriate compensation parameters for each frequency component, ensuring precise frequency correspondence between the real-time phase-inverse compensation signal and the rear acoustic wave from the main speaker. Based on the real-time phase-inverse compensation signal, the main and auxiliary speakers are synchronously driven, converting the time delay parameters obtained from the initial data processing into synchronous control commands, ensuring precise superposition of the two types of acoustic waves in a specific area behind the main speaker. The compensation acoustic wave radiated by the auxiliary speaker can specifically cancel the rear acoustic wave from the main speaker, resulting in a stable destructive interference effect. Furthermore, it translates the results of the initial data processing into actual acoustic effects, ensuring the acoustic quiet zone coverage meets the target without affecting the directional sound emission from the main speaker. Attached Figure Description

[0050] Figure 1 This is a schematic flowchart of a loudspeaker directional sound generation method based on offline simulation and feedforward control, provided by an embodiment of the present invention.

[0051] Figure 2 This is a schematic diagram of a loudspeaker directional sound generation system based on offline simulation and feedforward control, provided by an embodiment of the present invention. Detailed Implementation

[0052] Exemplary embodiments of the present disclosure will now be described in more detail with reference to the accompanying drawings. While exemplary embodiments of the present disclosure are shown in the drawings, it should be understood that the present disclosure may be implemented in various forms and should not be limited to the embodiments set forth herein. Rather, these embodiments are provided so that this disclosure will be thorough and complete, and will fully convey the scope of the disclosure to those skilled in the art.

[0053] like Figure 1 As shown, an embodiment of the present invention proposes a method for directional sound emission from a loudspeaker based on offline simulation and feedforward control. The method includes the following steps:

[0054] Step 100: Establish a high-precision three-dimensional acoustic simulation model to simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment;

[0055] Step 200: Use the high-precision three-dimensional acoustic simulation model to perform finite element simulation, calculate the sound field distribution of the main loudspeaker in the target frequency range, and obtain the sound pressure level, phase and time delay data of the area behind it;

[0056] Step 300: Based on the sound pressure level, phase and time delay data, construct a feedforward compensation database. The database stores the amplitude and phase of the rearward sound wave from the main loudspeaker and the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference.

[0057] Step 400: Based on the feedforward compensation database, during the online implementation phase, the input audio signal is frequency decomposed by a digital signal processor, and the database is queried to obtain the compensation parameters corresponding to each frequency component in order to obtain a real-time inverse compensation signal.

[0058] Step 500: Using the real-time phase-inverting compensation signal, the main speaker and the auxiliary speaker are driven synchronously, so that the compensation sound wave radiated by the auxiliary speaker and the sound wave behind the main speaker cause destructive interference, forming an acoustic quiet zone.

[0059] In this embodiment of the invention, the physical parameters of the main speaker and auxiliary speaker, as well as the characteristics of different working environments, are comprehensively incorporated to meticulously simulate the radiation, diffraction, reflection, and interference behavior of sound waves. By accurately setting parameters, dividing the grid reasonably, and setting multiple scene variations, it adapts to scenarios such as free fields and rooms with reflective surfaces, providing a realistic basis for subsequent data acquisition and providing comprehensive behavioral basis for finite element simulation, reducing data deviations caused by missing key factors in the model. Simulation is performed within the target frequency range at preset step sizes to refine frequency dimension data acquisition and capture sound pressure level, phase, and time delay data in a specific area behind the main speaker. Spatial acoustic data is systematically collected through a virtual measurement point grid to improve the spatial representativeness of the data. Sound field interference patterns are simulated in advance to obtain relevant data, providing a comprehensive reference for compensation parameter calculation and extracting simulation results in a structured manner. The rear sound wave data of the main speaker and the compensation signal parameters of the auxiliary speaker are stored accordingly to form a well-organized data set for easy and rapid online querying. Data is stored in tabular or curve format to improve readability and ease of retrieval; an integrated interpolation algorithm fills in the gaps in frequency interval parameters, and noise tolerance parameters are added to enhance the database's ability to handle minor noise interference; the input audio signal is frequency-decomposed, and compensation parameters are matched individually for each frequency component to avoid compensation deviations caused by mutual interference between different frequency components; parameters are obtained by querying the feedforward compensation database, eliminating the need for complex online acoustic calculations, reducing the real-time computation load of the digital signal processor, and improving the efficiency of compensation signal generation; the main and auxiliary speakers are driven synchronously to ensure the consistency of the time of the radiated sound waves, providing a time basis for destructive interference; the auxiliary speaker radiates sound waves according to the real-time anti-phase compensation signal, specifically canceling the sound waves behind the main speaker, efficiently forming an acoustic quiet zone; relying on the pre-existing offline database and efficient online signal processing, no additional real-time acoustic feedback adjustment is required, simplifying the online process, and the precise destructive interference does not affect the directional sound output in front of the main speaker, reducing the sound pressure in the rear area and improving the acoustic experience.

[0060] In a preferred embodiment of the present invention, step 100, establishing a high-precision three-dimensional acoustic simulation model to simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment, includes:

[0061] Step 101: Based on the physical parameters of the main speaker and auxiliary speaker and the geometric structure of the environment, establish a three-dimensional geometric model including the speakers and their spatial positional relationships. Specifically, this includes: first, collecting the physical parameters of the main speaker and auxiliary speaker, including but not limited to diaphragm diameter, cabinet volume, and material type; simultaneously, mapping the geometric structural features of the environment, such as the position and size of reflective surfaces within the environment, and the length, width, and height dimensions of the room (if it is a room environment) or the spatial range of the free field (if it is a free field environment); second, associating and integrating the collected speaker physical parameters with the environmental geometric structural features to determine the spatial position coordinates of the main speaker and auxiliary speaker in the environment, and determining the relative distance between them and their respective distances from the environmental boundary; then, using an acoustic simulation tool that supports three-dimensional modeling, converting the above physical parameters, geometric structure, and spatial positional relationships into a visualized three-dimensional geometric model to ensure that the model can fully represent the spatial relationship between the speakers and the environment.

[0062] Step 102: Based on the environment type defined by the three-dimensional geometric model, set material properties and physical parameters for the speaker diaphragm, enclosure, and air medium, and configure corresponding acoustic boundary conditions for the boundaries in the environment. The environment type includes free field, room environment, or sound-absorbing environment. Specifically, this includes: first, determining the environment type corresponding to the established three-dimensional geometric model based on its environmental form, i.e., whether it is a free field, room environment, or sound-absorbing environment; second, setting material properties and physical parameters for different components in the model. Specifically, setting the speaker diaphragm as an elastic solid material and recording its density, Poisson's ratio, and other parameters, and setting the speaker enclosure as a rigid boundary or a weakly reflective boundary. To match its actual acoustic characteristics, the sound velocity and density parameters of the air medium are entered to fit the actual medium conditions for sound wave propagation. Then, according to the determined environment type, the corresponding acoustic boundary conditions are configured for the boundaries of the environment. If it is a free field, a perfectly matched layer condition is set on the periphery of the air region to simulate the propagation of sound waves without reflection. If it is a room environment, a three-dimensional sub-model of the walls, floor and ceiling is established and rigid boundary conditions are set to simulate the reflection characteristics of the actual room. If it is a sound-absorbing environment, soft boundary conditions or impedance boundary conditions are applied in the simulation space and preset acoustic impedance or sound absorption coefficient is entered to simulate the characteristics of sound wave absorption. In this way, the material properties and boundary conditions of the model can accurately reflect the acoustic propagation basis in the actual environment.

[0063] Step 103: Based on the material properties and acoustic boundary conditions, the diaphragms of the main speaker and auxiliary speaker are defined as sound source boundaries, and their excitation conditions are set to simulate the sound generation process. Specifically, this includes: First, based on the material properties and acoustic boundary conditions set in Step 102, determining the specific position and range of the diaphragms of the main speaker and auxiliary speaker in the three-dimensional geometric model, and defining the diaphragm region as the sound source boundary to determine the source of sound waves; Second, combining the actual sound generation principle and working scenario of the speaker, setting the excitation conditions of the diaphragm as the sound source boundary. Specifically, the normal displacement parameter, vibration velocity parameter, or pressure excitation parameter of the diaphragm can be selected according to actual needs, and the set excitation parameters must fit the actual sound generation characteristics of the speaker under rated working conditions. For example, the intensity range of pressure excitation is determined by referring to the rated output power of the speaker, or the amplitude range of normal displacement is determined by referring to the maximum vibration stroke of the diaphragm; At the same time, by associating the excitation conditions with the material properties of the diaphragm, it is ensured that the excitation conditions can drive the diaphragm to simulate the actual sound generation process, so that the generation and propagation of sound waves in the subsequent simulation can correspond to the actual working state of the speaker.

[0064] Step 104 involves meshing the model with defined sound source boundaries and excitation conditions, and performing acoustic simulation within the target frequency range to obtain a high-precision three-dimensional acoustic simulation model for simulating sound wave propagation. Specifically, this includes: firstly, meshing the three-dimensional geometric model with defined sound source boundaries and excitation conditions; secondly, determining the size of the mesh cells based on the target frequency range, and calculating the maximum allowable size of the mesh cells using the sound wave wavelength corresponding to the target frequency, ensuring that the number of mesh cells within a single wavelength range is no less than 6 to 10 to guarantee calculation accuracy; and thirdly, refining the mesh in areas with significant changes in acoustic characteristics, such as the speaker surface and geometric abrupt changes at environmental boundaries, so that the mesh cell size in these key areas is smaller than in other areas, thereby improving the simulation accuracy of these areas. Calculation accuracy: After mesh generation, the target frequency range for this acoustic simulation needs to be determined. This range should cover the main operating frequency range of the loudspeaker (usually 200Hz to 4kHz), and several frequency points should be selected as simulation calculation points according to a preset frequency step size (such as 50Hz or 100Hz). Then, the frequency domain solution method is used to perform acoustic simulation calculations for each frequency point. During the solution process, the previously set material properties, boundary conditions, and excitation conditions are combined to calculate and extract data related to sound wave propagation, such as sound pressure distribution and sound field interference characteristics in the model space. Finally, the simulation results of all frequency points are integrated and verified to ensure that the model can accurately simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment, thereby obtaining a high-precision three-dimensional acoustic simulation model that meets the requirements of subsequent simulations.

[0065] In this embodiment of the invention, the physical parameters of the main speaker and auxiliary speaker, along with the geometric structure of their environment, can be transformed into a concrete three-dimensional geometric model, restoring their spatial relationship and providing a clear structural foundation for subsequent data processing steps such as setting material properties and configuring boundary conditions. Simultaneously, building the model based on actual physical and geometric information reduces subsequent data deviations caused by spatial ambiguity, ensuring the model's correspondence with the actual scene from the initial data processing stage. By setting appropriate material properties and physical parameters for the speaker diaphragm, enclosure, and air medium for different environmental types, and configuring corresponding acoustic boundary conditions for the environmental boundaries, the model's data processing capabilities can be significantly improved. The data processing is more tailored to specific application scenarios; for example, differentiated configurations for free fields, room environments, or sound-absorbing environments allow the calculation of sound wave propagation-related data in subsequent simulations to better reflect actual environmental characteristics, improving the data's adaptability to different scenarios and avoiding data distortion caused by parameter mismatch with the environment; defining the diaphragm as the sound source boundary and setting excitation conditions can determine the source data characteristics of sound wave propagation, making the input of sound source-related data in the simulation more targeted; by simulating the excitation conditions of the actual sound generation process, the simulation data of subsequent sound wave propagation behavior can be closer to the real sound generation situation, providing a reliable sound source data foundation for obtaining accurate sound field distribution data in the future;

[0066] Mesh generation of the model refines the spatial units for data calculation, especially the denser mesh at the speaker surface and geometric transitions, which improves the detail of local data calculation. Acoustic simulation within the target frequency range ensures that the solved data covers the frequency dimensions required for practical applications. The resulting high-precision three-dimensional acoustic simulation model provides comprehensive and accurate sound wave propagation simulation data for subsequent finite element simulations.

[0067] In a preferred embodiment of the present invention, step 200 above, which involves performing finite element simulation using the high-precision three-dimensional acoustic simulation model to calculate the sound field distribution of the main loudspeaker within the target frequency range and to obtain the sound pressure level, phase, and time delay data of the region behind it, includes:

[0068] Step 201: In the high-precision three-dimensional acoustic simulation model, a target area is set behind the main speaker, and multiple frequency points within the target frequency range are selected. Specifically, this includes: First, based on the simulation range and regional characteristics, the specific range of the target area behind the main speaker is determined, i.e., the cylindrical or fan-shaped area to be measured, which is 1m to 3m away from the speaker array. Combined with the actual application scenario requiring sound pressure reduction, the spatial coordinate boundary of this target area is precisely delineated to ensure complete coverage of the core diffusion area of ​​the sound waves behind the main speaker that needs to be canceled. Second, according to the parameter selection rules, the target frequency range for this finite element simulation is determined. The target frequency range is fixed at 200Hz to 4kHz, and multiple frequency points are selected with a frequency step size of 100Hz to ensure that the frequency point selection perfectly matches the parameter requirements and avoids the impact of step size deviation on the consistency of subsequent data. A virtual measurement point grid is arranged within the defined target area. The grid density must be sufficient to capture sound field data at different spatial locations within the target area at each frequency point. Finally, the spatial coordinate information of the target area and the list of selected frequency points are entered into the high-precision three-dimensional acoustic simulation model to define clear calculation boundaries and data acquisition nodes for the accurate execution of subsequent finite element simulations.

[0069] Step 202: Based on the frequency points and target area, perform finite element simulation to calculate the sound field distribution generated in the target area when only the main speaker is excited, and extract the sound pressure level, phase, and time delay data of each point in the target area as the main speaker sound field data. Specifically, this includes: First, configuring simulation parameters in a high-precision three-dimensional acoustic simulation model, activating only the excitation unit of the main speaker, keeping the auxiliary speaker in an unexcited state, and setting excitation conditions according to the actual working characteristics of the main speaker to ensure that the excitation signal conforms to the rated working parameters of the main speaker to simulate its real sound output state; Second, based on the frequency points and target area set in step 201, start the frequency domain solution process of the finite element simulation, and perform sound field calculations for each frequency point. During the solution process, call the basic parameters such as material properties and acoustic boundary conditions already configured in the model, and simultaneously quantify the diffraction effect of the sound wave and the non-uniformity of the rear sound field. The diffraction effect can be evaluated by calculating the far-field pattern to assess the sound wave bypassing the edge of the main speaker. For curved propagation, the non-uniformity of the rear sound field can be analyzed by phase gradient changes. Further, spatial distribution data is captured by defining multiple probe points, lines, or surface integration domains for the target area behind the main speaker. The sound pressure level, phase shift, and time delay data at various frequencies in these captured areas are extracted using the data analysis unit of a simulation tool. The extracted data is exported as CSV or other formats that can be used later for subsequent analysis or database construction. Finally, 2D / 3D plotting sets are generated, including sound pressure maps, sound pressure level maps, or external sound pressure level maps. By comparing the sound pressure levels in the areas before and after the main speaker, it is ensured that the sound pressure level in front is maintained at the target level (e.g., the sound pressure level measured 1 meter in front meets design requirements), while confirming that the sound pressure in the rear area shows an initial decreasing trend, providing benchmark data for subsequent auxiliary speaker parameter adjustments. Finally, the extracted sound pressure level, phase shift, and time delay data are associated and stored with the corresponding frequency points and measurement point coordinates to form a structured main speaker sound field data set.

[0070] Step 203: Based on the sound field data of the main loudspeaker, an auxiliary loudspeaker is introduced into the simulation and a sound field superposition simulation is performed. By iteratively adjusting the amplitude and phase compensation parameters of the auxiliary loudspeaker, the sound field interference pattern after the main loudspeaker and the auxiliary loudspeaker are superimposed is simulated, and the simulation results of the optimized sound field are obtained. Specifically, this includes: First, based on the sound field data of the main loudspeaker obtained in step 202, the excitation unit of the auxiliary loudspeaker is activated in the high-precision three-dimensional acoustic simulation model, and the geometric model and material property parameters of the auxiliary loudspeaker are loaded to ensure that its spatial position is consistent with the settings during the initial modeling and fully fits the constructed simulation model architecture; Second, based on the frequencies in the sound field data of the main loudspeaker... The amplitude and phase characteristics of the rearward sound wave are selected, and the initial compensation parameters of the auxiliary loudspeaker are initially set. The initial phase should initially point in the opposite direction to the phase of the rearward sound wave of the main loudspeaker. At the same time, the phase shift caused by the propagation path difference is considered in the parameter setting to lay the foundation for the subsequent destructive interference effect. On this basis, the sound field superposition simulation is started, and the main loudspeaker and the auxiliary loudspeaker are excited to perform single-frequency simulation. The total sound field after the two are superimposed is calculated, and 2D / 3D plotting groups (such as sound pressure diagram, sound pressure level diagram or external sound pressure level diagram) are generated to simulate the sound field interference pattern. Then, the destructive effect is evaluated through the interference pattern, specifically including the analysis of the total sound pressure distribution, phase difference distribution and energy attenuation.

[0071] The process then proceeds to iteratively optimize the auxiliary loudspeaker parameters. For each auxiliary loudspeaker, its amplitude and phase parameters are iteratively adjusted. During the adjustment, it is necessary to ensure that the phase difference between the rearward sound waves of the auxiliary loudspeaker and the main loudspeaker is close to 180°, while continuously considering the phase shift caused by the propagation path difference to avoid the phase deviation caused by the path difference affecting the destructive interference effect. During the parameter adjustment process, a frequency sweep operation is used to cover the frequency range of 200Hz to 4kHz set in step 201, and the parameter optimization method is combined to find the parameter combination that minimizes the objective function to achieve the minimization of the sound pressure in the target area. This objective function is used to quantify the sound pressure energy in the target quiet zone, and the formula is: where denoted by , where represents the number of measurement points in the target quiet zone, represents the position of the i-th measurement point, represents the frequency, and represent the amplitude and phase parameters of the auxiliary loudspeaker, respectively, and represents the total sound pressure level after the main loudspeaker and auxiliary loudspeaker are superimposed. After each parameter adjustment, the sound field superposition simulation is re-executed and a new sound field interference pattern is generated. The sound pressure cancellation effect of the target area under different parameter combinations is compared, and the amplitude and phase parameters of the auxiliary loudspeaker are gradually optimized. When the sound pressure level of the target area under a certain parameter combination meets the design expectation, the iteration is stopped, the amplitude and phase compensation parameters of the auxiliary loudspeaker at this time are recorded, and the optimized sound field simulation results and sound field interference pattern corresponding to this iteration are saved.

[0072] Step 204: Based on the simulation results of the optimized sound field and the sound field interference pattern, verify the formation effect and coverage of the acoustic quiet zone, and determine the sound pressure level, phase, and time delay data required to achieve the acoustic quiet zone. Specifically, this includes: analyzing the optimized sound field simulation results and sound field interference pattern obtained in step 203 using sound pressure level maps, profile maps, and probe data, focusing on verifying the sound pressure minimization effect within the target acoustic quiet zone, and evaluating interference uniformity to ensure that the quiet zone coverage is greater than 80%; secondly, combining the optimized sound field simulation results, calculate the average sound pressure level attenuation amplitude within the target area to determine whether it reaches the design threshold. If it does not reach the threshold or the interference uniformity does not meet the requirements, return to step 204. Step 203 involves readjusting the amplitude and phase parameters of the auxiliary loudspeaker and rerunning the simulation to generate an optimized dataset until the design requirements are met. After confirming that the acoustic quiet zone formation and coverage meet the specifications, key data are extracted from the results using expressions. Interpolation algorithms and noise tolerance parameters are integrated to improve the robustness of the data. Finally, the extracted key data are categorized and organized by frequency point. This data includes the sound pressure level, phase, and time delay data of the main loudspeaker in the target area at each frequency point, as well as the compensation signal parameters required by the auxiliary loudspeaker to achieve the acoustic quiet zone. This forms a parameter set directly related to the acoustic quiet zone, providing core data support for the subsequent construction of the feedforward compensation database.

[0073] In this embodiment of the invention, by setting a target area behind the main loudspeaker, the key research scope of sound wave propagation can be focused, ensuring that subsequent data acquisition and calculation always revolve around the core object of the rear sound field of the main loudspeaker, avoiding interference from data in irrelevant areas. Simultaneously, selecting multiple frequency points within the target frequency range enables full coverage of the main operating frequency range of the loudspeaker, allowing subsequent simulation data to form a complete sequence in the frequency dimension. By exciting only the main loudspeaker and performing finite element simulation, the sound field distribution data generated by the main loudspeaker in the target area can be acquired independently, avoiding interference from auxiliary loudspeaker signals on the rear sound field data of the main loudspeaker, ensuring that the extracted sound pressure level, phase, and time delay data are pure characteristics of the rear sound wave from the main loudspeaker. Furthermore, data extraction from each point within the target area forms a refined data set in the spatial dimension, providing support for subsequent analysis of the spatial distribution patterns of the rear sound field of the main loudspeaker, enabling subsequent... The calculation of auxiliary loudspeaker compensation parameters can accurately match the actual situation of the rear sound field of the main loudspeaker, improving the accuracy of data processing. By introducing the auxiliary loudspeaker into the superposition simulation based on the acquired main loudspeaker sound field data, the auxiliary loudspeaker parameters can be adjusted in a targeted manner based on the rear sound field characteristics of the main loudspeaker, ensuring that the parameter adjustments always revolve around the goal of canceling the rear sound waves of the main loudspeaker. By iteratively adjusting the amplitude and phase compensation parameters of the auxiliary loudspeaker, the superimposed sound field effect can be gradually optimized, and the dynamic optimization of compensation parameters can be achieved in the simulation stage. At the same time, the simulated sound field interference pattern can intuitively reflect the impact of parameter adjustments on the sound field superposition effect. Based on the simulation results of the optimized sound field and the sound field interference pattern, the formation effect and coverage of the acoustic quiet zone can be verified. This can confirm at the data level whether the quiet zone meets the design requirements under the current parameters, ensuring that the data used to build the feedforward compensation database has practical application value.

[0074] In a preferred embodiment of the present invention, step 300 involves constructing a feedforward compensation database based on the sound pressure level, phase, and time delay data. The database stores the amplitude and phase of the rearward sound wave from the main loudspeaker, as well as the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference, including:

[0075] Step 301: Based on the sound pressure level, phase, and time delay data, record the amplitude and phase parameters of the rear sound wave of the main loudspeaker at each frequency. Specifically, this includes: First, calling the sound pressure level, phase, and time delay data determined in step 204 that are required to achieve the acoustic quiet zone. These data have been classified and organized by frequency point and are the core basis for data recording in this step. Second, for each frequency point, extract the key characteristic parameters of the rear sound wave of the main loudspeaker from the above data. The amplitude parameter can be calculated by combining the sound pressure level data with the physical characteristics of sound wave propagation, while the phase parameter is directly taken from the phase value of the rear sound wave of the main loudspeaker recorded in step 204, ensuring that each frequency point corresponds to a complete set of amplitude and phase parameters. Then, in ascending or descending order of frequency, enter each frequency point and its corresponding amplitude and phase parameters of the rear sound wave of the main loudspeaker into a preset structured data form. The form must clearly label the frequency value, amplitude value, phase angle, and other fields to achieve accurate correlation between data and frequency, providing clear and traceable basic data support for the subsequent calculation of compensation signal parameters.

[0076] Step 302: Based on the amplitude and phase parameters of the rearward sound wave from the main loudspeaker, calculate the compensation signal parameters for the auxiliary loudspeaker used for destructive interference, including amplitude, phase, and time delay. Specifically, this includes: First, using the amplitude and phase parameters of the rearward sound wave from the main loudspeaker recorded in step 301 as a benchmark, determine the core calculation logic of the auxiliary loudspeaker's anti-phase compensation signal to ensure that the compensation signal can form effective destructive interference with the rearward sound wave from the main loudspeaker; Second, calculate the phase parameters of the compensation signal for each frequency point. Based on the basic principle of destructive interference, set the phase of the auxiliary loudspeaker's compensation signal to be opposite to the phase of the rearward sound wave from the main loudspeaker. Simultaneously, combine the time delay data obtained in step 204 to correct the time delay caused by the main loudspeaker's anti-phase interference. First, the phase shift caused by the propagation path difference of the auxiliary loudspeaker is calculated to ensure that the final phase difference meets the requirements of destructive interference. Then, the amplitude parameter of the compensation signal is calculated to match the amplitude of the auxiliary loudspeaker compensation signal with the amplitude of the rear sound wave of the main loudspeaker, so as to achieve maximum sound pressure cancellation. Finally, the time delay parameter of the compensation signal is calculated. Based on the time delay data of the rear sound wave of the main loudspeaker obtained in step 204, combined with the distance relationship between the auxiliary loudspeaker and the target area, the time delay value to be introduced by the compensation signal is determined to ensure that the compensation sound wave and the rear sound wave of the main loudspeaker can be synchronously superimposed in the target area. Through the above calculations, a set of compensation signal parameters containing amplitude, phase, and time delay is generated for each frequency point and associated with the corresponding frequency point for storage.

[0077] Step 303: Based on the compensation signal parameters, construct a data storage structure indexed by frequency to form a feedforward compensation database containing the rear acoustic wave parameters of the main speaker and the compensation parameters of the auxiliary speaker. Specifically, this includes: first, determining frequency as the core index of the data storage structure to ensure that the subsequent digital signal processor can quickly locate the required parameters through frequency. This index selection must closely align with the parameter query efficiency requirements of the online implementation phase; second, constructing a structured storage framework, setting two independent data units under each frequency index. The first data unit is specifically used to store the amplitude and phase parameters of the rear acoustic wave of the main speaker recorded in step 301, and the second data unit is specifically used to store the amplitude, phase, and time delay parameters of the auxiliary speaker compensation signal calculated in step 302. Through this dual data unit configuration under the same frequency index, the relevant parameters of the main speaker and the auxiliary speaker can be directly correlated. The association of speaker compensation parameters is established to avoid parameter mismatch during subsequent calls. A suitable structured data storage format is then selected, which can be a standardized table format or a binary data format that conforms to the reading habits of digital signal processors. All frequency indices and their corresponding data unit contents are arranged in a uniform ascending or descending order of frequency. Each parameter field within each data unit is clearly labeled with its physical unit (e.g., amplitude in Pa, phase in degrees, delay in ms) and data precision requirements (e.g., retaining two decimal places). Finally, the integrity of the constructed data storage structure is verified. The parameters in each of the two data units under each frequency index are checked one by one to ensure they are complete and that their values ​​match the output results of steps 301 and 302. After confirming that no frequency points are missing and no parameters are missing, a complete feedforward compensation database containing the associated parameters of the main speaker and auxiliary speaker is finally formed.

[0078] Step 304 involves optimizing the feedforward compensation database by integrating the interpolation algorithm and noise tolerance parameters to obtain a feedforward compensation database that can be queried by the digital signal processor. Specifically, this includes: firstly, integrating an interpolation algorithm based on the frequency coverage characteristics of the feedforward compensation database. This algorithm calculates the rear acoustic wave parameters of the main loudspeaker and the compensation parameters of the auxiliary loudspeaker corresponding to frequency points not directly simulated, based on the parameter variation patterns between adjacent frequency points, thus filling the parameter gaps between frequency intervals; secondly, introducing noise tolerance parameters, setting allowable deviation ranges for the amplitude and phase parameters of the auxiliary loudspeaker compensation signal based on the noise interference intensity in actual application scenarios. When the actual input audio signal has slight noise interference, the compensation parameters output by the database can still maintain the destructive interference effect without frequent adjustments. Furthermore, the database integrating the interpolation algorithm and noise tolerance parameters is functionally verified. By simulating audio signal inputs of different frequencies, the response speed and parameter output accuracy of the digital signal processor querying the database are tested to ensure that the database can meet the real-time query requirements of the online implementation phase. Finally, the verified database is format optimized to adapt it to the hardware interface and data reading protocol of the digital signal processor, ultimately resulting in a feedforward compensation database that can be directly queried and called by the digital signal processor.

[0079] In this embodiment of the invention, recording the amplitude and phase parameters of the rearward sound wave of the main speaker according to the frequency dimension allows for a one-to-one correspondence between data and frequency, ensuring that the characteristics of the rearward sound wave of the main speaker at each frequency point can be accurately captured. This method of organizing data by frequency provides a clear frequency reference for the subsequent calculation of compensation parameters, avoiding calculation deviations caused by unclear data-frequency relationships, and also giving the data of the rearward sound wave of the main speaker a structured characteristic. Calculating the compensation signal parameters based on the amplitude and phase parameters of the rearward sound wave of the main speaker ensures that the compensation parameters of the auxiliary speaker are highly matched with the actual characteristics of the rearward sound wave of the main speaker, ensuring that the compensation parameters can specifically address the differences in the rearward sound wave of the main speaker at different frequencies. Simultaneously acquiring the amplitude, phase, and time delay parameters of the compensation signal during the calculation process comprehensively covers the key parameters required for destructive interference. The database features several dimensions; a frequency-indexed data storage structure allows subsequent queries to quickly locate the corresponding data by frequency, improving data retrieval efficiency and meeting the data retrieval speed requirements of digital signal processors during online implementation; integrating the rear acoustic wave parameters of the main speaker and the compensation parameters of the auxiliary speaker into the same database enables associated storage of the two types of parameters, avoiding retrieval confusion caused by scattered parameter storage; integrating interpolation algorithms can fill parameter gaps between different frequency points, and even when faced with frequency components not directly recorded, appropriate compensation parameters can be obtained through interpolation calculations, avoiding compensation loss due to frequency discontinuities and improving the data coverage of the database; adding a noise tolerance parameter enhances the database's ability to cope with slight noise interference in practical applications, ensuring that the compensation parameters output by the database remain stable even with a small amount of noise.

[0080] In a preferred embodiment of the present invention, step 400, based on the feedforward compensation database, involves performing frequency decomposition on the input audio signal using a digital signal processor during the online implementation phase, and querying the database to obtain compensation parameters corresponding to each frequency component, in order to obtain a real-time inverse compensation signal, including:

[0081] Step 401 involves frequency decomposition of the input audio signal to obtain multiple frequency components. Specifically, this includes: First, during the online implementation phase, a digital signal processor (DSP) receives the original audio signal from an external source. This audio signal must meet the conventional audio format and sampling rate requirements adapted by this invention, providing a signal source that conforms to the processing standards for subsequent frequency decomposition. Second, the DSP activates its built-in frequency decomposition unit to process the input audio signal according to preset decomposition rules. The decomposition process must cover the target frequency range of 200Hz to 4kHz previously set by this invention, ensuring that the frequency components obtained by decomposition can completely match the frequency index range of the feedforward compensation database. Then, the decomposition unit decomposes the continuous audio signal into multiple discrete frequency components. Each frequency component must be labeled with its corresponding frequency value, and the amplitude ratio of each frequency component in the original audio signal must be recorded so that the actual sound intensity of the audio signal can be taken into account when generating the initial compensation signal.

[0082] Step 402: For each frequency component, query the feedforward compensation database to obtain the corresponding compensation signal parameters. Specifically, this includes: First, calling the frequency component list compiled in step 401, the digital signal processor extracts the frequency value of each frequency component sequentially according to the list order, using this frequency value as the core index for querying the feedforward compensation database. This operation relies on the previously constructed frequency-indexed storage structure of the database to ensure that the query logic is compatible with the database structure. Second, for each frequency value, the digital signal processor initiates a query request to the feedforward compensation database. If the frequency value happens to be a frequency index already stored in the database, the auxiliary loudspeaker compensation signal parameters stored under that index are directly retrieved. This includes amplitude, phase, and time delay parameters. If the frequency value lies between two adjacent frequency indices in the database, the interpolation algorithm integrated in the database is triggered. By calculating the changing trend of the compensation parameters under the two adjacent frequency indices, the compensation signal parameters corresponding to the intermediate frequency value are obtained, ensuring that even frequency components not directly stored can obtain suitable compensation parameters. Then, the validity of the queried compensation signal parameters is verified to confirm that the numerical range of the parameters meets the threshold set in the previous simulation, avoiding invalid parameters from entering the subsequent processing stage due to data transmission or query errors. Finally, each frequency component is associated with its corresponding compensation signal parameters and stored to form a correspondence table between frequency components and compensation parameters.

[0083] Step 403: Based on the corresponding compensation signal parameters, obtain the initial compensation signal corresponding to each frequency component. Specifically, this includes: First, based on the frequency component-compensation parameter correspondence table formed in step 402, the digital signal processor starts the signal generation unit for each frequency component. It needs to first read the frequency value and compensation parameter of the corresponding frequency component to determine the generation basis of the initial compensation signal; second, according to the amplitude parameter in the compensation parameters, set the signal strength of the initial compensation signal so that this strength matches the amplitude of the rearward sound wave of the main loudspeaker at the corresponding frequency, ensuring that effective destructive interference can be formed subsequently; simultaneously, according to the phase parameter in the compensation parameters, adjust the initial... The phase of the compensation signal is made to be opposite to the phase of the rearward sound wave from the main speaker, and time compensation corresponding to the time delay parameter must be incorporated to avoid phase superposition deviation due to propagation path differences. Then, the above parameters are converted into the corresponding initial compensation signal by the signal generation unit. This signal must use the same sampling rate and bit rate format as the original audio signal to ensure the signal format is consistent during subsequent synthesis processing. Finally, each generated initial compensation signal is verified. By comparing the frequency, amplitude, and phase of the signal with the corresponding compensation parameters, it is confirmed that the initial compensation signal meets the design requirements. Then, all initial compensation signals are temporarily stored in the order of their corresponding frequency components.

[0084] Step 404 involves synthesizing the initial compensation signals for all frequency components to obtain the final real-time inverted compensation signal. Specifically, this includes: first, the digital signal processor retrieves all the initial compensation signals temporarily stored in step 403, and simultaneously reads the amplitude proportion of each frequency component in the original audio signal recorded in step 401. This proportion is used as a weighting reference during signal synthesis to ensure that the synthesized compensation signal closely matches the actual sound characteristics of the original audio. Second, the signal synthesis unit is activated, and each initial compensation signal is superimposed one by one according to the frequency component order in step 401. During the superposition process, the phase synchronization of each signal is controlled to avoid distortion or interference in the synthesized signal due to phase misalignment. The signal disappears; therefore, the total amplitude and frequency coverage of the signal are monitored in real time during the synthesis process to ensure that the amplitude of the synthesized signal does not exceed the rated output range of the auxiliary speaker, and that the frequency coverage can completely cover all frequency components obtained by decomposition in step 401, without frequency breaks or signal loss; finally, the synthesized signal is finally verified, and by comparing the frequency range and phase characteristics of the signal with the target requirements, it is confirmed that the signal can be used as a real-time phase inversion compensation signal to drive the auxiliary speaker; after the verification is passed, the digital signal processor determines the synthesized signal as the final real-time phase inversion compensation signal and prepares to output it synchronously with the signal driving the main speaker, thus completing the online stage compensation signal generation process.

[0085] In this embodiment of the invention, by frequency decomposing the input audio signal, the originally continuous overall audio signal can be broken down into multiple independent frequency components, allowing subsequent data processing to focus on a single frequency dimension. This frequency-based decomposition method avoids signal interference between different frequency components, ensuring that the characteristics of each frequency component can be identified and processed individually. This lays the foundation for matching specific compensation parameters to each frequency component, while also making the data processing object clearer and reducing processing deviations caused by signal mixing. For each frequency component, the feedforward compensation database can be queried. Based on the frequency index structure already constructed in the database, the corresponding compensation signal parameters can be quickly located directly through the frequency component, eliminating the need for complex online acoustic calculations and improving the efficiency of acquiring compensation parameters. This direct frequency-parameter matching method ensures that each frequency component can obtain compensation parameters highly adapted to its own characteristics. Based on the corresponding compensation signal... The parameters generate initial compensation signals for each frequency component, ensuring that the amplitude, phase, and time delay of each initial compensation signal conform to the compensation parameter requirements. This guarantees that the initial compensation signal for each frequency component can accurately respond to the rearward sound waves from the main loudspeaker at that frequency. This direct parameter-to-signal conversion method avoids compensation signal failure caused by parameter interpretation deviations or signal generation errors. By synthesizing the initial compensation signals of all frequency components, multiple independent single-frequency compensation signals can be integrated into a complete compensation signal covering the entire frequency range of the input audio, ensuring that the final real-time inverted compensation signal can meet the rearward sound wave cancellation requirements of the entire input audio. This signal synthesis method maintains the original characteristics of each frequency component compensation signal, avoids frequency distortion or signal attenuation during the synthesis process, and ensures that the coverage of the compensation signal perfectly matches the frequency range of the input audio, ensuring effective destructive interference throughout the entire sound frequency band.

[0086] In a preferred embodiment of the present invention, step 500, which utilizes the real-time phase-inverting compensation signal to synchronously drive the main loudspeaker and the auxiliary loudspeaker, causing the compensation sound wave radiated by the auxiliary loudspeaker to destructively interfere with the sound wave behind the main loudspeaker, thus forming an acoustically quiet zone, includes:

[0087] Step 501 involves driving the main speaker based on the original audio signal, causing it to radiate directional sound waves forward and simultaneously generate rearward sound waves. Specifically, this includes: First, during the online implementation phase, the digital signal processor (DSP) receives the externally input original audio signal, which is preprocessed to match the rated operating parameters of the main speaker. Second, the DSP converts the preprocessed original audio signal into a drive signal recognizable by the main speaker. The format of this drive signal must be compatible with the hardware interface protocol of the main speaker to ensure accurate reception and response to drive commands. Then, under the action of the drive signal, the main speaker starts emitting sound, radiating directional sound waves forward that conform to the characteristics of the original audio signal. Simultaneously, due to the diffraction characteristics of sound waves, a rearward sound wave is naturally generated. The frequency range and amplitude characteristics of this rearward sound wave must be correlated with the forward directional sound wave and consistent with the rearward sound wave characteristic parameters of the main speaker recorded in step 204 above. This provides a reference sound wave standard for subsequent compensation operations of the auxiliary speaker, achieving technical integration with the previous offline simulation and online data processing stages.

[0088] Step 502, based on the real-time anti-phase compensation signal and the rearward acoustic wave characteristics generated by the main speaker, drives the auxiliary speaker to radiate anti-phase compensation sound waves with approximately equal amplitude and opposite phase. Specifically, the digital signal processor retrieves the real-time anti-phase compensation signal generated in step 404. The parameters of this signal are generated entirely based on the main speaker rearward acoustic wave characteristic reference (including amplitude and phase range) obtained by offline simulation in step 204. This reference is the inherent characteristic of the rearward sound wave of the main speaker under typical working conditions, which was determined in the early stage through high-precision three-dimensional acoustic simulation, and serves as the core basis for online compensation.

[0089] Secondly, the digital signal processor (DSP) adaptively fine-tunes the real-time phase-inverse compensation signal by combining the hardware characteristics of the main speaker and auxiliary speaker (such as rated output range and phase response deviation). The fine-tuning focuses on two dimensions: first, controlling the amplitude of the compensation signal within a range approximately equal to the amplitude of the rearward sound wave from the main speaker in the simulation benchmark (with a deviation not exceeding a preset threshold) to ensure effective cancellation; second, calibrating the phase response characteristics of the auxiliary speaker itself to ensure that the phase of the compensation signal is strictly opposite to the phase of the rearward sound wave from the main speaker in the simulation benchmark, eliminating phase shift errors caused by the hardware itself. The entire fine-tuning process strictly adheres to the upper and lower limits of the compensation parameters stored in the feedforward compensation database to avoid exceeding the physical output capacity of the auxiliary speaker and ensure equipment safety.

[0090] Then, the digital signal processor converts the finely tuned real-time inverted compensation signal into a drive signal (such as a voltage signal or a current signal) that matches the hardware interface of the auxiliary speaker, ensuring that the auxiliary speaker can accurately respond and radiate an inverted compensation sound wave that meets the expectations. The amplitude and phase characteristics of this sound wave form a targeted cancellation relationship with the rearward sound wave characteristics of the main speaker determined by offline simulation, providing a reliable signal basis for subsequent destructive interference.

[0091] In this process, the core relies on the backward acoustic wave characteristic benchmark and compensation parameters established by offline simulation in the early stage. It does not require the real-time acquisition of actual backward acoustic wave data through hardware detection unit, thus avoiding the delay problem that may be caused by real-time detection. At the same time, through hardware characteristic adaptation and fine-tuning, both compensation accuracy and system stability are taken into account.

[0092] Step 503: Utilizing the sound waves radiated by the main speaker and the auxiliary speaker, precise spatial superposition of the two in a specific area behind the main speaker is achieved through synchronous control, resulting in a destructive interference effect. Specifically, this includes: First, the digital signal processor retrieves the spatial coordinate data of the specific area behind the main speaker (a cylindrical or fan-shaped area 1 to 3 meters away from the main speaker) determined in step 204, as well as the compensation signal delay parameters recorded in step 403. Combining this with the actual installation positions of the main speaker and the auxiliary speaker, the propagation time required for the two types of sound waves to radiate from the speakers to the target area is calculated. Second, based on the difference in propagation time, the digital signal processor adjusts the timing of the sound waves from the main speaker and the auxiliary speaker... The timing of the drive signal transmission is synchronized by adjusting the transmission time of the auxiliary speaker drive signal to ensure that the anti-phase compensation sound wave and the rear sound wave of the main speaker arrive at the target area simultaneously, avoiding superposition misalignment caused by the propagation time difference. Furthermore, the digital signal processor needs to monitor the superposition state of the sound waves in the target area in real time and determine whether the superposition of the two types of sound waves is within the preset spatial range through the built-in sound field detection algorithm. If the superposition position is found to deviate from the target area, the synchronization timing of the drive signal needs to be adjusted in time until the two types of sound waves achieve precise spatial superposition in the target area, ultimately producing a destructive interference effect that meets the design requirements. This effect must be consistent with the sound field interference pattern characteristics simulated in step 203 above.

[0093] Step 504: Based on the destructive interference effect generated after spatial superposition, a corresponding acoustic quiet zone is formed behind the main speaker. Specifically, this includes: First, the digital signal processor uses a sound field detection unit to collect sound pressure level data from multiple measurement points within a specific area behind the main speaker in real time. The locations of the collected measurement points must correspond to the virtual measurement point grid positions arranged in step 201 above, ensuring the consistency and comparability of the data collection. Second, the collected actual sound pressure level data is compared with the sound pressure reduction threshold set in step 204 above (e.g., an average attenuation of 25 dB) to determine whether the sound pressure in the target area meets the "extremely low sound pressure" design standard, while simultaneously evaluating the uniformity of the sound pressure distribution. To ensure that the sound pressure at more than 80% of the measurement points within the area meets the threshold requirements, and to avoid situations where the local sound pressure is too high; further, if the sound pressure level data meets the design standards, it is necessary to further verify whether a stable acoustic quiet zone has been formed in the area. Specifically, this involves continuously monitoring the sound pressure fluctuation over a period of time (e.g., 10 seconds) to ensure that the sound pressure level in the quiet zone does not fluctuate significantly, and that the coverage of the quiet zone completely includes the specific area behind the main speaker, without affecting the directional sound waves radiated forward by the main speaker; finally, after confirming that the acoustic quiet zone has been stably formed, the digital signal processor can maintain the current driving parameters to ensure that the quiet zone effect continues to meet the design requirements, thus achieving the core technical objectives of directional sound generation and acoustic quiet zone construction of this invention.

[0094] In this embodiment of the invention, driving the main loudspeaker based on the original audio signal allows the forward directional sound wave to retain the original audio acoustic characteristics completely, meeting the requirements for directional sound generation. Simultaneously, the rearward sound wave of the main loudspeaker is clearly associated with the original audio, providing clear target characteristics for auxiliary loudspeaker compensation and avoiding parameter deviations. Driving the auxiliary loudspeaker based on a real-time inverse compensation signal accurately converts the previously calculated compensation parameters into actual sound waves, ensuring that the amplitude and phase are consistent with the calculation results. Combining the fine-tuning parameters of the rearward sound wave characteristics of the main loudspeaker ensures that the amplitudes of the two are approximately equal and their phases are opposite, preventing actual sound generation deviations from affecting the compensation effect. Synchronous control enables precise superposition of the sound waves from the main and auxiliary loudspeakers, converting the previous time delay parameters into synchronization commands, ensuring that both types of sound waves arrive simultaneously in the target area, avoiding superposition misalignment. The formation of an acoustic quiet zone based on the destructive interference effect translates all previous data processing results into actual acoustic effects, ensuring that the quiet zone sound pressure meets design requirements. Simultaneously, the quiet zone coverage area is highly matched to the target area and does not interfere with the forward directional sound wave, achieving the dual goals of directional sound generation and rear noise reduction, ultimately achieving a coexisting application effect.

[0095] like Figure 2 As shown, embodiments of the present invention also provide a loudspeaker directional sound generation system based on offline simulation and feedforward control, comprising:

[0096] A module is built to create a high-precision three-dimensional acoustic simulation model to simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment;

[0097] The simulation module is used to perform finite element simulation using the high-precision three-dimensional acoustic simulation model, calculate the sound field distribution of the main loudspeaker in the target frequency range, and obtain the sound pressure level, phase, and time delay data of the area behind it.

[0098] The module is used to construct a feedforward compensation database based on the sound pressure level, phase and time delay data. The database stores the amplitude and phase of the rear sound wave from the main loudspeaker and the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference.

[0099] The real-time signal module is used to perform frequency decomposition on the input audio signal through a digital signal processor during the online implementation phase based on the feedforward compensation database, and query the database to obtain the compensation parameters corresponding to each frequency component, so as to obtain a real-time inverse compensation signal.

[0100] The synchronous drive module is used to synchronously drive the main speaker and the auxiliary speaker using the real-time phase-inverse compensation signal, so that the compensation sound wave radiated by the auxiliary speaker and the sound wave behind the main speaker will cause destructive interference, forming an acoustic quiet zone.

[0101] It should be noted that this system is a system corresponding to the above method. All implementation methods in the above method embodiments are applicable to this embodiment and can achieve the same technical effect.

[0102] Embodiments of the present invention also provide a computing device, including: a processor and a memory storing a computer program, wherein the computer program, when executed by the processor, performs the method described above. All implementations in the above method embodiments are applicable to this embodiment and can achieve the same technical effects.

[0103] Embodiments of the present invention also provide a computer-readable storage medium storing instructions that, when executed on a computer, cause the computer to perform the method described above. All implementations in the above method embodiments are applicable to this embodiment and can achieve the same technical effects.

[0104] The above description represents the preferred embodiments of the present invention. It should be noted that those skilled in the art can make various improvements and modifications without departing from the principles of the present invention, and these improvements and modifications should also be considered within the scope of protection of the present invention.

Claims

1. A method for speaker directional sound production based on offline simulation and feedforward control, characterized in that, The method includes: A high-precision three-dimensional acoustic simulation model is established to simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment. This includes: establishing a three-dimensional geometric model containing the loudspeakers and their spatial relationships based on the physical parameters of the main loudspeakers and auxiliary loudspeakers and the geometry of their environment; setting material properties and physical parameters for the loudspeaker diaphragms, enclosure, and air medium based on the environment type defined by the three-dimensional geometric model, and configuring corresponding acoustic boundary conditions for the boundaries in the environment, wherein the environment type includes free field, room environment, or sound-absorbing environment; defining the diaphragms of the main loudspeakers and auxiliary loudspeakers as sound source boundaries based on the material properties and acoustic boundary conditions, and setting their excitation conditions to simulate the sound generation process; meshing the model with the set sound source boundaries and excitation conditions, and performing acoustic simulation solutions within the target frequency range to obtain the high-precision three-dimensional acoustic simulation model required to simulate sound wave propagation behavior. Finite element simulation is performed using the high-precision three-dimensional acoustic simulation model to calculate the sound field distribution of the main loudspeaker within the target frequency range and obtain the sound pressure level, phase, and time delay data of the area behind it. This includes: setting a target area behind the main loudspeaker in the high-precision three-dimensional acoustic simulation model and selecting multiple frequency points within the target frequency range; performing finite element simulation based on the frequency points and the target area to calculate the sound field distribution generated in the target area when only the main loudspeaker is excited, and extracting the sound pressure level, phase, and time delay data of each point within the target area as the main loudspeaker's sound field data; introducing an auxiliary loudspeaker into the simulation based on the main loudspeaker's sound field data and performing sound field superposition simulation; iteratively adjusting the amplitude and phase compensation parameters of the auxiliary loudspeaker to simulate the sound field interference pattern after the main loudspeaker and auxiliary loudspeaker are superimposed, and obtaining the simulation results of the optimized sound field; and verifying the formation effect and coverage of the acoustic quiet zone based on the simulation results of the optimized sound field and the sound field interference pattern, and determining the sound pressure level, phase, and time delay data required to achieve the acoustic quiet zone. Based on the sound pressure level, phase, and time delay data, a feedforward compensation database is constructed. This database stores the amplitude and phase of the rearward sound wave from the main loudspeaker, as well as the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference. This includes: recording the amplitude and phase parameters of the rearward sound wave from the main loudspeaker at each frequency based on the sound pressure level, phase, and time delay data; calculating the compensation signal parameters for destructive interference from the auxiliary loudspeaker based on the amplitude and phase parameters of the rearward sound wave from the main loudspeaker, including amplitude, phase, and time delay; constructing a frequency-indexed data storage structure based on the compensation signal parameters to form a feedforward compensation database containing the rearward sound wave parameters of the main loudspeaker and the compensation parameters of the auxiliary loudspeaker; and optimizing the feedforward compensation database by integrating interpolation algorithms and noise tolerance parameters to obtain a feedforward compensation database that can be queried by a digital signal processor. Based on the feedforward compensation database, during the online implementation phase, the input audio signal is frequency decomposed by a digital signal processor, and the database is queried to obtain the compensation parameters corresponding to each frequency component in order to obtain a real-time inverse compensation signal. Using the real-time phase-inverse compensation signal, the main loudspeaker and the auxiliary loudspeaker are driven synchronously, so that the compensation sound wave radiated by the auxiliary loudspeaker and the sound wave behind the main loudspeaker undergo destructive interference, forming an acoustic quiet zone.

2. The loudspeaker directional sound production method based on offline simulation and feedforward control according to claim 1, characterized in that, Based on the aforementioned feedforward compensation database, during the online implementation phase, a digital signal processor performs frequency decomposition on the input audio signal and queries the database to obtain the compensation parameters corresponding to each frequency component, thereby obtaining a real-time inverse compensation signal, including: The input audio signal is decomposed into multiple frequency components. For each frequency component, the feedforward compensation database is queried to obtain the corresponding compensation signal parameters; Based on the corresponding compensation signal parameters, an initial compensation signal corresponding to each frequency component is obtained; The initial compensation signals of all frequency components are synthesized to obtain the final real-time inverted compensation signal.

3. The loudspeaker directional sound generation method based on offline simulation and feedforward control according to claim 2, characterized in that, Using the aforementioned real-time phase-inverse compensation signal, the main loudspeaker and auxiliary loudspeaker are synchronously driven, causing the compensated sound wave radiated by the auxiliary loudspeaker to destructively interfere with the sound wave behind the main loudspeaker, forming an acoustically quiet zone, including: The main speaker is driven by the original audio signal, causing it to radiate directional sound waves forward and simultaneously generate rearward sound waves. Based on the real-time anti-phase compensation signal and the backward acoustic wave characteristics generated by the main loudspeaker, the auxiliary loudspeaker is driven to radiate anti-phase compensation sound waves with equal amplitude and opposite phase. By utilizing the sound waves radiated by the main speaker and the auxiliary speaker, and through synchronous control, the two are precisely spatially superimposed in a specific area behind the main speaker, resulting in a destructive interference effect after spatial superposition. Based on the destructive interference effect generated by the spatial superposition, a corresponding acoustic quiet zone is formed behind the main loudspeaker.

4. A loudspeaker directional sound generation system based on offline simulation and feedforward control, wherein the system implements the method as described in any one of claims 1 to 3, characterized in that, include: A module is built to create a high-precision three-dimensional acoustic simulation model to simulate the sound wave propagation behavior of the main loudspeaker and auxiliary loudspeaker in a specific environment; The simulation module is used to perform finite element simulation using the high-precision three-dimensional acoustic simulation model, calculate the sound field distribution of the main loudspeaker in the target frequency range, and obtain the sound pressure level, phase, and time delay data of the area behind it. The construction module is used to construct a feedforward compensation database based on the sound pressure level, phase and time delay data. The database stores the amplitude and phase of the rear sound wave of the main loudspeaker and the compensation signal parameters required by the auxiliary loudspeaker to achieve destructive interference. The real-time signal module is used to perform frequency decomposition on the input audio signal through a digital signal processor during the online implementation phase based on the feedforward compensation database, and query the database to obtain the compensation parameters corresponding to each frequency component, so as to obtain a real-time inverse compensation signal. The synchronous drive module is used to synchronously drive the main speaker and the auxiliary speaker using the real-time phase-inverse compensation signal, so that the compensation sound wave radiated by the auxiliary speaker and the sound wave behind the main speaker will cause destructive interference, forming an acoustic quiet zone.

5. A computing device, characterized in that, include: One or more processors; A storage device for storing one or more programs, which, when executed by one or more processors, cause the one or more processors to implement the method as described in any one of claims 1 to 3.

6. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores a program that, when executed by a processor, implements the method as described in any one of claims 1 to 3.

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