Loudspeaker array beam forming method, electronic equipment and storage medium

Abstract

The invention discloses a loudspeaker array beam forming method, electronic equipment and a storage medium, and relates to the technical field of beam forming, and the method comprises the steps: obtaining a distance value between each loudspeaker unit in a loudspeaker array and a corresponding impedance boundary, and an impedance value of the impedance boundary; matching a target beam forming filter in a plurality of preset beam forming filters according to the distance value and the impedance value; and filtering the input signal of the corresponding loudspeaker unit based on each target beam forming filter to form an output beam. In practical application, the applicable beam forming filter can be quickly matched only by acquiring the specific impedance boundary parameters of the current environment, the method can adapt to the boundary environment with limited acoustic impedance characteristics, the influence of the impedance boundary on sound wave propagation is accurately reflected, the beam pointing precision and the acoustic contrast are remarkably improved, and the method is suitable for popularization and application. And the grating lobes of the dark area are reduced, so that the optimization of the beam performance is realized on the premise of not needing on-site acoustic measurement, and the universality and the accuracy are both considered.

Description

Technical Field

[0001] This application relates to the field of beamforming technology, and in particular to a method for beamforming a loudspeaker array, an electronic device, and a storage medium. Background Technology

[0002] Loudspeaker array beamforming technology is widely used in home theaters, large stadium sound reproduction, and other scenarios to achieve directional sound field control and spatial sound reproduction. However, most current beamforming methods are based on the free field assumption, ignoring the influence of impedance boundary reflections on sound wave propagation in the actual environment. This leads to a decrease in beamforming performance in loudspeaker arrays actually placed near impedance boundaries such as walls. The root cause lies in the mismatch between the transfer function used in the design and the real environment.

[0003] To address the aforementioned mismatch issue, the main approach currently is to directly obtain the transfer function through on-site acoustic measurements. However, while this method yields good beamforming results, it is time-consuming and impractical for consumer-grade products. Another approach utilizes the transfer function of the speaker array in a free field to deduce the transfer function in an approximate room environment, thereby optimizing beamforming. However, while this method is applicable to consumer-grade products, its derivation process relies on the assumption of rigid boundaries, thus failing to accurately handle reflections from finite acoustic impedance boundaries such as TV background walls. Summary of the Invention

[0004] The main objective of this application is to provide a speaker array beamforming method, electronic device, and storage medium, aiming to solve the technical problem that current speaker beamforming schemes in real-world environments cannot simultaneously achieve both versatility and accuracy.

[0005] To achieve the above objectives, this application proposes a loudspeaker array beamforming method, which includes: Obtain the distance between each speaker unit in the speaker array and its corresponding impedance boundary, as well as the impedance value of the impedance boundary; The target beamforming filter corresponding to each speaker unit is matched among a plurality of preset beamforming filters based on the distance value and the impedance value, wherein each preset beamforming filter is determined under different impedance boundary conditions. The input signal of the corresponding loudspeaker unit is filtered by each of the target beamforming filters to form the output beam of the loudspeaker array.

[0006] In one embodiment, each of the preset beamforming filters is associated with and stored with a corresponding preset distance value and a preset impedance value. The step of matching the target beamforming filter corresponding to each loudspeaker among the multiple preset beamforming filters according to the distance value and the impedance value includes: Based on the distance value and the impedance value, the associated target preset distance value and target preset impedance value are determined by matching them among multiple associated preset distance values ​​and preset impedance values; The preset beamforming filter corresponding to the target preset distance value and the target preset impedance value is used as the target beamforming filter.

[0007] In one embodiment, before the step of matching the target beamforming filter corresponding to each loudspeaker among a plurality of preset beamforming filters based on the distance value and the impedance value, the method further includes: Acquire the sound pressure data of each speaker unit at the corresponding control point in the free field, wherein the free field has no impedance boundary; The spherical harmonic function expansion coefficients of the corresponding loudspeaker unit are determined based on the sound pressure data. For any loudspeaker unit, multiple transfer function matrices of the loudspeaker unit in different half-spaces are determined according to the spherical harmonic function expansion coefficients of the loudspeaker unit, wherein any half-space corresponds to a combination of a preset distance value and a preset impedance value; Each of the transfer function matrices is converted into a corresponding beamforming filter to obtain each of the preset beamforming filters.

[0008] In one embodiment, the step of determining the multiple transfer function matrices of the loudspeaker unit in different half-spaces based on the spherical harmonic expansion coefficients of the loudspeaker unit includes: In a half-space under any combination of preset distance and preset impedance values, the spherical wave function of the half-space is determined based on the preset distance and preset impedance values. Based on the half-space spherical wave function, the expansion coefficients of the spherical harmonic function, and the preset truncation order of the spherical harmonic expansion term corresponding to the expansion coefficients of the spherical harmonic function, the half-space sound pressure data of the loudspeaker unit at the corresponding control point in the half-space is determined. The transfer function matrix of the loudspeaker unit is constructed by traversing each control point corresponding to the loudspeaker unit and based on the half-space sound pressure data at each control point.

[0009] In one embodiment, the step of determining the half-space spherical wavefunction based on the preset distance value and the preset impedance value includes: Obtain the first spherical coordinates and the second spherical coordinates of the control point corresponding to the speaker unit, wherein the first spherical coordinates are located in a spherical coordinate system with the speaker unit as the center, and the second spherical coordinates are located in a spherical coordinate system with the mirror speaker unit as the center. The mirror speaker unit and the speaker unit are distributed with the impedance boundary in the half space as the plane of symmetry. The relative positional relationship between the first spherical coordinates and the second spherical coordinates is determined by the preset distance value. The complex angle of the control point in the half-space is determined based on the preset impedance value; The wall loss factor and plane wave reflection coefficient of the impedance boundary in the half-space are determined based on the preset impedance value and the second spherical coordinates. The spherical wave function of the real sound source is determined based on the first spherical coordinates; The first component of the spherical wave function of the mirror sound source is determined based on the second spherical coordinates, and the second component of the spherical wave function of the mirror sound source is determined based on the second spherical coordinates, the complex angle, the wall loss factor, and the plane wave reflection coefficient. The half-space spherical wave function is determined based on the real sound source spherical wave function, the first component, and the second component.

[0010] In one embodiment, the step of determining the spherical wavefunction of the true sound source based on the first spherical coordinates includes: The first and second types of Hankel functions are determined based on the radial distance in the first spherical coordinates; The first spherical harmonic function is determined based on the polar angle and azimuth angle in the first spherical coordinate system. The true sound source spherical wave function is determined based on the first and second type Hankel functions and the first spherical harmonic function.

[0011] In one embodiment, the step of determining the first component of the spherical wave function of the mirror sound source based on the second spherical coordinates, and determining the second component of the spherical wave function of the mirror sound source based on the second spherical coordinates, the wall loss factor, the plane wave reflection coefficient, and the complex angle includes: The second type of Hankel function is determined based on the radial distance in the second spherical coordinate system. The second spherical harmonic function is determined based on the polar angle and azimuth angle in the second spherical coordinate system, and the third spherical harmonic function is determined based on the complex angle and the azimuth angle in the second spherical coordinate system. The first component of the spherical wave function of the mirror sound source is determined based on the second type of Hankel function and the second spherical harmonic function. Based on the plane wave reflection coefficient, the second type Hankel function, the second spherical harmonic function, the third spherical harmonic function, and the wall loss factor, the second component of the spherical wave function of the mirror sound source is determined.

[0012] In one embodiment, the step of converting each of the transfer function matrices into corresponding beamforming filters to obtain each of the preset beamforming filters includes: For any given transfer function matrix, the beamforming filter corresponding to the transfer function matrix is ​​determined based on the transfer function matrix, the conjugate transpose of the transfer function matrix, the regularization control matrix corresponding to the transfer function matrix, and the target sound pressure corresponding to the transfer function matrix. By traversing each of the transfer function matrices, each of the preset beamforming filters is obtained.

[0013] In addition, to achieve the above objectives, this application also proposes an electronic device, the device comprising: a memory, a processor, and a computer program stored in the memory and executable on the processor, the computer program being configured to implement the steps of the loudspeaker array beamforming method as described above.

[0014] In addition, to achieve the above objectives, this application also proposes a storage medium, which is a computer-readable storage medium, on which a computer program is stored, and when the computer program is executed by a processor, it implements the steps of the loudspeaker array beamforming method described above.

[0015] One or more technical solutions proposed in this application have at least the following technical effects: This application effectively solves the problem that current beamforming methods cannot simultaneously achieve both versatility and accuracy by employing a technical solution that involves obtaining the distance between each speaker unit in a speaker array and its corresponding impedance boundary, as well as the impedance value of the impedance boundary, and matching a target beamforming filter among multiple preset beamforming filters based on these parameters. Finally, the input signal is filtered based on the matched beamforming filter to form an output beam.

[0016] Specifically, this technical solution pre-stores multiple preset beamforming filters for different impedance boundary conditions. In practical applications, only the specific impedance boundary parameters of the current environment are needed to quickly match the appropriate beamforming filter, thus avoiding the transfer function mismatch problem caused by relying on free-field assumptions or rigid boundary models in traditional methods. This solution not only adapts to boundary environments with finite acoustic impedance characteristics, such as TV background walls, improving the versatility of beamforming in various practical room scenarios, but also accurately reflects the impact of impedance boundaries on sound wave propagation by obtaining specific impedance boundary parameters, significantly improving beam pointing accuracy and acoustic contrast, and reducing dark area lobes. Thus, it optimizes beam performance without the need for on-site acoustic measurements, balancing versatility and accuracy. Attached Figure Description

[0017] The accompanying drawings, which are incorporated in and form part of this specification, illustrate embodiments consistent with this application and, together with the description, serve to explain the principles of this application.

[0018] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, for those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0019] Figure 1 This is a flowchart illustrating an embodiment of the loudspeaker array beamforming method of this application. Figure 2 This is a flowchart illustrating Embodiment 2 of the loudspeaker array beamforming method of this application; Figure 3 This is a schematic diagram of the spatial distribution of the loudspeaker unit and the mirror loudspeaker unit in the loudspeaker array beamforming method provided in Embodiment 2 of this application; Figure 4 This is a schematic diagram showing the geometric relationship between the loudspeaker unit and the mirror loudspeaker unit in the loudspeaker array beamforming method provided in Embodiment 2 of this application; Figure 5 This is a schematic diagram of a bright and dark area scene of the loudspeaker array beamforming method provided in Embodiment 2 of this application; Figure 6 This is a beam diagram of the loudspeaker array beamforming method provided in Embodiment 2 of this application; Figure 7 This is another beam diagram of the loudspeaker array beamforming method provided in Embodiment 2 of this application; Figure 8 This is a schematic diagram of acoustic contrast comparison of the loudspeaker array beamforming method provided in Embodiment 2 of this application; Figure 9 This is a schematic diagram of the device structure of the hardware operating environment involved in the speaker array beamforming method in the embodiments of this application.

[0020] The purpose, features, and advantages of this application will be further explained in conjunction with the embodiments and with reference to the accompanying drawings. Detailed Implementation

[0021] It should be understood that the specific embodiments described herein are merely illustrative of the technical solutions of this application and are not intended to limit this application.

[0022] To better understand the technical solution of this application, a detailed description will be provided below in conjunction with the accompanying drawings and specific implementation methods.

[0023] The main solution of this application embodiment is: to obtain the distance value between each speaker unit in the speaker array and the corresponding impedance boundary, and the impedance value of the impedance boundary; to match the target beamforming filter corresponding to each speaker unit among a plurality of preset beamforming filters according to the distance value and the impedance value, wherein each preset beamforming filter is determined under different impedance boundary conditions; and to filter the input signal of the corresponding speaker unit based on each target beamforming filter to form the output beam of the speaker array.

[0024] Because most current beamforming methods are based on the free-field assumption, they neglect the impact of impedance boundary reflections on sound wave propagation in real-world environments. This leads to a decline in beamforming performance in speaker arrays placed near impedance boundaries such as walls. The root cause lies in the mismatch between the transfer function used in the design and the actual environment. To address this mismatch, the main approach is to directly obtain the transfer function through on-site acoustic measurements. However, while this method produces good beamforming results, it is time-consuming and impractical for consumer products. Another approach uses the transfer function of the speaker array in a free field to calculate the transfer function in an approximate room environment, thereby optimizing beamforming. However, while this method is applicable to consumer products, its derivation relies on the assumption of rigid boundaries, thus failing to accurately handle reflections from finite acoustic impedance boundaries such as TV background walls.

[0025] This application provides a solution that pre-stores multiple preset beamforming filters for different impedance boundary conditions. In practical applications, only the specific impedance boundary parameters of the current environment are needed to quickly match the appropriate beamforming filter, thus avoiding the transfer function mismatch problem caused by reliance on free-field assumptions or rigid boundary models in traditional methods. This solution not only adapts to boundary environments with finite acoustic impedance characteristics, such as TV background walls, improving the versatility of beamforming in various practical room scenarios, but also accurately reflects the influence of impedance boundaries on sound wave propagation by obtaining specific impedance boundary parameters, significantly improving beam pointing accuracy and acoustic contrast, and reducing dark area grating lobes. Thus, beam performance optimization is achieved without the need for on-site acoustic measurements, balancing versatility and accuracy.

[0026] It should be noted that the executing entity in this embodiment can be a computing service device with data processing, network communication, and program execution functions, such as a tablet computer, personal computer, or mobile phone, or an electronic device capable of performing the above functions. The following description uses an electronic device as an example to illustrate this embodiment and the subsequent embodiments.

[0027] Based on this, embodiments of this application provide a method for beamforming a loudspeaker array, referring to... Figure 1 , Figure 1 This is a schematic flowchart of the first embodiment of the loudspeaker array beamforming method of this application.

[0028] In this embodiment, the loudspeaker array beamforming method includes steps S10 to S30: Step S10: Obtain the distance between each speaker unit in the speaker array and its corresponding impedance boundary, as well as the impedance value of the impedance boundary; It should be noted that impedance boundaries refer to physical surfaces in a room environment that affect sound wave propagation, such as walls, TV background walls, or ceilings. These surfaces have specific acoustic impedance characteristics, capable of reflecting or absorbing sound wave energy. Unlike ideal free fields or rigid boundaries, impedance boundaries have finite acoustic impedance, altering the phase and amplitude of sound waves, thus affecting the beamforming effect of the speaker array. In this embodiment, the impedance boundary is the boundary that the speaker array approaches during actual deployment. Its characteristics need to be accurately modeled to address transfer function mismatch issues and ensure that beamforming performance remains consistent across various real-world environments.

[0029] The distance value is the linear physical distance between each speaker unit in the speaker array and its corresponding impedance boundary. This value quantifies the spatial spacing between the speaker unit and the boundary surface. This distance value directly affects the sound wave reflection path and sound field distribution at the boundary. For example, when a speaker unit is close to a wall, reflected sound will superimpose with direct sound, causing beam distortion. Obtaining the distance value is to accurately describe the geometric relationship between the sound source and the boundary during beamforming, thereby compensating for the influence of boundary reflections on beam control and improving beam pointing accuracy.

[0030] Impedance value represents the acoustic impedance characteristics of an impedance boundary. This value is a physical parameter describing the degree to which the boundary resists sound waves, including resistive and reactive components. It can be obtained through measurement or modeling, for example, using the impedance tube method. The impedance value determines the boundary's reflection coefficient and absorption behavior of incident sound waves. For example, non-rigid boundaries such as TV background walls have specific impedance values, which affect the energy distribution and phase of sound waves. In this embodiment, the impedance value is used to characterize the acoustic behavior of the impedance boundary so that these practical effects can be incorporated into the beamforming filter, avoiding beam performance degradation due to inaccurate assumptions about the impedance boundary.

[0031] Step S20: Match the target beamforming filter corresponding to each speaker unit among a plurality of preset beamforming filters according to the distance value and the impedance value, wherein each preset beamforming filter is determined under different impedance boundary conditions. It should be noted that the preset beamforming filters are a set of pre-calculated and stored digital filters. Each filter is designed for different impedance boundary conditions and is used to process the input signal of the speaker unit to form the desired output beam. These filters are generated offline, covering a variety of possible environmental scenarios, allowing the system to quickly call them up in practical applications without the need for complex real-time acoustic measurements or optimizations. By using preset beamforming filters, the efficiency and practicality of beamforming can be improved, while ensuring that beam performance is maintained under different boundary conditions.

[0032] Impedance boundary conditions can be a specific set of environmental parameters defined by a combination of distance and impedance values. These conditions describe the geometric and acoustic relationship between the loudspeaker array and the impedance boundary, including the boundary's location, acoustic characteristics, and its impact on the sound field. Each impedance boundary condition corresponds to a unique sound wave propagation scenario, such as walls at different distances or boundaries of different materials. These conditions determine the behavior of sound wave reflection and diffraction. In this embodiment, impedance boundary conditions are used to distinguish and generate multiple preset beamforming filters, enabling the system to adapt to diverse real-world room environments and improving the versatility and accuracy of beamforming.

[0033] The target beamforming filter is a specific filter selected from multiple preset beamforming filters. This filter is matched based on the currently acquired distance and impedance values ​​and applied to the input signal of the corresponding speaker unit to generate the output beam. This filter ensures that beamforming is consistent with the current impedance boundary conditions by mapping actual environmental parameters to a pre-stored filter library, thereby optimizing beam directivity and acoustic contrast. Using the target beamforming filter allows for dynamic adaptation to environmental changes, solving the beam performance degradation problem caused by environmental mismatch in traditional methods, and achieving fast and precise beam control.

[0034] Understandably, the influence of impedance boundaries on sound wave propagation in real-world room environments can lead to transfer function mismatch, resulting in decreased beam performance. Traditional methods cannot adapt to various impedance boundary conditions while avoiding on-site measurements. Therefore, this embodiment pre-calculates and stores multiple preset beamforming filters for different impedance boundary conditions. In practical applications, it quickly matches the target beamforming filter based on real-time distance and impedance values. This avoids the inaccuracy of the transfer function caused by free-field or rigid boundary assumptions, as well as the impracticality and time-consuming nature of on-site acoustic measurements. Ultimately, it improves the versatility and accuracy of beamforming in various real-world environments. By precisely matching environmental parameters, it optimizes beam directivity and acoustic contrast, reduces dark area grating lobes, and thus ensures rapid and stable performance optimization of the speaker array beam without the need for complex measurements.

[0035] In one specific implementation, each of the preset beamforming filters is associated with and stored with a corresponding preset distance value and preset impedance value. Step S20 may include steps S21 to S22: Step S21: Based on the distance value and the impedance value, match them among multiple associated preset distance values ​​and preset impedance values ​​to determine the associated target preset distance value and target preset impedance value; It should be noted that the associated target preset distance and target preset impedance values ​​refer to the specific set of preset distance and preset impedance values ​​that are perfectly matched by comparing the actual acquired distance and impedance values ​​with the preset distance and preset impedance values ​​in the list of multiple preset beamforming filters that are stored in advance. This set of parameters corresponds exactly to the current actual environmental parameters and is the boundary condition on which the target beamforming filter was calculated in advance. Its function is to serve as a unique index to directly retrieve and call the precisely associated preset beamforming filter from the stored filter library, thereby ensuring that the matched filter is the result of precise optimization for this specific impedance boundary condition.

[0036] Step S22: Use the preset beamforming filter corresponding to the target preset distance value and the target preset impedance value as the target beamforming filter.

[0037] Understandably, inaccurate matching mechanisms can lead to inefficient or inaccurate matching processes in practical applications. Therefore, this implementation method associates and stores each preset beamforming filter with its corresponding preset distance and impedance values. Based on the obtained actual distance and impedance values, it directly matches these associated preset parameters to determine the target preset distance and impedance values, and then uses the corresponding preset beamforming filter as the target beamforming filter. This effectively avoids selection errors or calculation delays that may be caused by unclear matching logic, and achieves fast, accurate, and reliable retrieval of the target beamforming filter. This ensures that the beamforming system can respond to changes in specific environmental parameters in real time and accurately, thereby maintaining low computational complexity while steadily improving the beamforming performance of the loudspeaker array in actual impedance boundary environments.

[0038] In another specific implementation, a preset beamforming filter is pre-calculated and stored for a set of discrete distance and impedance value combinations, forming a filter library. When the actual distance and impedance values ​​are obtained, the system uses a bilinear interpolation algorithm to find multiple preset beamforming filters in the filter library that are closest to the current parameters. It then performs a weighted calculation based on the relative positions of the actual distance and impedance values ​​with these neighboring preset parameters, thereby generating a target beamforming filter that adapts to the current impedance boundary conditions, rather than directly matching the stored preset values. This implementation achieves coverage of the continuous parameter space through interpolation, avoiding matching errors caused by the discretization of preset parameters. This allows for smoother adjustment of the beamforming effect when environmental parameters change, improving the accuracy and stability of beam pointing.

[0039] Step S30: Filter the input signal of the corresponding loudspeaker unit based on each of the target beamforming filters to form the output beam of the loudspeaker array.

[0040] For example, firstly, in a laboratory environment, multiple preset beamforming filters are pre-calculated offline using beamforming algorithms such as sound pressure matching for a series of discrete distance values ​​(e.g., 0.1m, 0.2m, 0.3m) and impedance values ​​(e.g., complex impedance corresponding to a specific sound-absorbing material). These filters are then stored in the built-in memory of the speaker array product to form a filter library. During actual deployment, the user inputs the actual distance between the speaker array and the TV background wall via a mobile device and selects the corresponding impedance value from several preset room boundary materials (e.g., plasterboard, glass, wood). The system then searches the filter library based on the input distance and impedance values, directly selecting the preset beamforming filter with identical preset parameters as the target beamforming filter. Finally, the system loads this target beamforming filter into the digital signal processor, performing real-time convolution operations on the input audio signals of each speaker unit to generate a precisely directional output beam, effectively overcoming the beam distortion problem caused by reflections from finite impedance boundaries such as the TV background wall.

[0041] This embodiment provides a speaker array beamforming method. By pre-storing multiple preset beamforming filters for different impedance boundary conditions, in practical applications, only the specific impedance boundary parameters of the current environment need to be obtained to quickly match the appropriate beamforming filter. This avoids the transfer function mismatch problem caused by relying on free-field assumptions or rigid boundary models in traditional methods. This solution can not only adapt to boundary environments with finite acoustic impedance characteristics, such as TV background walls, improving the versatility of beamforming in various practical room scenarios, but also accurately reflects the influence of impedance boundaries on sound wave propagation by obtaining specific impedance boundary parameters, significantly improving beam pointing accuracy and acoustic contrast, and reducing dark area grating lobes. Thus, beam performance optimization is achieved without the need for on-site acoustic measurements, balancing versatility and accuracy.

[0042] Based on the first embodiment of this application, in the second embodiment of this application, the content that is the same as or similar to that in Embodiment 1 above can be referred to the above description, and will not be repeated hereafter. Based on this, please refer to... Figure 2 Before step S20, the loudspeaker array beamforming method further includes steps S01 to S04: Step S01: Obtain the sound pressure data of each speaker unit at the corresponding control point in the free field, wherein the free field has no impedance boundary; It should be noted that a free field is an ideal acoustic environment in which sound waves can propagate freely outward without reflection or refraction, and there are no boundaries (such as walls or ceilings) that cause reflection or absorption effects on the sound waves. In this embodiment, the free field specifically refers to the experimental environment used for preliminary measurements of the speaker unit. Its purpose is to obtain acoustic data that is only related to the physical characteristics of the speaker unit itself and is not disturbed by the room environment, as the basis for subsequent calculations.

[0043] Control points are pre-defined discrete locations in the sound field space used to analyze sound pressure distribution. These control points are typically regularly distributed around the speaker unit in the free field space. Their function is to serve as sampling locations for sound pressure data. By measuring or calculating sound pressure at these points, the radiated sound field pattern of the speaker unit in space can be captured, providing data support for sound field reconstruction and modeling.

[0044] Sound pressure data, on the other hand, are sound pressure values ​​measured or calculated at various control points in a free-field environment. These data are quantified values ​​of the pressure fluctuations generated at a specified point when sound waves propagate in the air. They reflect the sound field intensity and phase information generated by the loudspeaker unit at that point and serve as the raw input data for sound field analysis and calculation of spherical harmonic function expansion coefficients.

[0045] Step S02: Determine the spherical harmonic function expansion coefficients of the corresponding loudspeaker unit based on the sound pressure data. It should be noted that the spherical harmonic function expansion coefficients are a set of mathematical coefficients used to characterize the sound field characteristics of a sound source in spherical coordinates. They are a set of core parameters obtained by combining sound pressure data measured in a free field with inversion methods such as singular value decomposition in near-field sound holography and other sound field reconstruction techniques. These coefficients can completely describe the complex directivity characteristics of the loudspeaker unit in a free field and are a key bridge for accurately predicting its sound field from a free field to a bounded half-space environment.

[0046] Step S03: For any loudspeaker unit, determine multiple transfer function matrices of the loudspeaker unit in different half-spaces based on the spherical harmonic function expansion coefficients of the loudspeaker unit, wherein any half-space corresponds to a combination of a preset distance value and a preset impedance value; It should be noted that a half-space is an acoustic space model divided by an infinitely large planar boundary, which represents the impedance boundary (such as a wall) in a room. In this embodiment, the half-space specifically refers to the semi-infinite space on one side of the speaker unit after it is divided by the impedance boundary during actual deployment. This model, by introducing boundary conditions, allows the sound field calculation to take into account the effects of boundary reflection and absorption in the real environment, making it an acoustic model that is closer to the actual application scenario than a free field.

[0047] The transfer function matrix is ​​a mathematical matrix in which each element represents the acoustic transfer function from a loudspeaker unit to a control point. It represents the set of acoustic transfer relationships between all loudspeaker units and all control points under a specific half-space environment (defined by a set of preset distance and impedance values). This matrix accurately describes the system response from electrical signal input to spatial sound pressure output under the boundary conditions of that half-space, and is the acoustic model data necessary for the final calculation of the beamforming filter.

[0048] Step S04: Convert each of the transfer function matrices into a corresponding beamforming filter to obtain each of the preset beamforming filters.

[0049] Understandably, given the extreme complexity of the sound field response of a single loudspeaker unit in a real impedance boundary environment, directly measuring the transfer function matrix under each boundary condition is impractical. Furthermore, calculations based on the free-field assumption or simple point source models cannot accurately reflect the impact of real boundaries on complex directional sound sources. Therefore, this embodiment first obtains sound pressure data characterizing the loudspeaker unit's radiation properties in a free field and then inverts to obtain its unique spherical harmonic function expansion coefficients. Based on these coefficients and different preset distance and impedance values, the transfer function matrix of the loudspeaker unit in various corresponding half-space environments is derived through acoustic model calculations. This avoids the engineering impossibility of measuring each boundary combination on-site and the severe transfer function mismatch problem caused by the inability to model complex directional sound sources and finite impedance boundaries in traditional methods. It achieves accurate prediction of the sound field response of any loudspeaker unit under various real room boundary conditions with only one free-field measurement, providing a universal, accurate, and efficient acoustic model foundation for the offline calculation of the core beamforming filter, fundamentally ensuring the superiority of the final beamforming effect in various application scenarios.

[0050] For example, in an anechoic chamber acting as a free field, the loudspeaker unit under test is fixed at the center of a turntable, and 73 microphones are evenly spaced on a semicircle centered on it as control points. Sound pressure data is obtained by exciting the unit and simultaneously acquiring the complex sound pressure values ​​of all microphones at different frequencies. Then, near-field acoustic holography is used to invert the sound pressure data, solving for a set of spherical harmonic function expansion coefficients that uniquely characterize the complex directivity of the unit. Subsequently, for various preset combinations of distance values ​​(e.g., 0.1m, 0.2m, 0.3m) and impedance values ​​(e.g., complex impedance values ​​corresponding to rigid walls, gypsum boards, and glass), each combination is defined as a specific half-space scene. Based on the obtained spherical harmonic function expansion coefficients, the sound pressure response of the unit to all control points in the corresponding half-space scene is calculated using an impedance boundary half-space transfer function model (which integrates spherical harmonic function superposition, the mirror source method, and boundary reflection coefficients), and then assembled into a transfer function matrix for that scene. This process iterates through all preset parameter combinations, ultimately generating multiple transfer function matrices for the speaker unit that correspond to different half-space environments, laying the foundation for subsequent offline calculation of the preset beamforming filter library.

[0051] In this embodiment, the spherical harmonic function expansion coefficients, which characterize the complex directivity of the speaker unit, are first determined by sound pressure measurement in a free field. Then, based on these coefficients, the transfer function matrix of the unit in a half-space environment defined by various preset distance and preset impedance values ​​is derived through an acoustic model. This avoids the technical problem of traditional methods failing to accurately handle real speaker units with complex directivity and finite impedance boundaries such as TV background walls due to reliance on point source models and rigid boundary assumptions, resulting in severe transfer function mismatch. This achieves the goal of accurately and efficiently constructing an acoustic model library covering various potential real room environments for any speaker unit with only one basic free-field acoustic measurement. This provides a universal and reliable input for the prefabrication of the core beamforming filter, fundamentally ensuring that the final beamforming scheme can achieve excellent performance under diverse real impedance boundary conditions.

[0052] In one feasible implementation, the step of determining the multiple transfer function matrices of the loudspeaker unit in different half-spaces based on the spherical harmonic function expansion coefficients of the loudspeaker unit in step S03 may include steps S031 to S033: Step S031: In a half-space under any combination of preset distance value and preset impedance value, determine the spherical wave function of the half-space based on the preset distance value and the preset impedance value; It should be noted that the half-space spherical wavefunction is a special sound field basis function that satisfies impedance boundary conditions. It is used to accurately describe the propagation behavior of sound waves in a half-space environment with impedance boundaries. This function may not be a single wavefunction, but rather a superposition of three parts: a spherical wavefunction representing the direct sound from the real sound source, a spherical wavefunction representing the geometrically reflected sound from the mirror source, and an additional wavefunction term used to correct for finite impedance boundary effects. The core value of this function lies in its ability to combine the complex directivity characteristics of a loudspeaker unit with arbitrary boundary impedance, source-boundary distance, and other parameters through rigorous mathematical forms, based on the spherical harmonic expansion coefficients measured in a free field, thereby calculating the sound field produced by the unit in a real half-space environment.

[0053] For example, using the numerical Green's function method based on the wave equation, the spatial location of the impedance boundary is first determined according to the preset distance value, and the local response impedance boundary condition on the boundary is directly defined in combination with the preset impedance value. Then, the Helmholtz equation containing the boundary condition is solved by numerical method to calculate the half-space Green's function from the sound source location to the field point location. This Green's function is equivalent to the required half-space spherical wave function.

[0054] Step S032: Determine the sound pressure data of the loudspeaker unit at the corresponding control point in the half-space based on the half-space spherical wave function, the expansion coefficient of the spherical harmonic function, and the preset truncation order of the spherical harmonic expansion term corresponding to the expansion coefficient of the spherical harmonic function. It should be noted that the preset truncation order refers to the highest order of the spherical harmonic expansion terms, pre-set to balance computational accuracy and complexity when using spherical harmonic function expansion to describe the sound field. Theoretically, spherical harmonic function expansion is an infinite series, but in practical numerical calculations, truncation is necessary. The preset truncation order is this cutoff point, which determines the total number of terms retained in the spherical harmonic expansion. The total number of terms can be (truncation order + 1). 2 However, if the order is too low, the sound field model will lack accuracy and will not be able to accurately describe the high frequency or complex directivity of the speaker unit. If the order is too high, it will unnecessarily increase the computational load and will have limited effect on improving accuracy.

[0055] Half-space sound pressure data is a set of complex sound pressure values ​​at various control points, calculated using a half-space spherical wave function model for a specific half-space environment. This data is based on the spherical harmonic function expansion coefficients obtained in a free field and predicted by an acoustic theory model. These data accurately simulate the sound pressure amplitude and phase generated by the sound waves radiated by the loudspeaker unit (including direct sound and sound waves reflected and absorbed by the boundary) at various control points in space under specific impedance boundary conditions. It serves as a direct data source for subsequently constructing the transfer function matrix used in beamforming design.

[0056] For example, based on the half-space spherical wave function spherical harmonic function expansion coefficients , and the expansion coefficients of spherical harmonic functions Preset truncation order for the corresponding spherical harmonic expansion term The sound pressure level data of the speaker unit at the corresponding control point in the half-space is determined by the following formula. :

[0057] in, This represents the total number of items retained during the expansion. To correspond to the free field order coefficient The spherical harmonic expansion coefficients.

[0058] Step S033: Traverse each control point corresponding to the loudspeaker unit, and construct the transfer function matrix of the loudspeaker unit based on the half-space sound pressure data at each control point.

[0059] Understandably, the lack of a unified sound field calculation method may lead to inaccurate reconstruction of the sound field under impedance boundaries or inconsistent calculation methods during implementation. Therefore, this implementation method determines the half-space spherical wave function that satisfies the boundary conditions based on preset distance and preset impedance values. Then, it calculates the half-space sound pressure data at each control point in sequence by combining the known spherical harmonic function expansion coefficients and preset truncation order, and finally constructs the transfer function matrix. This effectively avoids the problem of half-space sound field prediction distortion caused by missing or inappropriate sound field calculation models, especially the problem of inaccurate simulation of reflection effects at finite impedance boundaries. Ultimately, it achieves a precise and controllable conversion from basic acoustic parameters to the target environment acoustic model, providing a reliable and reproducible calculation path for generating a high-fidelity transfer function matrix. This ensures that the influence of complex impedance boundaries can be accurately considered in the design stage of the subsequent beamforming filter.

[0060] For example, an initially empty complex matrix is ​​created, with the number of rows equal to the total number of control points M and the number of columns equal to the total number of speaker units L. Then, a nested loop process is executed, with the outer loop iterating through each control point (index m) and the inner loop iterating through each speaker unit (index l). In each inner loop iteration, the pre-calculated half-space sound pressure data generated by the l-th speaker unit at the m-th control point is used. As a matrix element, it is filled into the m-th row and l-th column position of the matrix; after completing all M×L assignment operations, a complete M×L dimensional transfer function matrix is ​​constructed. This matrix systematically characterizes the acoustic transmission relationship of all loudspeaker units to all control points under specific half-space boundary conditions.

[0061] In this embodiment, the spherical wave function of the half-space is determined based on preset distance and preset impedance values. Then, the sound pressure data of the half-space at each control point is calculated by combining the expansion coefficient of the spherical harmonic function and the preset truncation order. Finally, the transfer function matrix is ​​constructed. This avoids the problem that the complex sound wave reflection and diffraction behavior under finite impedance boundaries cannot be accurately simulated by directly using the free field transfer function or a simple mirror source model. It is possible to couple the inherent directivity characteristics of the loudspeaker unit with arbitrary boundary impedance and distance parameters through a rigorous acoustic model, so as to accurately predict its sound field response in a real half-space environment. This provides an accurate and reliable transfer function data basis for generating high-fidelity beamforming filters.

[0062] In one feasible implementation, the step of determining the half-space spherical wave function based on the preset distance value and the preset impedance value in step S031 may include steps S100~S600: Step S100: Obtain the first spherical coordinates and the second spherical coordinates of the control point corresponding to the speaker unit. The first spherical coordinates are located in a spherical coordinate system with the speaker unit as the center, and the second spherical coordinates are located in a spherical coordinate system with the mirror speaker unit as the center. The mirror speaker unit and the speaker unit are distributed with the impedance boundary in the half-space as the plane of symmetry. The relative positional relationship between the first spherical coordinates and the second spherical coordinates is determined by the preset distance value. It should be noted that the first spherical coordinate system is a spherical coordinate system with the acoustic center of the real loudspeaker unit as the origin. It is used to describe the spatial position of any control point in the sound field relative to the real sound source. This coordinate system uniquely determines the geometric relationship between the control point and the real sound source through three parameters: radial distance, polar angle, and azimuth angle. It is the basis for calculating the direct sound field generated by the real sound source.

[0063] The second spherical coordinate system is a spherical coordinate system with the acoustic center of the mirror loudspeaker unit as the origin. It is used to describe the spatial position of any control point in the sound field relative to the mirror sound source. The definition of this coordinate system is similar to that of the first spherical coordinate system, but its origin is located at the symmetrical point on the other side of the boundary. It is the basis for calculating the reflected sound field generated by the mirror sound source.

[0064] A mirror loudspeaker unit is a virtual sound source that is symmetrical to the impedance boundary relative to a real loudspeaker unit, introduced to satisfy the impedance boundary condition. This loudspeaker unit is not a physical entity, but a mathematically equivalent sound source used to simulate the contribution of sound waves emitted by a real sound source to the original sound field after being reflected by the impedance boundary.

[0065] For example, a schematic diagram of the spatial distribution of the loudspeaker unit and the mirror loudspeaker unit can be referred to Figure 3 , Figure 3 The real sound source in That is, the loudspeaker unit, mirroring the sound source. This refers to a mirrored speaker unit. The actual sound source. With mirror sound source The distances to the impedance boundaries are all preset distance values. Furthermore, for ease of representation, Figure 3 Only the actual sound source is shown. Cartesian coordinate system and mirror sound source Cartesian coordinate system .

[0066] Step S200: Determine the complex angle of the control point in the half-space based on the preset impedance value; The complex angle is a mathematical variable introduced when calculating the second component of the spherical wave function of a mirror sound source. This angle is a complex value determined by the impedance value of the impedance boundary, that is, the preset impedance value. Its function is to enable the calculation of the spherical harmonic function to be extended to the complex space so as to accurately satisfy the finite impedance boundary conditions.

[0067] For example, based on a preset impedance value air characteristic impedance Through the relation: Determine complex angles .

[0068] Step S300: Determine the wall loss factor and plane wave reflection coefficient of the impedance boundary in the half-space based on the preset impedance value and the second spherical coordinates. The wall loss factor is a dimensionless parameter used to correct the second component of the spherical wave function of the mirror sound source. This factor takes into account the partial sound wave transmission loss caused by the finite (non-rigid) boundary acoustic impedance. The factor is related to the preset impedance value, the sound wave incident angle (i.e., the polar angle in the second spherical coordinates), and the sound wave number. Its function is to accurately simulate the energy attenuation that occurs when the sound wave is reflected on the finite impedance boundary.

[0069] For example, based on a preset impedance value Angle of incidence of sound waves Harmony wave number Through formula Calculate the wall loss factor ,in, , Radial distance, Let be the characteristic impedance of the air, and erfc be the residual error function. This parameter describes the energy propagation of the wall wave.

[0070] The plane wave reflection coefficient is a complex number representing the ratio of the reflected sound pressure to the incident sound pressure when a plane sound wave is incident on an impedance boundary. This coefficient is a function of the impedance value and the incident angle of the sound wave, describing the amplitude and phase change of the sound wave reflection caused by the boundary. It is a key parameter that distinguishes between rigid boundaries and finite impedance boundaries, and directly affects the calculation of the second component of the spherical wave function of the mirror sound source.

[0071] For example, a preset impedance value is used. air characteristic impedance Substituting into the plane wave reflection coefficient formula: The plane wave reflection coefficient in complex form can be directly calculated. .

[0072] Step S300: Determine the complex angle based on the preset impedance value; The complex angle is a mathematical variable introduced when calculating the second component of the spherical wave function of a mirror sound source. This angle is a complex value determined by the boundary impedance value. Its function is to extend the calculation of the spherical harmonic function to complex space so as to accurately satisfy the finite impedance boundary conditions.

[0073] For example, based on a preset impedance value air characteristic impedance Through the relation: Determine complex angles .

[0074] Step S400: Determine the spherical wave function of the real sound source based on the first spherical coordinates; It should be noted that the real sound source spherical wave function is a mathematical function that describes the sound wave emitted from the real loudspeaker unit and propagating directly to the control point. This function represents the direct sound component in the sound field, and its form is usually the product of a spherical harmonic function and a second-kind Hankel function, which is only related to the first spherical coordinates.

[0075] In the specific implementation process, step S400 may include steps S401 to S403: Step S401: Determine the first and second type of Hankel functions based on the radial distance in the first spherical coordinates; Step S402: Determine the first spherical harmonic function based on the polar angle and azimuth angle in the first spherical coordinates; Step S403: Determine the true sound source spherical wave function based on the first and second type Hankel functions and the first spherical harmonic function.

[0076] For example, further reference can be made. Figure 4 According to the first sphere coordinates radial distance The first and second types of Hankel functions can be determined. ,in, , Angular frequency, The speed of sound. Based on the first spherical coordinates. polar angle and azimuth The first spherical harmonic function can be determined. Thus, the true spherical wave function of the sound source. It can be represented as: ,in, This represents the distance from the center of the sound source to the impedance boundary. for A unit vector in direction.

[0077] Step S500: Determine the first component of the spherical wave function of the mirror sound source based on the second spherical coordinates, and determine the second component of the spherical wave function of the mirror sound source based on the second spherical coordinates, the complex angle, the wall loss factor, and the plane wave reflection coefficient; It should be noted that the first component of the spherical wave function of the mirror sound source is a part of the spherical wave function of the mirror sound source. Its mathematical form is exactly the same as that of the spherical wave function of the real sound source, but the coordinates used in the calculation are second spherical coordinates. This component simulates the reflected sound field generated by the mirror sound source under ideal rigid boundary conditions.

[0078] The second component of the spherical wave function of the mirror sound source is a unique correction term in the spherical wave function of the mirror sound source. This component is determined by parameters such as the plane wave reflection coefficient, the wall loss factor, and the complex angle. This component is specifically used to correct the deviation between the first component and the real reflected sound field under finite impedance boundary conditions, thereby accurately simulating the absorption and phase effect of non-rigid boundaries on reflected sound waves.

[0079] In the specific implementation process, step S500 may include steps S501 to S503: Step S501: Determine the second type of Hankel function based on the radial distance in the second spherical coordinates; Step S502: Determine the second spherical harmonic function based on the polar angle and azimuth angle in the second spherical coordinates, and determine the third spherical harmonic function based on the complex angle and the azimuth angle in the second spherical coordinates; Step S503: Determine the first component of the spherical wave function of the mirror sound source based on the second type of Hankel function and the second spherical harmonic function; Step S504: Based on the plane wave reflection coefficient, the second type Hankel function, the second spherical harmonic function, the third spherical harmonic function, and the wall loss factor, determine the second component of the spherical wave function of the mirror sound source.

[0080] For example, further reference can be made. Figure 4 According to the second spherical coordinates radial distance The second type of Hankel function can be determined. According to the second spherical coordinates polar angle and azimuth The second spherical harmonic function can be determined. According to the complex number perspective Second spherical coordinates Azimuth in The third spherical harmonic function can be determined. Therefore, the first component of the spherical wave function of the mirror sound source. It can be represented as: The second component of the spherical wave function of a mirror sound source. It can be represented as: .

[0081] Step S600: Determine the half-space spherical wave function based on the real sound source spherical wave function, the first component, and the second component.

[0082] Understandably, since using the half-space spherical wave function for calculation may lead to inaccurate modeling of complex reflected sound fields under finite impedance boundaries during implementation, this embodiment establishes spherical coordinate systems with the real sound source and the mirror sound source as the origins, derives the plane wave reflection coefficient based on preset impedance values, and determines the wall loss factor based on preset distance values. This allows for the precise decomposition of the half-space spherical wave function into three parts: the real sound source term, the mirror sound source rigid reflection term, and the impedance boundary correction term, which are then superimposed for calculation. This avoids sound field prediction errors caused by using approximate reflection models or ignoring the frequency characteristics of boundary impedance, particularly the inability to accurately simulate the absorption and phase distortion of mid-to-high frequency sound waves by finite impedance boundaries such as TV background walls. This enables physically accurate modeling of the half-space sound field under arbitrary impedance boundaries. The rigorous analytical method ensures the calculation accuracy of the transfer function matrix over a wide bandwidth, laying a reliable acoustic theoretical foundation for generating high-performance beamforming filters.

[0083] For example, based on the spherical wave function of a real sound source The first component of the spherical wave function of the mirror sound source and the second component of the spherical wave function of the mirror sound source The wave function of the half-space spherical surface can be determined. for: .

[0084] In this embodiment, by accurately decomposing the half-space spherical wavefunction into the first component (corresponding to rigid reflection) and the second component (corresponding to impedance correction) of the real sound source spherical wavefunction and the mirror sound source spherical wavefunction, and superimposing them, the problem of not being able to accurately simulate complex reflected sound fields under finite impedance boundaries due to the use of a single point source model or neglecting the frequency characteristics of boundary impedance is avoided. This achieves physically accurate modeling of arbitrary impedance boundary conditions. The comprehensive contribution of direct sound, geometrically reflected sound, and boundary impedance effects is fully characterized by analytical methods, thus providing a rigorous theoretical basis for the calculation of the transfer function matrix and ensuring the accuracy and reliability of beamforming design in a real room environment.

[0085] In one feasible implementation, step S04 may further include steps S041 to S042: Step S041: For any one of the transfer function matrices, determine the beamforming filter corresponding to the transfer function matrix based on the transfer function matrix, the conjugate transpose of the transfer function matrix, the regularization control matrix corresponding to the transfer function matrix, and the target sound pressure corresponding to the transfer function matrix. It should be noted that the regularization control matrix is ​​a diagonal matrix introduced when solving beamforming filters to improve the ill-conditioned nature of the problem and enhance the stability of the solution. Its typical form is as follows: ,in, For regularization parameters, For the transfer function matrix The largest eigenvalue, This is a custom value used to control the degree of regularization. This is the identity matrix. By adding a penalty term to the objective function of the optimization problem, this matrix limits the energy of the filter coefficients, thereby avoiding drastic fluctuations in the solution or numerical non-convergence caused by ill-conditioned transfer function matrix. Its role is to ensure that the solved beamforming filter has good robustness and feasibility while achieving high directional beams.

[0086] The target sound pressure level is a predefined complex vector describing the desired ideal sound pressure distribution at each control point. In this embodiment, the target sound pressure level can be set to 1 (representing unit excitation) at the control point in the bright area (the area the main beam points to) and 0 at the control point in the dark area (the area where sound radiation needs to be suppressed). Its function is to serve as an optimization target in the sound pressure matching method, guiding the design direction of the beamforming filter, so that the final actual sound field can approximate this ideal "clear bright areas and silent dark areas" sound pressure distribution to the greatest extent possible, thereby achieving a high-contrast beamforming effect. The schematic diagram of the bright and dark area scenes in this embodiment can be found in [reference needed]. Figure 5 Consider a loudspeaker array consisting of M loudspeaker units. Select I points (centered on the center point of the loudspeaker line array) located on equally spaced semicircles as control points. The corresponding radiation direction angles are... It means that, among them The region where the beam points is called the bright area, corresponding to... Figure 5 The solid dot is in the center, and the other directions are called dark areas, corresponding to... Figure 5 Hollow center point.

[0087] For example, for any transfer function matrix According to the transfer function matrix The conjugate transpose of the transfer function matrix The regularization control matrix corresponding to the transfer function matrix. and the target sound pressure corresponding to this transfer function matrix The beamforming filter corresponding to the transfer function matrix can be determined. for .

[0088] Step S042: Traverse each of the transfer function matrices to obtain each of the preset beamforming filters.

[0089] Understandably, obtaining only an accurate transfer function matrix is ​​insufficient to guarantee that the final beamforming filter will possess the numerical stability and acoustic performance required for practical engineering applications. This is because the transfer function matrix is ​​often ill-conditioned, and direct inversion can lead to unstable solutions and excessive gain. Therefore, this implementation introduces a regularization control matrix composed of regularization parameters and an identity matrix into the solution formula for the beamforming filter. It also uses a sound pressure matching method with the target sound pressure vector (1 in the bright area and 0 in the dark area) as the optimization target. This avoids the problems of filter coefficient explosion, poor system robustness, and high sound pressure grating lobes in the dark area caused by ill-conditioned matrices. It achieves a stable solution for a beamforming filter with finite gain and good realizability based on an accurate acoustic model, thus physically achieving a beamforming effect with high sound contrast and high directivity.

[0090] Exemplarily, to illustrate the technical effects of the loudspeaker array beamforming method obtained by combining this embodiment with the first embodiment described above, please refer to... Figure 6 and Figure 7 , Figure 6 and Figure 7 The beam patterns obtained using existing methods and by combining this embodiment with the technical solution described in Embodiment 1 are shown respectively. It can be seen that... Figure 6 Existing methods have resulted in beams that have deviated significantly from the intended beam direction and angle range. Furthermore, numerous grating lobes have appeared in the dark areas, leading to a severe degradation in beam performance. Figure 7 The beam obtained in this embodiment, combined with the technical solution of Embodiment 1 above, has a significant reduction in grating lobes in the dark region.

[0091] Furthermore, the acoustic contrast ratio can be used to measure beam performance. Acoustic contrast ratio is defined as the ratio of the average energy in the bright area to the average energy in the dark area, that is, the ratio of the average energy of the main lobe of the beam to the average energy in the other directions, expressed in decibels (dB). A higher acoustic contrast ratio indicates better beam performance from the loudspeaker array. Figure 8 The acoustic contrast results of existing methods and the present application are shown. The dashed line corresponds to the existing method, and the solid line corresponds to the present application. It can be clearly observed that the acoustic contrast of the present application is greatly improved compared with the existing method.

[0092] It should be noted that the above examples are only for understanding this application and do not constitute a limitation on the speaker array beamforming method of this application. Any simple modifications based on this technical concept are within the protection scope of this application.

[0093] This application provides an electronic device, which includes: at least one processor; and a memory communicatively connected to the at least one processor; wherein the memory stores instructions executable by the at least one processor, the instructions being executed by the at least one processor to enable the at least one processor to perform the loudspeaker array beamforming method of the above embodiment 1.

[0094] The following is for reference. Figure 9 The diagram illustrates a structural schematic of an electronic device suitable for implementing embodiments of this application. The electronic devices in these embodiments may include, but are not limited to, mobile terminals such as mobile phones, laptops, digital broadcast receivers, PDAs (Personal Digital Assistants), PADs (Portable Application Descriptions), PMPs (Portable Media Players), in-vehicle terminals (e.g., in-vehicle navigation terminals), and fixed terminals such as digital TVs and desktop computers. Figure 9 The electronic device shown is merely an example and should not impose any limitation on the functionality and scope of use of the embodiments of this application.

[0095] like Figure 9As shown, the electronic device may include a processing unit 1001 (e.g., a central processing unit, a graphics processing unit, etc.), which can perform various appropriate actions and processes according to a program stored in a read-only memory 1002 or a program loaded from a storage device 1003 into a random access memory 1004. The random access memory 1004 also stores various programs and data required for the operation of the electronic device. The processing unit 1001, the read-only memory 1002, and the random access memory 1004 are interconnected via a bus 1005. An input / output interface 1006 is also connected to the bus. Typically, the following systems can be connected to the input / output interface 1006: input devices 1007 including, for example, touchscreens, touchpads, keyboards, mice, image sensors, microphones, accelerometers, gyroscopes, etc.; output devices 1008 including, for example, liquid crystal displays (LCDs), speakers, vibrators, etc.; storage devices 1003 including, for example, magnetic tapes, hard disks, etc.; and communication devices 1009. The communication device 1009 allows the electronic device to communicate wirelessly or wiredly with other devices to exchange data. Although the diagrams show electronic devices with various systems, it should be understood that it is not required to implement or have all of the systems shown. More or fewer systems may be implemented alternatively.

[0096] Specifically, according to the embodiments disclosed in this application, the processes described above with reference to the flowcharts can be implemented as computer software programs. For example, embodiments disclosed in this application include a computer program product comprising a computer program carried on a computer-readable medium, the computer program containing program code for performing the methods shown in the flowcharts. In such embodiments, the computer program can be downloaded and installed from a network via a communication device, or installed from storage device 1003, or installed from read-only memory 1002. When the computer program is executed by processing device 1001, it performs the functions defined in the methods of the embodiments disclosed in this application.

[0097] The electronic device provided in this application, employing the speaker array beamforming method in the above embodiments, can solve the technical problem that current speaker beamforming schemes in real-world environments cannot simultaneously achieve both versatility and accuracy. Compared with the prior art, the beneficial effects of the electronic device provided in this application are the same as those of the speaker array beamforming method provided in the above embodiments, and other technical features of this electronic device are the same as those disclosed in the previous embodiment method, and will not be repeated here.

[0098] It should be understood that the various parts disclosed in this application can be implemented using hardware, software, firmware, or a combination thereof. In the description of the above embodiments, specific features, structures, materials, or characteristics can be combined in any suitable manner in one or more embodiments or examples.

[0099] The above description is merely a specific embodiment of this application, but the scope of protection of this application is not limited thereto. Any variations or substitutions that can be easily conceived by those skilled in the art within the scope of the technology disclosed in this application should be included within the scope of protection of this application. Therefore, the scope of protection of this application should be determined by the scope of the claims.

[0100] This application provides a computer-readable storage medium having computer-readable program instructions (i.e., a computer program) stored thereon, the computer-readable program instructions being used to execute the loudspeaker array beamforming method in the above embodiments.

[0101] The computer-readable storage medium provided in this application may be, for example, a USB flash drive, but is not limited to, electrical, magnetic, optical, electromagnetic, infrared, or semiconductor systems or devices, or any combination thereof. More specific examples of computer-readable storage media may include, but are not limited to: electrical connections having one or more wires, portable computer disks, hard disks, random access memory (RAM), read-only memory (ROM), erasable programmable read-only memory (EPROM or flash memory), optical fiber, portable compact disk read-only memory (CD-ROM), optical storage devices, magnetic storage devices, or any suitable combination thereof. In this embodiment, the computer-readable storage medium may be any tangible medium containing or storing a program that can be used by or in conjunction with an instruction execution system or device. The program code contained on the computer-readable storage medium may be transmitted using any suitable medium, including but not limited to: wires, optical cables, RF (Radio Frequency), etc., or any suitable combination thereof.

[0102] The aforementioned computer-readable storage medium may be included in an electronic device or may exist independently without being assembled into an electronic device.

[0103] The aforementioned computer-readable storage medium carries one or more programs that, when executed by an electronic device, cause the electronic device to: acquire the distance values ​​between each speaker unit in the speaker array and its corresponding impedance boundary, and the impedance value of the impedance boundary; match the target beamforming filter corresponding to each speaker unit among a plurality of preset beamforming filters based on the distance values ​​and the impedance values, wherein each preset beamforming filter is determined under different impedance boundary conditions; and filter the input signal of the corresponding speaker unit based on each target beamforming filter to form the output beam of the speaker array.

[0104] Computer program code for performing the operations of this application can be written in one or more programming languages ​​or a combination thereof, including object-oriented programming languages ​​such as Java, Smalltalk, and C++, and conventional procedural programming languages ​​such as the "C" language or similar programming languages. The program code can be executed entirely on the user's computer, partially on the user's computer, as a standalone software package, partially on the user's computer and partially on a remote computer, or entirely on a remote computer or server. In cases involving remote computers, the remote computer can be connected to the user's computer via any type of network—including a Local Area Network (LAN) or a Wide Area Network (WAN)—or can be connected to an external computer (e.g., via the Internet using an Internet service provider).

[0105] The flowcharts and block diagrams in the accompanying drawings illustrate the architecture, functionality, and operation of possible implementations of systems, methods, and computer program products according to various embodiments of this application. In this regard, each block in a flowchart or block diagram may represent a module, segment, or portion of code containing one or more executable instructions for implementing a specified logical function. It should also be noted that in some alternative implementations, the functions indicated in the blocks may occur in a different order than those indicated in the drawings. For example, two consecutively indicated blocks may actually be executed substantially in parallel, and they may sometimes be executed in reverse order, depending on the functions involved. It should also be noted that each block in the block diagrams and / or flowcharts, and combinations of blocks in the block diagrams and / or flowcharts, can be implemented using a dedicated hardware-based system that performs the specified function or operation, or using a combination of dedicated hardware and computer instructions.

[0106] The modules described in the embodiments of this application can be implemented in software or hardware. The names of the modules do not necessarily limit the functionality of the unit itself.

[0107] The readable storage medium provided in this application is a computer-readable storage medium that stores computer-readable program instructions (i.e., a computer program) for executing the above-described loudspeaker array beamforming method. This solves the technical problem that current loudspeaker beamforming schemes in real-world environments cannot simultaneously achieve both versatility and accuracy. Compared with the prior art, the beneficial effects of the computer-readable storage medium provided in this application are the same as those of the loudspeaker array beamforming method provided in the above embodiments, and will not be repeated here.

[0108] This application also provides a computer program product, including a computer program that, when executed by a processor, implements the steps of the speaker array beamforming method described above.

[0109] The computer program product provided in this application can solve the technical problem that current loudspeaker beamforming schemes in real-world environments cannot simultaneously achieve both versatility and accuracy. Compared with the prior art, the beneficial effects of the computer program product provided in this application are the same as those of the loudspeaker array beamforming method provided in the above embodiments, and will not be repeated here.

[0110] The above description is only a part of the embodiments of this application and does not limit the patent scope of this application. All equivalent structural transformations made under the technical concept of this application and using the contents of the specification and drawings of this application, or direct / indirect applications in other related technical fields, are included in the patent protection scope of this application.

Claims

1. A loudspeaker array beamforming method, characterized by, The loudspeaker array beamforming method comprises: obtaining distance values between each loudspeaker unit in the loudspeaker array and a corresponding impedance boundary, and impedance values of the impedance boundary; matching a target beamforming filter corresponding to each loudspeaker unit from a plurality of preset beamforming filters according to the distance values and the impedance values, wherein each preset beamforming filter is determined under different impedance boundary conditions; filtering input signals of the corresponding loudspeaker unit based on each target beamforming filter to form an output beam of the loudspeaker array. 2.The loudspeaker array beamforming method of claim 1, wherein, Each preset beamforming filter is associated with a corresponding preset distance value and a preset impedance value, and the step of matching a target beamforming filter corresponding to each loudspeaker unit from a plurality of preset beamforming filters according to the distance values and the impedance values comprises: matching a target preset distance value and a target preset impedance value from a plurality of associated preset distance values and preset impedance values according to the distance values and the impedance values; taking the preset beamforming filter corresponding to the target preset distance value and the target preset impedance value as the target beamforming filter. 3.The loudspeaker array beamforming method of claim 2, wherein, Before the step of matching a target beamforming filter corresponding to each loudspeaker unit from a plurality of preset beamforming filters according to the distance values and the impedance values, the method further comprises: obtaining sound pressure data of each loudspeaker unit at a corresponding control point in a free field, wherein the free field has no impedance boundary; determining spherical harmonic function expansion coefficients of the corresponding loudspeaker unit according to each sound pressure data; for any one loudspeaker unit, determining a plurality of transfer function matrices of the loudspeaker unit in different half-spaces according to the spherical harmonic function expansion coefficients of the loudspeaker unit, wherein any one half-space corresponds to a combination of a preset distance value and a preset impedance value; converting each transfer function matrix into a corresponding beamforming filter to obtain each preset beamforming filter. 4.The loudspeaker array beamforming method of claim 3, wherein, The step of determining a plurality of transfer function matrices of the loudspeaker unit in different half-spaces according to the spherical harmonic function expansion coefficients of the loudspeaker unit comprises: determining a half-space spherical wave function under a half-space of any one combination of a preset distance value and a preset impedance value based on the preset distance value and the preset impedance value; determining half-space sound pressure data of the loudspeaker unit at a corresponding control point in the half-space according to the half-space spherical wave function, the spherical harmonic function expansion coefficients, and a preset truncation order of a spherical harmonic expansion term corresponding to the spherical harmonic function expansion coefficients; iterating through each control point corresponding to the loudspeaker unit, and constructing a transfer function matrix of the loudspeaker unit based on half-space sound pressure data at each control point. 5.The loudspeaker array beamforming method of claim 4, wherein, The step of determining a half-space spherical wave function based on the preset distance value and the preset impedance value comprises: obtaining a first spherical coordinate and a second spherical coordinate of a control point corresponding to the loudspeaker unit, wherein the first spherical coordinate is located in a spherical coordinate system with the loudspeaker unit as a spherical center, the second spherical coordinate is located in a spherical coordinate system with a mirror loudspeaker unit as a spherical center, the mirror loudspeaker unit is distributed with the impedance boundary in the half space as a plane of symmetry, and a relative position relationship of the first spherical coordinate and the second spherical coordinate is determined by the preset distance value; determining a complex angle of the control point in the half space according to the preset impedance value; determining a wall loss factor and a plane wave reflection coefficient of the impedance boundary in the half space according to the preset impedance value and the second spherical coordinate; determining a real sound source spherical wave function based on the first spherical coordinate; determining a first component of a mirror sound source spherical wave function based on the second spherical coordinate, and determining a second component of the mirror sound source spherical wave function based on the second spherical coordinate, the complex angle, the wall loss factor and the plane wave reflection coefficient; determining a half-space spherical wave function according to the real sound source spherical wave function, the first component and the second component. 6.The loudspeaker array beamforming method of claim 5, wherein, The step of determining the real sound source spherical wave function based on the first spherical coordinate comprises: determining a first second-type Hankel function according to a radial distance in the first spherical coordinate; determining a first spherical harmonic function according to a polar angle and an azimuth angle in the first spherical coordinate; determining the real sound source spherical wave function based on the first second-type Hankel function and the first spherical harmonic function. 7.The loudspeaker array beamforming method of claim 5, wherein, The step of determining the first component of the mirror sound source spherical wave function based on the second spherical coordinate, and determining the second component of the mirror sound source spherical wave function based on the second spherical coordinate, the complex angle, the wall loss factor and the plane wave reflection coefficient comprises: determining a second second-type Hankel function according to a radial distance in the second spherical coordinate; determining a second spherical harmonic function according to a polar angle and an azimuth angle in the second spherical coordinate, and determining a third spherical harmonic function according to the complex angle and the azimuth angle in the second spherical coordinate; determining the first component of the mirror sound source spherical wave function based on the second second-type Hankel function and the second spherical harmonic function; determining the second component of the mirror sound source spherical wave function based on the plane wave reflection coefficient, the second second-type Hankel function, the second spherical harmonic function, the third spherical harmonic function and the wall loss factor. 8.The loudspeaker array beamforming method of claim 3, wherein, The step of converting each transfer function matrix into a corresponding beamforming filter to obtain each preset beamforming filter comprises: for any one of the transfer function matrices, determining a beamforming filter corresponding to the transfer function matrix according to the transfer function matrix, a conjugate transpose matrix of the transfer function matrix, a regularization control matrix corresponding to the transfer function matrix and a target sound pressure corresponding to the transfer function matrix; iterating through each of the transfer function matrices to obtain each of the preset beamforming filters.

9. An electronic device, comprising: The device comprises a memory, a processor, and a computer program stored on the memory and executable on the processor, and the computer program is configured to implement the steps of the loudspeaker array beamforming method according to any one of claims 1 to 8.

10. A storage medium, characterized by The storage medium is a computer readable storage medium, and the storage medium stores a computer program, and the computer program is executed by a processor to implement the steps of the loudspeaker array beamforming method according to any one of claims 1 to 8.

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