A loudspeaker array design method, structure and device based on sound path difference compensation

Abstract

This invention discloses a speaker array design method and structure based on path difference compensation, belonging to the field of acoustic structure design technology. The method includes: establishing a parametric three-dimensional model of the speaker array, dividing the speakers into a right channel group, a reverse right channel group, a left channel group, and a reverse left channel group; setting listening area parameters and setting multiple sampling positions within the range; setting left and right ear reference points at each sampling position; calculating the path difference at each sampling position through acoustic simulation; and using a numerical optimization algorithm to obtain the optimal speaker layout parameters by minimizing the maximum path difference across all sampling positions. This invention, through a four-channel structure and optimization design method, achieves stereo separation over a wide listening distance, solving the problem of lack of stereo sound in narrow-box electronic devices. It eliminates the need for DSP chips and complex algorithms, offering advantages such as low cost, natural sound, and no "sweet spot" effect.

Description

Technical Field

[0001] This invention relates to the fields of acoustic structure design and intelligent manufacturing technology, and more specifically, to a loudspeaker array design method, structure, and device based on acoustic path difference compensation. Background Technology

[0002] As consumer electronics become increasingly miniaturized and intelligent, the acoustic performance design of audio devices faces increasingly severe challenges. In compact electronic devices (such as smartphones, tablets, and portable speakers), due to physical size limitations, traditional acoustic structures struggle to achieve ideal stereo effects and low-frequency response, resulting in limited sound quality and a poor user experience.

[0003] Currently, acoustic structure design mainly relies on empirical formulas and simple geometric layouts, improving acoustic performance by adjusting speaker positions and adding sound insulation materials. However, these methods often have the following problems: First, they lack accurate modeling of sound wave propagation paths, making it difficult to predict the sound field distribution at different listening distances; second, existing design methods struggle to achieve wide-band sound wave interference control within limited spaces, resulting in insufficient low-frequency response and poor sound field uniformity; third, traditional design methods often require extensive physical prototype testing and repeated adjustments, leading to long development cycles and high costs.

[0004] In the field of architectural acoustics, although various sound insulation and absorption technologies have been applied to spatial acoustic design, these technologies are mainly designed for large-space environments and are difficult to directly apply to the acoustic systems of small electronic devices. For example, while the sound insulation quality law indicates the influence of material surface density on sound insulation performance, it does not provide methods for optimizing acoustic structures for small devices; composite acoustic structures such as honeycomb sandwich structures are widely used in aerospace and vehicle panels, but their design principles and parameter optimization methods are not applicable to the acoustic systems of electronic devices.

[0005] In the consumer electronics sector, with the increasing prevalence of AI-powered smart terminals and foldable screen devices, the demands on acoustic performance are constantly rising. However, existing acoustic structure design methods struggle to meet the high requirements of these emerging products for sound quality and spatial awareness. This is especially true in compact devices where the space behind the speaker is limited, making it difficult for traditional design methods to achieve an ideal low-frequency response, resulting in sound quality loss.

[0006] Furthermore, existing acoustic design methods rely heavily on empirical parameters and manual adjustments, lacking a systematic parametric optimization model, resulting in low design efficiency and difficulty in adapting to the rapidly iterative development needs of electronic products. Summary of the Invention

[0007] The purpose of this invention is to provide a speaker array design method and speaker array structure based on sound path difference compensation, which solves the technical problem of lack of stereo sound in existing narrow-box electronic devices, so that listeners can clearly perceive the sound source from the external space on the left and right sides of the device within a wide listening distance range. At the same time, it provides a quantifiable and reproducible parametric design method based on computer-aided design and acoustic simulation to meet the needs of intelligent manufacturing for precision assembly and rapid iteration.

[0008] To achieve the above objectives, the present invention adopts the following technical solution: The first aspect of this application provides a loudspeaker array design method based on acoustic path difference compensation, comprising the following steps: A parametric 3D model of a loudspeaker array is established. The loudspeaker array includes a cabinet and multiple loudspeakers mounted on the front panel of the cabinet. The loudspeakers are divided into four channel groups: a right channel group, a reverse right channel group, a left channel group, and a reverse left channel group. The left channel group is used to play left channel audio signals; the right channel group is used to play right channel audio signals; the reverse left channel group is used to play audio signals that are in opposite phase to those of the left channel group; and the reverse right channel group is used to play audio signals that are in opposite phase to those of the right channel group. Set the listening area parameters, and set multiple sampling positions within the range defined by the listening area parameters; At each sampling location, a left ear reference point and a right ear reference point are set, and the left ear reference points and right ear reference points are symmetrically distributed with respect to the sampling location; Based on the parametric 3D model, the acoustic path difference at each sampling position is calculated through acoustic simulation. The acoustic path difference is the maximum value of the first acoustic path difference and the second acoustic path difference at that sampling point. Specifically, the first acoustic path difference is the absolute value of the difference between the acoustic path from the left channel group to the right ear reference point and the acoustic path from the reverse left channel group to the right ear reference point; the second acoustic path difference is the absolute value of the difference between the acoustic path from the right channel group to the left ear reference point and the acoustic path from the reverse right channel group to the left ear reference point. A numerical optimization algorithm is used to obtain the optimal loudspeaker layout parameters with the goal of minimizing the maximum sound path difference at all sampling locations; The optimal speaker layout parameters are output to guide the manufacturing of the speaker array.

[0009] Furthermore, each of the left channel group, right channel group, reverse left channel group, and reverse right channel group includes at least one loudspeaker.

[0010] Furthermore, the speaker array adopts a rectangular layout, with the speakers arranged in two rows and two columns on the front panel of the cabinet. The optimal speaker layout parameters include a depth offset, which is the distance difference between adjacent rows of speakers in the rectangular layout in the direction perpendicular to the front panel of the cabinet. The optimal value of the depth offset is determined by the numerical optimization algorithm, so that the sound path difference of all sampling positions is minimized within the range defined by the listening area parameters. The value of the depth offset ranges from 0.5cm to 3.0cm.

[0011] Furthermore, when determining the optimal value of the depth offset through the numerical optimization algorithm, the optimization process of the depth offset satisfies the following constraint: the absolute value of the sound path difference at all sampling positions is less than a preset threshold, which is set according to the phase resolution capability of the human ear.

[0012] Furthermore, the speaker array adopts a linear layout, with the speakers arranged in a straight line along the horizontal direction and mounted on the front panel of the enclosure; the optimal speaker layout parameters include the horizontal spacing between each channel group, which is determined by the numerical optimization algorithm to minimize the sound path difference of all sampling positions within the range of the listening area parameter definition.

[0013] Furthermore, the numerical optimization algorithm adopts a genetic algorithm or a particle swarm optimization algorithm, with a population size of 50–100 and an iteration count of 100–300.

[0014] Furthermore, the acoustic simulation calculation adopts the boundary element method, the mesh size is no larger than 1 / 10 of the speaker diaphragm diameter, and the simulation frequency range covers 200Hz to 10kHz.

[0015] A second aspect of this application provides a loudspeaker array structure based on acoustic path difference compensation, comprising: Box; Multiple loudspeakers are mounted on the enclosure, and the multiple loudspeakers are divided into four channel groups: left channel group, right channel group, reverse left channel group, and reverse right channel group. In the power amplifier circuit, the right channel group and the reverse right channel group are connected in series in opposite phase and then connected to the right channel power amplifier, and the left channel group and the reverse left channel group are connected in series in opposite phase and then connected to the left channel power amplifier.

[0016] Furthermore, it also includes an independent subwoofer, which is connected to the power amplifier circuit, and the crossover point of the independent subwoofer is set between 200Hz and 300Hz.

[0017] A third aspect of this application provides an electronic device including the speaker array structure described above, wherein the electronic device is any one of a speaker, a television, a smart display device, a soundbar, a portable tablet computer, or a car audio system.

[0018] Compared with the prior art, the loudspeaker array design method and structure based on acoustic path difference compensation provided by the present invention have the following beneficial effects: The beneficial effects of this invention are as follows: This invention achieves stereo separation over a wide listening distance through the spatial layout and phase control of a four-channel group structure (left channel group, right channel group, reverse left channel group, and reverse right channel group). The reverse channel group plays audio signals that are out of phase with the corresponding main channel group, achieving sound wave cancellation at the listener's opposite ear. This ensures that the left ear primarily hears the left channel and the right ear primarily hears the right channel, forming clear stereo localization. Experimental data shows that the stereo separation of this invention reaches 15.8dB, the subjective stereo angle can reach 120°-150°, and the sound field width far exceeds the physical enclosure boundaries of traditional equipment.

[0019] For a rectangular layout (two rows and two columns), this invention optimizes the depth offset to create a distance difference between adjacent rows of speakers in the direction perpendicular to the front panel of the enclosure. This depth offset compensates for dynamic changes in the sound path difference over a wide listening distance, ensuring that the sound path from the reverse channel group to the target ear is similar to that of the corresponding main channel group, thus achieving precise sound wave cancellation. Experimental data shows that when the depth offset is between 0.5cm and 3.0cm, the maximum sound path difference can be controlled within 2.0cm to 3.0cm within a listening distance range of 0.5m to 3.0m.

[0020] For a linear layout (single-row, straight-line arrangement), this invention achieves sound path compensation by optimizing the horizontal spacing between each channel group. This layout requires no depth space and is particularly suitable for ultra-thin devices (such as televisions and smart displays). The optimal combination of horizontal spacing is determined through a numerical optimization algorithm, minimizing the sound path difference over a wide listening distance range.

[0021] This invention optimizes the depth offset by constraining the absolute value of the sound path difference at all sampling locations to be less than a preset threshold (2.0cm to 3.0cm), ensuring that the optimization results meet practical application requirements. This threshold range corresponds to a time difference of approximately 0.06ms to 0.09ms for sound waves to travel through the air. Within this range, the human ear can hardly distinguish the phase deviation, thus ensuring a good sound wave cancellation effect.

[0022] This invention specifies the parameters of the numerical optimization algorithm: a genetic algorithm or a particle swarm optimization algorithm, a population size of 50–100, and 100–300 iterations. These parameters have been experimentally verified to achieve a balance between computational efficiency and optimization quality, avoid getting trapped in local optima, and ensure the convergence of the algorithm.

[0023] This invention employs the boundary element method for acoustic simulation, with a mesh size no larger than 1 / 10 of the speaker diaphragm diameter, and a simulation frequency range covering 200Hz to 10kHz. These parameter settings meet the accuracy requirements of acoustic simulation, enabling accurate calculation of the sound field distribution of the speaker array within the listening area, and providing reliable input data for the optimization algorithm.

[0024] This invention achieves stereo separation using a purely physical structure, eliminating the need for DSP chips and complex signal processing algorithms. Through an anti-phase series circuit connection, the reverse channel group automatically plays audio signals that are inversely phase to the main channel group, resulting in a simple and reliable implementation. Compared to virtual stereo algorithm solutions, this invention reduces hardware costs by 30-50% and avoids the sound quality degradation caused by algorithm processing, resulting in a more natural listening experience.

[0025] This invention can be optionally equipped with an independent subwoofer, with the crossover point set between 200Hz and 300Hz to supplement the low-frequency response. This crossover point range is well-connected to the lower limit of the operating frequency of the full-range speaker, enabling uniform coverage across the entire frequency band.

[0026] This invention can be applied to a variety of electronic devices such as speakers, televisions, smart display devices, soundbars, portable tablets, and car audio systems, demonstrating good industry versatility. Two layout options (rectangular and linear) can be flexibly selected according to the device's form factor, achieving full coverage from thick enclosures to ultra-thin devices.

[0027] The parametric design method based on acoustic simulation provided by this invention transforms traditional experience-based design into a quantifiable and reproducible standardized process. By establishing a parametric 3D model, setting listening region parameters, performing acoustic simulation calculations, and employing numerical optimization algorithms to solve for the optimal parameters, the design process is automated. This method facilitates rapid iteration during product development and also enables precision assembly and quality control on automated production lines. Attached Figure Description

[0028] Figure 1 A flowchart illustrating a loudspeaker array optimization design method based on acoustic path difference compensation provided in this application; Figure 2 This is a front view of a rectangular layout speaker array provided in an embodiment of the present invention; Figure 3This is a side view of a rectangular layout speaker array provided in an embodiment of the present invention; Figure 4 A front view schematic diagram of the single-row linear layout speaker array structure provided in this application; Figure 5 This is a schematic diagram of the sound path compensation geometry principle provided in an embodiment of the present invention; Figure 6 This is a circuit connection schematic diagram provided in an embodiment of the present invention. Detailed Implementation

[0029] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the present application will be briefly introduced below in conjunction with the accompanying drawings and descriptions of the embodiments or the prior art. Obviously, the following description of the structure of the accompanying drawings is only some embodiments of this application. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort. It should be noted that the description of these embodiments is used to help understand this application, but does not constitute a limitation on this application.

[0030] The core of this application lies in accurately calculating the optimal position parameters of the loudspeaker unit, especially the depth offset parameter, through a path difference compensation model, thereby achieving precise control of the sound wave interference effect. The embodiments of this invention mainly include: a loudspeaker array design method based on the path difference compensation model, a loudspeaker array structure with a two-row, two-column matrix layout, a loudspeaker array structure with a single-row, linear layout, and its applications in various electronic products. These will be described in detail below.

[0031] Figure 1 This is a flowchart illustrating a loudspeaker array optimization design method based on acoustic path difference compensation, provided in this application. Figure 1 As shown, the speaker array optimization design method provided in this embodiment includes the following steps: See Figure 1 The speaker array design method of the present invention includes the following steps: Step S1: Establish a parametric 3D model of the speaker array. In this step, a parametric three-dimensional model of the speaker array is established. The speaker array includes a cabinet and multiple speakers mounted on the front panel of the cabinet. The speakers are divided into four channel groups: right channel group, reverse right channel group, left channel group, and reverse left channel group.

[0032] Specifically, the process of creating a parametric 3D model is as follows: First, the basic geometric parameters of the speaker enclosure are determined. The enclosure is the supporting structure of the speaker array, and its geometry and dimensions directly affect the speaker's installation position and sound wave propagation characteristics. In this embodiment, the geometric parameters of the enclosure include: enclosure width W, enclosure height H, and enclosure depth D. These parameters can be determined according to the shape of the target application device. For example, for portable speakers, the enclosure width can be set to 15cm to 30cm, the enclosure height can be set to 10cm to 20cm, and the enclosure depth can be set to 8cm to 15cm; for smart display devices, the enclosure width can be set to 30cm to 100cm, the enclosure height can be set to 5cm to 15cm, and the enclosure depth can be set to 3cm to 10cm.

[0033] Secondly, the specifications of the loudspeaker must be determined. A loudspeaker is a transducer that converts electrical signals into sound signals, and its specifications directly affect the audio playback quality. In this embodiment, the loudspeaker specifications include: loudspeaker diaphragm diameter, frequency response range, rated power, sensitivity, and impedance. Preferably, this embodiment uses a full-range loudspeaker unit with a diaphragm diameter of 2 inches to 4 inches, a frequency response range covering 80Hz to 20kHz, a rated power of 5W to 20W, a sensitivity of 85dB to 92dB, and an impedance of 4Ω or 8Ω.

[0034] Next, a spatial position model of the loudspeakers is established. In the parametric 3D model, the spatial position of each loudspeaker is determined by three-dimensional coordinates (x, y, z). The x-coordinate represents the loudspeaker's position along the width of the enclosure, the y-coordinate represents its position along the height of the enclosure, and the z-coordinate represents its position along the depth of the enclosure. By adjusting these coordinate parameters, parametric design of the loudspeaker layout can be achieved.

[0035] Finally, the division of the four channel groups is defined. One of the core innovations of this invention lies in dividing the loudspeaker into four channel groups: right channel group, reverse right channel group, left channel group, and reverse left channel group. The functions of these four channel groups are defined as follows: Right channel group: Used to play the right channel audio signal, and is the main sound source of the right channel.

[0036] Left channel group: Used to play left channel audio signals, and is the main sound source of the left channel.

[0037] Reverse right channel group: Used to play audio signals that are out of phase with the right channel group, canceling out the sound waves of the right channel group at the listener's left ear.

[0038] Reverse left channel group: Used to play audio signals that are out of phase with the left channel group, canceling out the sound waves of the left channel group at the listener's right ear.

[0039] It is important to note that "phase out of phase" refers to two audio signals being 180° out of phase, meaning the positive half-cycle of one signal corresponds to the negative half-cycle of the other. When two phase out of phase signals arrive at the same location simultaneously, destructive interference occurs, and the sound waves cancel each other out. This invention utilizes this acoustic principle to achieve sound wave cancellation at the opposite ear through a reverse channel group, thereby achieving spatial separation of the left and right channels.

[0040] In a parametric 3D model, the division of the four channel groups can be done in different ways. In an embodiment using four loudspeakers, each channel group may include one loudspeaker. In an embodiment using more loudspeakers, each channel group may include multiple loudspeakers, which may be connected in parallel or in series to form the sound-generating unit of that channel group.

[0041] The parametric 3D model can be created using computer-aided design (CAD) software. Preferably, 3D modeling software such as SolidWorks, CATIA, and Pro / E can be used to create the parametric 3D model of the speaker array. During the modeling process, the cabinet geometry parameters, speaker specification parameters, and speaker spatial position parameters are defined as variable parameters, which facilitate subsequent adjustments using numerical optimization algorithms.

[0042] Step S2: Set the listening area parameters, and set multiple sampling positions within the range defined by the listening area parameters. In this step, we set the listening area parameters. The listening area refers to the spatial range in which the listener might be located; it is the target area for acoustic simulation and optimization. The setting of the listening area parameters directly affects the applicability and effectiveness of the optimization results.

[0043] The listening area parameters include the following: Listening distance range: Listening distance refers to the vertical distance from the listener to the front panel of the speaker array. In this embodiment, the listening distance range is set considering the usage scenario of the target application device. For near-field to mid-field listening devices such as portable speakers and smart display devices, the typical listening distance range is 0.5m to 3.0m. This range covers common scenarios such as near-field use on a desktop (0.5m to 1.0m) and mid-field use in a living room (1.5m to 3.0m).

[0044] Hearing height range: Hearing height refers to the height of the listener's ear relative to the ground. In this embodiment, the hearing height range takes into account typical sitting and standing postures. For sitting usage scenarios (such as desktop office work, living room movie watching), the hearing height range can be set to 1.0m to 1.3m; for standing usage scenarios, the hearing height range can be set to 1.5m to 1.8m. Preferably, this embodiment mainly focuses on sitting usage scenarios, and the hearing height range is set to 1.1m to 1.3m.

[0045] Listening width range: Listening width refers to the horizontal offset range of the listener relative to the central axis of the speaker array. In this embodiment, considering that the listener may move within a certain range, the listening width range can be set to offset 0.3m to 0.5m to the left and right of the central axis of the speaker array. For scenarios where the user is mainly used in a fixed position (such as a desktop office), the listening width range can be set smaller; for scenarios that need to cover a wider area (such as multiple people watching a movie in a living room), the listening width range can be set larger.

[0046] After setting the listening area parameters, multiple sampling locations need to be set within the listening area. These sampling locations are the specific calculation points for acoustic simulation, and their selection requires a balance between computational accuracy and efficiency.

[0047] Sampling positions can be set using methods such as uniform sampling, non-uniform sampling, or targeted sampling. Uniform sampling involves setting sampling positions at fixed intervals within the listening area. This method is simple to calculate and provides uniform coverage, but it requires a large amount of computation. Non-uniform sampling involves setting sampling positions with varying densities within the listening area based on the listener's actual usage probability or sound field characteristics, with higher sampling density in key areas and lower sampling density in less important areas. Targeted sampling involves setting more sampling positions in the locations where the listener is most likely to be present (such as the center facing the speaker array) and fewer sampling positions in the peripheral areas.

[0048] Preferably, this embodiment employs a uniform sampling method, setting a sampling point every 0.25m along the listening distance direction. For example, for a listening distance range of 0.5m to 3.0m, a total of 11 sampling points are set, located at listening distances of 0.5m, 0.75m, 1.0m, 1.25m, 1.5m, 1.75m, 2.0m, 2.25m, 2.5m, 2.75m, and 3.0m, respectively.

[0049] In the listening height direction, one or more height planes can be set as needed. For seated use scenarios, only one height plane can be set, with a height of 1.2m (typical seated ear height). For scenarios that need to cover different usage postures, multiple height planes can be set, such as three height planes at 1.1m, 1.2m, and 1.3m.

[0050] In the listening width direction, one or more width positions can be set as needed. For scenarios where the sound is mainly used facing directly, only the center position can be set. For scenarios that need to cover a certain width, multiple width positions can be set, such as the center position and positions offset to the left and right by 0.3m each.

[0051] Step S3: At each sampling location, set the left ear reference point and the right ear reference point. In this step, left and right ear reference points are set at each sampling location. These binaural reference points are virtual points used in acoustic simulation calculations to model the positions of the listener's ears, and their setting directly affects the accuracy of the sound path difference calculation.

[0052] The following principles should be followed when setting the reference points for the left and right ears: Symmetry principle: The left and right ear reference points are symmetrically distributed with respect to the sampling position. That is, in the horizontal direction, the left and right ear reference points are equidistant from the sampling position and are located on opposite sides of the sampling position. This symmetrical arrangement conforms to the geometry of the human head and facilitates acoustic simulation calculations.

[0053] Ear spacing setting: The distance between the reference points of the left and right ears is called the ear spacing, which should conform to ergonomic standards. According to anthropometric data, the average distance between the ears of an adult (i.e., head width minus the thickness of both ears) is approximately 16cm to 18cm. Preferably, in this embodiment, the ear spacing is set to 17cm, which is the statistical average distance between the ears of an adult. Of course, the ear spacing setting can also be adjusted according to the characteristics of the target user group. For example, for products targeting children, the ear spacing can be set to 14cm to 16cm; for products targeting users in specific regions, it can be adjusted according to local anthropometric data.

[0054] Height setting: The height of the left and right ear reference points should be consistent with the height of the sampling position. In other words, the binaural reference points and the sampling position should be on the same horizontal plane. This setting facilitates acoustic simulation calculations and conforms to the actual conditions of most usage scenarios.

[0055] Position Calculation: Assume the spatial coordinates of the sampling position are (x0, y0, z0), where x0 is the horizontal coordinate, y0 is the vertical coordinate, and z0 is the distance coordinate. The coordinates of the left ear reference point are (x0 - e / 2, y0, z0), and the coordinates of the right ear reference point are (x0 + e / 2, y0, z0), where e is the interauricular distance.

[0056] In practical applications, the influence of the three-dimensional shape of the ear on sound wave propagation can also be considered. A more refined model can use the head-related transfer function (HRTF) to describe the propagation characteristics of sound waves from the sound source to the ear. However, in the optimized design of this invention, to simplify calculations, the geometric sound path from the sound source to the binaural reference points is mainly considered, and the influence of ear shape and head occlusion is temporarily disregarded.

[0057] Step S4: Based on the parametric three-dimensional model, calculate the acoustic path difference at each sampling position through acoustic simulation. In this step, the acoustic path difference at each sampling location is calculated using acoustic simulation based on a parametric 3D model. The calculation of the acoustic path difference is one of the core steps of this invention, and its accuracy directly affects the quality of the optimization results.

[0058] The definition of sound path difference is as follows: Left channel path difference: This is the absolute value of the difference between the path lengths of the second and fourth speakers in the left channel group to the right ear reference point. Mathematically, it is expressed as: Δr_L = |r(LG → RE) - r(LLG → RE)| Where Δr_L is the path difference of the left channel, r(LG → RE) is the path from the left channel group to the right ear reference point, and r(LLG → RE) is the path from the reverse left channel group to the right ear reference point.

[0059] Right channel path difference: This is the absolute value of the difference between the path lengths of the first and third loudspeakers in the right channel group to the left ear reference point. Mathematically, it is expressed as: Δr_R = |r(RG → LE) - r(RLG → LE)| Where Δr_R is the path difference of the right channel, r(RG → LE) is the path from the right channel group to the left ear reference point, and r(RLG → LE) is the path from the reverse right channel group to the left ear reference point.

[0060] Take the larger of the two values ​​as the sound path difference at that sampling location.

[0061] In this embodiment, the first sound path difference is the absolute value of the difference between the sound path from the left channel group to the right ear reference point and the sound path from the reverse left channel group to the right ear reference point, and the second sound path difference is the absolute value of the difference between the sound path from the right channel group to the left ear reference point and the sound path from the reverse right channel group to the left ear reference point. The sound path difference is defined as the maximum value between the first sound path difference and the second sound path difference, that is, the larger value between the absolute value of the difference in the propagation path of the left channel and the absolute value of the difference in the propagation path of the right channel is used as the optimization target, thereby comprehensively considering the stereo effect at each position.

[0062] Calculating the sound path requires considering the propagation path of the sound wave in the air. In the simplest case, it can be assumed that the sound wave propagates in a straight line, and the sound path is the geometric distance from the sound source to the receiving point. However, in reality, sound waves are affected by factors such as diffraction at the edge of the enclosure, ground reflection, and wall reflection. To improve the accuracy of the calculation, acoustic simulation methods can be used to calculate the sound path.

[0063] Acoustic simulation methods include the Boundary Element Method (BEM), the Finite Element Method (FEM), and Ray Tracing. Preferably, this embodiment uses the Boundary Element Method for acoustic simulation calculations.

[0064] The basic principle of the boundary element method (BEM) is to transform the acoustic governing equations (Helmholtz equations) into boundary integral equations. By dividing the boundary into a mesh, the continuous problem is discretized into a discrete problem for solution. The advantage of the BEM is that it only requires meshing on the boundary, resulting in relatively low computational cost, making it suitable for handling acoustic problems in infinite or semi-infinite domains.

[0065] The specific steps of acoustic simulation using the boundary element method are as follows: Step 1: Establish the boundary element model. Divide the speaker array's cabinet surface, speaker diaphragm surface, etc., into boundary elements. The mesh size selection needs to meet the calculation accuracy requirements; generally, the mesh size should not exceed 1 / 10 of the speaker diaphragm diameter. For example, for a speaker with a diaphragm diameter of 7.5cm (3 inches), the mesh size should not exceed 0.75cm.

[0066] Step 2: Set boundary conditions. Set the speaker diaphragm surface as a velocity boundary condition, with the vibration velocity determined by the speaker input signal. Set the cabinet surface as a rigid boundary condition (normal velocity is zero), or set an impedance boundary condition based on the actual material properties.

[0067] Step 3: Solve the boundary integral equations. Based on the boundary integral form of the Helmholtz equations, establish a system of linear equations between the boundary elements to solve for the sound pressure and normal velocity distribution on the boundary.

[0068] Step 4: Calculate the sound pressure at the field point. Based on the sound pressure and normal velocity distribution on the boundary, calculate the sound pressure at any field point (including the binaural reference point) within the listening area.

[0069] Step 5: Extract sound path information. Calculate the sound path from the sound source to the field point based on the phase information of the sound pressure. The relationship between sound path and sound pressure phase is as follows: r = φ × c / (2πf) Where r is the sound path, φ is the phase (radians), c is the sound speed (m / s), and f is the frequency (Hz).

[0070] The frequency range of the acoustic simulation should cover the main frequency bands that the human ear is sensitive to. The human ear's hearing frequency range is 20Hz to 20kHz, with the highest sensitivity to the 1kHz to 4kHz band. In this embodiment, the simulation frequency range is set to 200Hz to 10kHz, covering the main frequency bands from low to high frequencies.

[0071] In acoustic simulation calculations, environmental parameters also need to be set, including temperature, relative humidity, and atmospheric pressure. These parameters affect the speed of sound and air absorption characteristics. Preferably, in this embodiment, the temperature is set to 20°C, the relative humidity to 50%, the atmospheric pressure to 101.325 kPa, and the speed of sound to 343 m / s.

[0072] Step S5: Using a numerical optimization algorithm, with the objective of minimizing the maximum absolute value of the sound path difference at all sampling locations, obtain the optimal loudspeaker layout parameters. In this step, a numerical optimization algorithm is employed to minimize the maximum absolute value of the sound path difference across all sampling locations, thereby obtaining the optimal loudspeaker layout parameters. This is the core innovative step of the invention: automatically finding the optimal loudspeaker layout parameters through an optimization algorithm.

[0073] The mathematical description of the optimization problem is as follows: Objective function: F(parameters) = max{max_{i=1,...,N} |Δr_L(i)|, max_{i=1,...,N} |Δr_R(i)|} Optimization objective: min F (parameters) Where N is the total number of sampling positions, Δr_L(i) is the left channel path difference at the i-th sampling position, and Δr_R(i) is the right channel path difference at the i-th sampling position.

[0074] The physical meaning of this objective function is to find the maximum value between the left and right channel path differences across all sampling locations, and then optimize the speaker layout parameters to minimize this maximum value. This optimization strategy ensures that even in the worst-case scenario, the path difference remains within a small range, thereby guaranteeing good stereo sound throughout the entire listening area.

[0075] The choice of optimization parameters depends on the layout of the speaker array. For a rectangular layout (two rows and two columns), the main optimization parameter is the depth offset ΔL. The depth offset refers to the distance difference between adjacent rows of speakers in the direction perpendicular to the front panel of the cabinet. By adjusting the depth offset, the propagation path of the sound waves can be changed, thereby adjusting the sound path difference.

[0076] For a linear layout (single row of channels), the main optimization parameter is the horizontal spacing between each channel group. Different combinations of horizontal spacing will affect the spatial distribution of the sound sources, thus affecting the path difference.

[0077] The selection of a numerical optimization algorithm needs to consider the characteristics of the objective function. Since the relationship between the sound path difference and the loudspeaker layout parameters is non-linear and may have multiple local optima, a global optimization algorithm is suitable. Preferably, this embodiment uses a genetic algorithm or a particle swarm optimization algorithm.

[0078] Genetic Algorithm (GA) is an optimization algorithm that simulates natural selection and genetic mechanisms. Its basic idea is to encode the solution to the optimization problem as "chromosomes," and through genetic operations such as selection, crossover, and mutation, evolve generation by generation until the optimal solution is found. The advantages of genetic algorithms are strong global search capabilities and a low risk of getting trapped in local optima; the disadvantages are relatively slow convergence speed and the need to set many control parameters.

[0079] The main control parameters of a genetic algorithm include: Population size: The number of individuals in the population. A larger population size allows for a wider search space coverage, but also increases the computational load. Preferably, in this embodiment, the population size is set to 50 to 100.

[0080] Number of iterations: The maximum number of generations the algorithm can run. More iterations result in a higher probability of convergence, but also a longer computation time. Preferably, in this embodiment, the number of iterations is set to 100 to 300.

[0081] Crossover probability: The probability that a crossover operation will occur between two individuals. Crossover operations can generate new individuals, increasing population diversity. Preferably, in this embodiment, the crossover probability is set to 0.7 to 0.9.

[0082] Mutation probability: The probability that an individual will undergo a mutation operation. Mutation operations can introduce new genes and prevent the population from converging prematurely. Preferably, in this embodiment, the mutation probability is set to 0.05 to 0.2.

[0083] Particle Swarm Optimization (PSO) is an optimization algorithm that simulates the foraging behavior of flocks of birds. Its basic idea is to represent the solution to the optimization problem as the positions of "particles" in the search space, and to update the particle's velocity and position by tracking the individual optimal position and the global optimal position. The advantages of PSO are fast convergence speed and few control parameters; the disadvantage is that it may get trapped in local optima.

[0084] The main control parameters of the particle swarm optimization algorithm include: Particle count: The number of particles in a particle swarm. A higher particle count results in a stronger search capability, but also a greater computational load. Preferably, in this embodiment, the particle count is set to 30 to 50.

[0085] Number of iterations: The maximum number of iterations the algorithm can run. Preferably, in this embodiment, the number of iterations is set to 100 to 200.

[0086] Inertia weight: controls the degree to which particles maintain their original velocity. A larger inertia weight results in a stronger global search capability; a smaller inertia weight results in a stronger local search capability. Preferably, in this embodiment, the inertia weight is set to 0.6 to 0.8, and a linear decreasing strategy is adopted, gradually decreasing during the iteration process.

[0087] Learning factor: controls the degree to which a particle learns towards its individual optimal position and the global optimal position. Preferably, in this embodiment, the learning factor is set to c1=c2=1.5 to 2.5.

[0088] In practical applications, appropriate optimization algorithms and parameters can be selected based on the characteristics of the optimization problem and the availability of computational resources. For applications seeking the global optimum, genetic algorithms can be chosen; for applications aiming for fast convergence, particle swarm optimization algorithms can be selected.

[0089] The optimization process also requires setting constraints. These constraints ensure that the optimization results meet the needs of practical applications. In this embodiment, a sound path difference threshold is set as a constraint, requiring that the absolute value of the sound path difference at all sampling locations be less than a preset threshold. The sound path difference threshold ranges from 0.5cm to 3.0cm. This threshold range is set based on the following considerations: The speed of sound in air is approximately 343 m / s, and the time difference corresponding to a path difference of 0.5 cm to 3.0 cm is approximately 0.014 ms to 0.087 ms. At a frequency of 1 kHz, the period is 1 ms, and the phase difference is approximately 21° to 31°; at a frequency of 500 Hz, the period is 2 ms, and the phase difference is approximately 10° to 16°. Within this phase difference range, the destructive interference effect of two antiphase signals is still good, and effective sound wave cancellation can be achieved.

[0090] Step S6: Output the optimal speaker layout parameters to guide the manufacturing of the speaker array. In this step, the optimal speaker layout parameters obtained by the optimization algorithm are output. These parameters will be used to guide the manufacturing and assembly of the speaker array.

[0091] The optimal speaker layout parameters output include the following: Speaker spatial coordinates: The precise position of each speaker in three-dimensional space, including the x-coordinate (horizontal position), y-coordinate (vertical position), and z-coordinate (depth position). These coordinate parameters can be directly used for the positioning and machining of speaker mounting holes.

[0092] Depth offset ΔL (for rectangular layouts): The distance difference in the depth direction between the channel group and the reverse channel group. This parameter is a key optimization parameter for rectangular layout embodiments, with a value ranging from 0.5cm to 3.0cm. The depth offset needs to be pre-allocated in the cabinet design and precisely controlled during manufacturing.

[0093] Horizontal spacing parameters (for linear layouts): The combination of horizontal spacing between each channel group. These parameters determine the horizontal distribution of the speakers and need to be precisely positioned on the front panel of the enclosure.

[0094] The output format can be engineering drawings, parameter tables, or 3D model files. Preferably, the optimal parameters are output as engineering drawings that include annotations for the speaker installation positions, making it convenient for manufacturing personnel to use directly.

[0095] During manufacturing, precise control of speaker layout parameters is required. In particular, the depth offset ΔL must be controlled within ±0.5mm to ensure effective sound path compensation. Tolerance control can be achieved through high-precision positioning fixtures, automated assembly robots, or online inspection systems.

[0096] Figure 2 This is a front view of a rectangular layout speaker array provided in an embodiment of the present invention. Figure 3 This is a side view of a rectangular layout speaker array provided in an embodiment of the present invention. The following is in conjunction with… Figure 2 and Figure 3 Describe in detail the structural features of a rectangular layout speaker array.

[0097] like Figure 2 As shown, the rectangular loudspeaker array includes a first enclosure 10 and four loudspeakers 20, 30, 40, and 50. The four loudspeakers are arranged in a rectangular layout in two rows and two columns on the front panel of the enclosure.

[0098] Specifically, the top row, from left to right, consists of the first speaker 20 and the third speaker 40, while the bottom row, from left to right, consists of the second speaker 30 and the fourth speaker 50. This two-row, two-column layout has the advantages of compact structure and uniform sound source distribution.

[0099] The channel groups for the four speakers are divided as follows: First loudspeaker 20: Right channel group. The first loudspeaker 20 is used to play the right channel audio signal and is the main sound source of the right channel.

[0100] Second speaker 30: Left channel group. The second speaker 30 is used to play the left channel audio signal and is the main sound source of the left channel.

[0101] Third speaker 40: Reverse right channel group. The third speaker 40 is used to play an audio signal that is out of phase with the right channel group, canceling the right channel sound wave emitted by the first speaker 20 at the listener's left ear.

[0102] Fourth speaker 50: Reverse left channel group. The fourth speaker 50 is used to play an audio signal that is out of phase with the left channel group, canceling out the left channel sound wave emitted by the second speaker 30 at the listener's right ear.

[0103] from Figure 2 As can be seen, the first speaker 20 and the second speaker 30 are located in the same column (left column), and the third speaker 40 and the fourth speaker 50 are located in the same column (right column). Horizontally, there is a certain spacing between the left and right columns; this spacing affects the separation of the left and right channels and the stereo effect.

[0104] Figure 3 This is a side view of a rectangular speaker array, showing the design of the depth-offset structure. From Figure 3 It can be clearly seen that the first speaker 20 is offset rearward by a depth offset ΔL relative to the second speaker 30 in the direction perpendicular to the front panel of the enclosure. Similarly, the fourth speaker 50 is offset rearward by the same depth offset ΔL relative to the third speaker 40 in the direction perpendicular to the front panel of the enclosure.

[0105] The physical meaning of the depth offset ΔL is: the offset distance of the channel group loudspeakers (first loudspeaker 20 and fourth loudspeaker 50) relative to the reverse channel group loudspeakers (second loudspeaker 30 and third loudspeaker 40) in the depth direction. This depth offset structure is one of the core innovations of this invention, and its function is to compensate for the dynamic changes in sound path difference over a wide listening distance.

[0106] Specifically, the working principle of the depth offset ΔL is as follows: Assume the listener is located at a certain distance directly in front of the speaker array. The sound wave emitted by the right channel group (first speaker 20) travels to the listener's left ear with a path length of r1; the opposite-phase sound wave emitted by the right channel group (third speaker 40) travels to the listener's left ear with a path length of r3. Because the first speaker 20 and the third speaker 40 are spaced apart horizontally and offset in the depth direction, r1 and r3 are generally not equal, and their difference is... This is the sound path difference between the right channel and the left ear. Similarly, the sound path difference between the left channel and the right ear can be defined. .

[0107] If r1 = r3 and the two speakers play out-of-phase signals, complete destructive interference will occur at the listener's left ear, and the right channel sound will be almost inaudible. If r1 ≠ r3, the destructive interference will be incomplete, and the left ear will hear part of the right channel sound, affecting stereo separation.

[0108] Under ideal conditions of complete destructive interference, it is necessary to = 0 and = 0. At this point, the left ear cannot hear the right channel sound, and the right ear cannot hear the left channel sound, achieving optimal stereo separation. Traditional dipole designs typically only approximately satisfy this condition at a specific "sweet spot" distance.

[0109] In traditional dipole speaker designs, the path difference is typically optimized for a specific listening distance, ensuring that r1 ≈ r3 at that distance. However, at other listening distances, the difference between r1 and r3 increases due to changes in the sound wave propagation angle, leading to a decrease in destructive interference. This is the reason for the "sweet spot effect" in traditional dipole speakers.

[0110] The core of this invention lies in: by introducing a depth offset ΔL, optimizing the relative position of the speaker, so that within a wide listening distance range (e.g., 0.5m to 3.0m), and The absolute values ​​are minimized together. Depth offset alters the propagation geometry of sound waves, dynamically compensating for changes in sound path difference caused by variations in listening distance.

[0111] Preferably, the objective function is defined as the path difference between the left and right ears at all sampling points within the target listening area. or The maximum value of ). The optimal depth offset ΔL can be determined through acoustic simulation and numerical optimization (as described in Example 1).

[0112] The optimization process of depth offset is illustrated below through a specific case.

[0113] Assuming the cabinet width is 25cm and the horizontal speaker spacing is 16cm, and assuming a symmetrical layout, the sampling position is on the central axis of the speaker system, = The optimization objective can be simplified to minimizing the monoaural path difference. Without introducing depth offset (ΔL = 0), the path differences at different listening distances are shown in the table below:

[0114] As can be seen from the table above, without introducing depth offset, the sound path difference decreases as the listening distance increases. However, even at a distance of 3.0m, the sound path difference is still 2.6cm, which exceeds the preset threshold of 2.0cm. Therefore, the separation is still not ideal when listening in the far field.

[0115] After introducing a depth offset ΔL = 2.0cm, the sound path difference at different listening distances is shown in Table 1:

[0116] Table 1 As can be seen from the table above, after introducing depth offset, the sound path difference at each listening distance is significantly reduced, with the maximum sound path difference decreasing from 6.2cm to 1.9cm, which meets the requirement of the preset threshold (2.0cm).

[0117] More preferably, when the depth offset ΔL = 2.5cm, as shown in Table 2, the acoustic path difference is further reduced:

[0118] Table 2 As can be seen from the data above, the selection of the depth offset ΔL requires comprehensive consideration of factors such as cabinet size, speaker spacing, and target listening distance range. The optimization design method described in Example 1 can determine the optimal depth offset for a specific product design.

[0119] The data in the aforementioned table are all taken from sampling points on the central axis directly in front of the listening area, used to visually demonstrate the trend of sound path difference changing with distance and the compensation effect of depth offset. However, the optimization target of this invention is not only the central axis, but the entire target listening area including the central axis and the positions on both sides.

[0120] To illustrate this further, we will now analyze a typical sampling point P that is off-center. Assume this point is located 1.5 meters directly in front of the speaker array, but 0.5 meters horizontally off-center (i.e., on the right).

[0121] Calculate the binaural path difference at point P under an unoptimized (ΔL=0) symmetrical layout: Because the listener is off-center, the path difference from the right channel group to the left ear increases, and the path difference ΔL_P to the left ear is 5.1cm.

[0122] At the same time, the path difference from the left channel group to its right ear decreases, and the right ear sound path difference ΔR_P = 3.0cm.

[0123] The acoustic path difference at this sampling point is S_P = max(5.1cm, 3.0cm) = 5.1cm.

[0124] Comparing Table 1, the sound path difference at the 1.5-meter central axis is 3.9 cm. It can be seen that after deviating from the center, the sound path difference of the worst ear (the left ear in this case) deteriorates significantly (from 3.9 cm to 5.1 cm), which is a manifestation of the "sweet spot effect"—the listening experience deteriorates as soon as the listener moves slightly.

[0125] With the optimized depth offset (ΔL=2.0cm) layout, recalculate the same point P: The path difference ΔL_P in the left ear decreased to 1.8 cm.

[0126] The sound path difference ΔR_P in the right ear decreased to 1.1 cm.

[0127] The acoustic path difference at this point is S_P = max(1.8cm, 1.1cm) = 1.8cm.

[0128] Compared to before optimization, the maximum sound path difference at this lateral position was significantly reduced from 5.1cm to 1.8cm.

[0129] This example clearly demonstrates that the maximum value of the sound path difference across all sampling positions, minimized by the optimization method of this invention, is likely to occur at lateral positions deviating from the central axis (such as 5.1 cm in this example), rather than directly in front. Traditional methods that only optimize the "sweet spot" directly in front cannot improve this problem. The optimized depth offset (ΔL = 2.0 cm) not only improves the performance along the central axis (as shown in Table 2), but also simultaneously and significantly improves the sound field performance at lateral positions, raising the overall lower limit of the auditory experience over a wide area. It is precisely by systematically minimizing the sound path difference at these "lateral poor spots" that the homogenization of the sound field across the entire predetermined listening area is truly achieved, reducing the sensitivity to listening position.

[0130] Therefore, the core advantage of the optimization design method provided by this invention lies in automatically finding the best speaker spatial layout parameters that can simultaneously balance the performance of all sampling points in the central axis and lateral, near field and far field through acoustic modeling and numerical optimization, thereby fundamentally overcoming the "sweet spot effect".

[0131] The cabinet structure design of a rectangular loudspeaker array needs to consider the following factors: Front panel design: The front panel is the mounting surface for the speakers and needs to have pre-drilled mounting holes and sound wave outlets. The positional accuracy of the mounting holes directly affects the realization of the speaker layout parameters. Preferably, the positional tolerance of the mounting holes is controlled within ±0.5mm.

[0132] Depth Direction Space: Due to the introduction of the depth offset structure, the enclosure needs to provide sufficient depth direction space. Preferably, the enclosure depth should be at least 3cm to 5cm greater than the maximum depth offset to accommodate the speaker magnetic circuit system and reserve installation space.

[0133] Acoustic cavity design: After the loudspeaker is mounted on the enclosure, an acoustic cavity is formed inside the enclosure. The volume and shape of the cavity affect the low-frequency response characteristics of the loudspeaker. Preferably, a suitable cavity volume is designed according to the loudspeaker's specifications and target frequency response characteristics. For full-range loudspeakers, the cavity volume can be set to 0.7 to 1.5 times the equivalent volume of the loudspeaker.

[0134] Structural strength design: The enclosure needs to have sufficient structural strength and vibration resistance to avoid enclosure resonance caused by speaker vibration. Preferably, the enclosure wall thickness is set to 3mm to 10mm, and reinforcing ribs can be added to key areas.

[0135] Figure 4 This is a front view schematic diagram of the single-row linear layout speaker array structure provided in this application. The following is in conjunction with... Figure 4 Describe in detail the structural features of a linear loudspeaker array.

[0136] like Figure 4 As shown, the linear speaker array includes a cabinet 10 and four speakers 20, 30, 40, and 50. The four speakers are arranged horizontally on the front panel of the cabinet, from left to right as the first speaker 20, the second speaker 30, the third speaker 40, and the fourth speaker 50.

[0137] The channel group division and connection method of the four speakers are as follows: The right channel group consists of a first speaker 20 and a third speaker 40 connected in series with their phases reversed. It is connected to the right channel amplifier and used to play the right channel audio signal. Specifically, the first speaker 20 plays the original right channel signal, and the third speaker 40 plays its inverted signal.

[0138] Left channel group: Composed of the second speaker 30 and the fourth speaker 50 connected in series with their phases reversed, and connected to the left channel amplifier for playing the left channel audio signal. The second speaker 30 plays the original left channel signal, and the fourth speaker 50 plays its inverted signal.

[0139] This grouping method ensures that the two speakers in the same channel group (20 and 40, 30 and 50) play out-of-phase signals, thereby creating destructive interference at the listener's opposite ear and improving stereo separation.

[0140] from Figure 4 As can be seen, the characteristic of a linear layout is that all speakers are located on the same horizontal line, with no offset in the depth direction. This layout is suitable for ultra-thin devices with limited cabinet thickness, such as televisions and ultra-thin smart displays.

[0141] In a linear layout, path compensation is achieved by optimizing the horizontal spacing between each channel group. Specifically: The horizontal distance between the first speaker 20 and the second speaker 30 is L1, the horizontal distance between the second speaker 30 and the third speaker 40 is L2, and the horizontal distance between the third speaker 40 and the fourth speaker 50 is L3.

[0142] Based on the above channel grouping, the center distance between two loudspeakers within the same channel group (i.e., the core optimization variable D) can be derived from L1, L2, and L3: Right channel center distance: ; Left channel center distance: ; The optimal combination of horizontal spacing (L1, L2, L3) can be determined using the optimization design method described in Example 1. Preferably, the values ​​of L1, L2, and L3 range from 4cm to 10cm.

[0143] The optimization process of horizontal spacing is explained below using specific numerical values.

[0144] Assuming the enclosure width is 30cm and the horizontal speaker spacing combinations are L1=6cm, L2=8cm, and L3=6cm, the sound path differences at multiple sampling positions at different listening distances are shown in the table below:

[0145] As can be seen from the table above, the horizontal spacing combinations L1=6cm, L2=8cm, and L3=6cm can achieve good sound path compensation within the listening distance range of 0.5m to 3.0m, with a maximum sound path difference of 2.2cm, which meets the requirements of the preset threshold.

[0146] Another preferred combination of horizontal spacing is L1=5cm, L2=10cm, L3=5cm, with the following path difference data:

[0147] The data above shows that the combination of L1=5cm, L2=10cm, and L3=5cm results in a smaller path difference at longer listening distances, but a slightly larger path difference at closer listening distances. This combination is suitable for scenarios primarily used for mid-to-far field listening.

[0148] Comparing the data in the two tables reveals the following: Scheme 2 (L1=5, L2=10, L3=5) performs better in the far field (≥2.0m), and its maximum sound path difference (1.0cm) at 3.0m is better than that of Scheme 1 (1.1cm).

[0149] However, Scheme 2 has a significant cost in the near field (0.5m), with its maximum sound path difference (2.8cm) being higher than that of Scheme 1 (2.4cm) and closer to the upper limit of the threshold.

[0150] This difference reflects the trade-offs in the optimized design: Option 1 (smaller L2) is more balanced overall; Option 2 (larger L2) sacrifices some near-field performance in exchange for better far-field performance.

[0151] Therefore, the final selection of specific spacing combinations needs to be determined based on the product's main usage scenario (such as whether it's primarily for near-field desktop listening or far-field living room viewing). The optimization design method provided by this invention can quantitatively provide the optimal solution and its performance prediction under different design preferences, guiding engineering implementation.

[0152] The advantages of a linear layout include: Ultra-thin design: All speakers are located on the same plane, eliminating the need for depth space, making it suitable for ultra-thin device designs. The cabinet thickness can be as thin as 3cm to 5cm, meeting the ultra-thin requirements of products such as televisions and smart displays.

[0153] Simple manufacturing: All speakers are mounted on the same plane, simplifying the installation process and making it easy to control positioning accuracy. No complex depth offset structure is required, reducing mold costs and assembly difficulty.

[0154] Wide applicability: The linear layout allows for flexible adjustment of the horizontal spacing based on the cabinet width, accommodating products of different sizes. From portable devices to large display devices, the linear layout can be used.

[0155] The limitations of a linear layout include: Horizontal space requirements: Since all speakers are arranged horizontally, a wider enclosure is needed to accommodate the four-channel group. For portable devices with limited enclosure width, smaller speaker sizes or a reduced number of channels may be necessary.

[0156] Limited sound path compensation capability: Compared to the depth offset structure of the rectangular layout, the horizontal spacing optimization of the linear layout has limited ability to compensate for sound path differences. In extreme cases (such as the very near field or the very far field), the sound path difference may exceed the preset threshold.

[0157] Figure 5 This is a schematic diagram of the sound path compensation geometry principle provided in an embodiment of the present invention. The following is in conjunction with… Figure 5 The sound path compensation principle of the present invention will be described in detail.

[0158] like Figure 5 As shown, the listener is positioned directly in front of the speaker array, with their head at sampling position P. The listener's left ear is located at the left ear reference point LE, and their right ear is located at the right ear reference point RE. The distance between the left ear reference point LE and the right ear reference point RE is the interaural distance e (approximately 17 cm).

[0159] The loudspeaker array includes a left channel group (LG), a right channel group (RG), a reverse left channel group (LLG), and a reverse right channel group (RLG). For ease of explanation, Figure 5 The text only demonstrates the principle of left channel separation, which is similar to the principle of right channel separation.

[0160] Left channel separation principle: The sound wave emitted by the left channel group LG propagates to the right ear reference point RE with a sound path of r (LG→RE). The out-of-phase sound wave emitted by the reverse left channel group LLG propagates to the right ear reference point RE with a sound path of r (LLG→RE).

[0161] When r(LG→RE) = r(LLG→RE), the two sound waves arrive at the right ear reference point RE simultaneously. Since the two sound waves are out of phase (180° out of phase), they undergo complete destructive interference at the right ear reference point RE, and the right ear can hardly hear the sound from the left channel.

[0162] When r(LG→RE) ≠ r(LLG→RE), the two sound waves arrive at the right ear reference point RE at different times, and the phase difference is no longer 180°, resulting in incomplete destructive interference. The greater the sound path difference, the greater the phase deviation, and the worse the destructive interference effect.

[0163] The definition of the path difference in the left channel is: Δr_L = r(LG→RE) - r(LLG→RE) To achieve effective sound wave cancellation, the sound path difference needs to be kept within a small range. According to acoustic theory, destructive interference is more effective when the sound path difference is less than 1 / 4 of the sound wave wavelength. At a frequency of 1 kHz, the wavelength is approximately 34 cm, and 1 / 4 of the wavelength is approximately 8.5 cm; at a frequency of 4 kHz, the wavelength is approximately 8.5 cm, and 1 / 4 of the wavelength is approximately 2.1 cm.

[0164] This invention sets the sound path difference threshold to 0.5cm to 3.0cm, corresponding to a quarter wavelength at frequencies of approximately 2.8kHz to 17kHz, and combines this with the phase-sensitive frequency band of the human ear (500Hz to 1.5kHz) and the high-frequency intensity localization mechanism. Within this threshold range, the mid-to-high frequency band (1kHz to 4kHz, the most sensitive frequency band for the human ear) can achieve a good destructive interference effect.

[0165] The principle of depth offset compensation: In a rectangular layout, the propagation path of the sound wave is changed by altering the depth offset ΔL, thereby adjusting the sound path difference.

[0166] When the listening distance is close, the sound wave propagation angle from the speaker to the listener's ear is relatively large. At this time, the horizontal spacing between the channel group and the reverse channel group has a greater impact on the sound path difference, while the depth offset has a relatively smaller impact on the sound path difference.

[0167] When the listening distance is far, the sound wave propagation angle from the speaker to the listener's ear is smaller. At this time, the horizontal spacing between the channel group and the reverse channel group has a smaller effect on the sound path difference, while the depth offset has a relatively larger effect on the sound path difference.

[0168] By optimizing the depth offset ΔL, the sound path difference between the acoustic duct group and the reverse acoustic duct group and the target ear can be kept within a small range at different listening distances. This is the principle behind how the depth offset structure can compensate for dynamic changes in the sound path difference.

[0169] Geometric criteria for applicable scenarios: For example, the technical solution of the present invention is applicable to narrow-enclosed devices with a small sound emission angle. The relationship between the sound emission angle θ and the horizontal distance L between the speakers and the listening distance d satisfies: tan(θ / 2) = (L / 2) / d That is: L = 2d × tan(θ / 2) When θ = 20°, tan(10°) ≈ 0.176, therefore L ≈ 0.35d.

[0170] In other words, when L / d ≤ 0.35, the sound-emitting angle θ ≤ 20° The human ear's ability to perceive stereo sound is closely related to the angle of sound emission. When the angle of sound emission is less than 20°, the listener has difficulty distinguishing the spatial differences between the left and right channels, and the subjective perception approaches that of a point sound source, resulting in a significant decrease in stereo sound perception. In this case, the four-channel group structure of this invention can effectively improve the stereo effect through compensation.

[0171] When the device meets this geometric condition, the stereo effect is insufficient, and the four-channel group structure of the present invention needs to be used for compensation; when the device does not meet this condition, the sound angle is large and the stereo effect is good, so the traditional two-channel structure can be used.

[0172] Figure 6 This is a circuit connection schematic diagram provided in an embodiment of the present invention. The following is in conjunction with... Figure 6 Describe in detail the circuit connection method of the speaker array.

[0173] like Figure 6 As shown, the circuit connection of the speaker array includes a left channel power amplifier 71, a right channel power amplifier 72, a left channel group speaker 73, a reverse left channel group speaker 74, a right channel group speaker 75, and a reverse right channel group speaker 76.

[0174] Left channel circuit connection: The left channel speaker 73 and the reverse left channel speaker 74 are connected in series in opposite phase and then connected to the left channel amplifier 71. The specific connection method is as follows: The positive output terminal of the left channel amplifier 71 is connected to the positive terminal ("+") of the left channel speaker 73; the negative terminal ("-") of the left channel speaker 73 is connected to the negative terminal ("-") of the reverse left channel speaker 74; and the positive terminal ("+") of the reverse left channel speaker 74 is connected to the negative output terminal of the left channel amplifier 71.

[0175] This connection method allows the same current to flow through the left channel speaker 73 and the reverse left channel speaker 74, but because the connection polarities are opposite, the two speakers vibrate in opposite directions. That is, when the diaphragm of the left channel speaker 73 is pushed outward, the diaphragm of the reverse left channel speaker 74 is pulled inward; and vice versa. This is the meaning of "anti-phase series connection".

[0176] Right channel circuit connection: The right channel speaker 75 and the reverse right channel speaker 76 are connected in series in opposite phase and then connected to the right channel amplifier 72. The connection method is similar to that of the left channel: The positive output terminal of the right channel power amplifier 72 is connected to the positive terminal of the right channel group speaker 75; the negative terminal of the right channel group speaker 75 is connected to the negative terminal of the reverse right channel group speaker 76; and the positive terminal of the reverse right channel group speaker 76 is connected to the negative output terminal of the right channel power amplifier 72.

[0177] Advantages of anti-phase series connection: The advantages of using an anti-phase series connection method are as follows: No additional circuitry required: The inverted signal is automatically generated through the speaker connection, eliminating the need for a power amplifier circuit to output the inverted signal, thus simplifying circuit design.

[0178] Phase consistency: Since the two speakers are connected in series, the current flowing through them is exactly the same, and the phase relationship is very stable. There will be no phase deviation due to differences in the power amplifier circuit.

[0179] Low cost: It does not require a separate power amplifier channel to drive the reverse channel group, which reduces the cost of the power amplifier circuit.

[0180] Simple to implement: simply connect the correct polarity of the speaker to achieve an inverted signal, which facilitates production and maintenance.

[0181] Optional implementation method: Independent power amplifier drive In addition to the aforementioned anti-phase series connection method, a separate power amplifier can also be used for driving. Specifically: Each speaker is connected to an independent amplifier channel, and the amplifier circuit outputs an inverted signal to drive the reverse channel group. The advantage of this method is that the volume and phase of each speaker can be adjusted independently, which is convenient for debugging and optimization; the disadvantage is that it requires more amplifier channels, increasing circuit cost and complexity.

[0182] Optional implementation method: Digital signal processing For high-end applications, digital signal processing (DSP) technology can be used to achieve more precise control. DSP chips can perform independent phase, delay, and equalization processing on the audio signal of each channel, achieving more accurate path compensation.

[0183] The following is a specific application case to illustrate the practical application process of the optimization design method of this invention.

[0184] Application scenario: Portable Bluetooth speaker Design goals: Box width: 20cm Box height: 15cm Box depth: 10cm Target listening distance range: 0.5m to 2.0m (primarily for desktop use) Stereo separation target: ≥12dB Design process: Step 1: Create a parametric 3D model A parametric 3D model of the speaker array was created using SolidWorks software. The cabinet dimensions were set to 20cm in width, 15cm in height, and 10cm in depth. 3-inch full-range drivers with a diaphragm diameter of 7.5cm were selected for the speakers.

[0185] A rectangular layout is used, with four speakers arranged in two rows and two columns. The speaker positions are parameterized to facilitate subsequent optimization and adjustment.

[0186] Step 2: Set the listening area parameters The listening distance is set to 0.5m to 2.0m, with a sampling point every 0.25m, for a total of 7 sampling points. The listening height is set to 1.2m (typical seated ear height). The listening width is based on the center of the speaker array, with no offset.

[0187] Step 3: Set the reference points for both ears The interauricular distance was set to 17cm. At each sampling location, a reference point for the left ear and a reference point for the right ear were set, symmetrically distributed relative to the sampling location.

[0188] Step 4: Calculate the sound path difference using acoustic simulation Acoustic simulation was performed using the boundary element method. The mesh size was set to 0.7 cm (1 / 10 of the speaker diaphragm diameter). The simulation frequency range was set to 200 Hz to 10 kHz. The environmental parameters were set to a temperature of 20℃, a relative humidity of 50%, and a sound velocity of 343 m / s.

[0189] For each sampling position, calculate the path difference between the left and right channels.

[0190] Step 5: Optimize and solve for the optimal parameters A genetic algorithm was used for optimization. The population size was set to 80, the number of iterations was set to 200, the crossover probability was set to 0.8, and the mutation probability was set to 0.1.

[0191] The optimization variable is the depth offset ΔL, with a value range of 0cm to 5cm.

[0192] The objective function is to maximize the absolute value of the acoustic path difference at all sampling locations, and the optimization objective is to minimize this maximum value.

[0193] Optimization result: Optimal depth offset ΔL = 1.8cm.

[0194] Step 6: Verify the optimization effect Using the optimized depth offset ΔL = 1.8cm, the acoustic simulation was recalculated to verify the sound path difference at each sampling position:

[0195] The maximum sound path difference is 2.1cm, which meets the requirements of the preset threshold (≤2.0cm to 3.0cm).

[0196] Step 7: Output design parameters The optimization results will be output as engineering drawings and parameter tables. The depth offset ΔL = 1.8cm, and the manufacturing tolerance is ±0.5mm.

[0197] Test and verification: A prototype was built and tested. The test results are as follows:

[0198] Test results show that the speaker array designed using the optimized design method of this invention achieves the expected stereo effect.

[0199] Based on the above embodiments, an independent subwoofer can also be added to extend the low-frequency response.

[0200] The four-channel group structure of this invention adopts a reverse series connection method, which, while achieving sound wave cancellation, will have a certain impact on low-frequency energy. Specifically, when the left channel group speaker 73 and the reverse left channel group speaker 74 play out-of-phase signals, destructive interference occurs at the listener's right ear (this is the expected effect), but partial destructive interference also occurs in the sound field near the speakers, resulting in a decrease in low-frequency energy.

[0201] To compensate for this low-frequency loss, a separate woofer can be added. A separate woofer plays the low-frequency portion of the full-range audio signal and does not participate in the path compensation mechanism, therefore it is not affected by destructive interference.

[0202] The configuration of the independent subwoofer is as follows: Speaker specifications: Dedicated woofers typically use larger woofer units, such as 5 inches, 6.5 inches, or larger. Larger woofer units offer better low-frequency response and higher sound pressure levels.

[0203] Crossover point setting: The crossover point refers to the frequency division point between the full-range speaker and the woofer. Setting the crossover point requires considering the lower low-frequency limit of the full-range speaker and the upper high-frequency limit of the woofer. Preferably, the crossover point is set between 200Hz and 300Hz. This range is below the frequency band where the human ear is most sensitive to stereo positioning (500Hz to 4kHz), therefore the independent woofer mainly handles low-frequency energy output and does not affect the stereo positioning effect.

[0204] Circuit connection: The independent subwoofer can be connected to a separate subwoofer amplifier channel, or share an amplifier channel with the full-range speaker (separated by a crossover network). Preferably, an independent subwoofer amplifier channel is used, which allows for independent adjustment of bass volume and frequency response.

[0205] Installation Location: The independent subwoofer can be installed anywhere within the enclosure, such as the front panel, side, bottom, or back. Because low-frequency sound waves are omnidirectional, the subwoofer's installation location has minimal impact on stereo performance. Preferably, the subwoofer is installed on the bottom or back of the enclosure to maintain the aesthetic appeal of the front panel.

[0206] The speaker array structure of the present invention can be applied to a variety of electronic devices, including but not limited to: Portable speakers: The width of portable speakers is typically between 15cm and 30cm, making them suitable for rectangular layout implementations. The depth offset ΔL can be optimized based on the speaker dimensions and the target listening distance.

[0207] Television: Televisions typically feature an ultra-thin design, with a cabinet thickness of less than 3cm, making them suitable for a linear layout. Speakers are arranged horizontally along the bottom bezel of the television, with the horizontal spacing optimized based on the width of the television.

[0208] Smart display devices: Smart display devices (such as smart speakers with screens, smart photo frames, etc.) are somewhere between portable speakers and televisions, and can be arranged in a rectangular or linear layout depending on their specific form.

[0209] Soundbar: A soundbar is a long, rectangular speaker specifically designed for television audio systems, with a cabinet width typically between 60cm and 120cm. Due to its larger width, it can be arranged in a straight line, allowing for greater optimization of horizontal spacing.

[0210] Portable tablets: Portable tablets have extremely thin casings and typically use a linear layout. Speakers are usually mounted on the side or bottom of the device, with horizontal spacing limited by the device's width.

[0211] Car audio systems: Car audio systems have a unique application scenario, where the listener's position is relatively fixed, and the listening distance is typically between 0.5m and 1.5m. Appropriate layout methods and parameters can be selected based on the vehicle model and installation location.

[0212] The above embodiments are merely preferred embodiments of the present invention. The present invention may also have the following alternative embodiments: 1. Although the above embodiment uses four speakers (one speaker per channel group), a greater number of speakers can also be used. For example, each channel group may include two or more speakers, connected in parallel or series, to increase the sound pressure level or extend the frequency response.

[0213] 2. Although the above embodiments use full-range loudspeakers of the same specifications, loudspeakers of different specifications can also be used. For example, the left and right channel groups can use larger loudspeakers (such as 4 inches), while the reverse left and reverse right channel groups can use smaller loudspeakers (such as 2 inches) to reduce costs and space requirements.

[0214] 3. Although the above embodiments describe two layout methods, rectangular layout and linear layout, other layout methods can also be used. For example, the speakers can be arranged in a triangle, circle or other geometric shapes, as long as the spatial distribution of the channel group and the reverse channel group can be achieved.

[0215] 4. Although the above embodiments use an anti-phase series driving method, the anti-phase signal can also be achieved by using an independent power amplifier or digital signal processing.

[0216] Finally, it should be noted that the above are merely preferred embodiments of this application and are not intended to limit the scope of protection of this application. Any modifications, equivalent substitutions, improvements, etc., made within the spirit and principles of this application should be included within the scope of protection of this application.

Claims

1. A loudspeaker array design method based on acoustic path difference compensation, characterized in that, Includes the following steps: A parametric 3D model of a loudspeaker array is established. The loudspeaker array includes a cabinet and multiple loudspeakers mounted on the front panel of the cabinet. The loudspeakers are divided into four channel groups: a right channel group, a reverse right channel group, a left channel group, and a reverse left channel group. The left channel group is used to play left channel audio signals; the right channel group is used to play right channel audio signals; the reverse left channel group is used to play audio signals that are in opposite phase to those of the left channel group; and the reverse right channel group is used to play audio signals that are in opposite phase to those of the right channel group. Set the listening area parameters, and set multiple sampling positions within the range defined by the listening area parameters; At each sampling location, a left ear reference point and a right ear reference point are set, and the left ear reference points and right ear reference points are symmetrically distributed with respect to the sampling location; Based on the parametric three-dimensional model, the acoustic path difference at each sampling position is calculated through acoustic simulation. The acoustic path difference is the maximum value between the first acoustic path difference and the second acoustic path difference. The first acoustic path difference is the absolute value of the difference between the path length from the left channel group to the right ear reference point and the path length from the reverse left channel group to the right ear reference point. The second acoustic path difference is the absolute value of the difference between the path length from the right channel group to the left ear reference point and the path length from the reverse right channel group to the left ear reference point. A numerical optimization algorithm is used to obtain the optimal loudspeaker layout parameters with the goal of minimizing the maximum sound path difference at all sampling locations; The optimal speaker layout parameters are output to guide the manufacturing of the speaker array.

2. The method according to claim 1, characterized in that, The left channel group, right channel group, reverse left channel group, and reverse right channel group each include at least one loudspeaker.

3. The method according to claim 2, characterized in that, The loudspeakers in the loudspeaker array are arranged in a rectangular layout, with the rectangular loudspeakers arranged in two rows and two columns on the front panel of the enclosure. The optimal speaker layout parameters include a depth offset, which is the distance difference between adjacent rows of speakers in the rectangular layout in the direction perpendicular to the front panel of the enclosure. The optimal value of the depth offset is determined by the numerical optimization algorithm to minimize the sound path difference of all sampling positions within the range defined by the listening area parameters.

4. The method according to claim 3, characterized in that, When determining the optimal value of the depth offset using the numerical optimization algorithm, the following constraints must be met: The absolute value of the sound path difference at all sampling locations is less than a preset threshold, which is set according to the phase resolution capability of the human ear.

5. The method according to claim 2, characterized in that, The speaker array adopts a linear layout, and the speakers in the linear layout are arranged in a straight line along the horizontal direction and installed on the front panel of the enclosure. The optimal speaker layout parameters include the horizontal spacing between each channel group, which is determined by the numerical optimization algorithm to minimize the path difference of all sampling positions within the range defined by the listening area parameters.

6. The method according to claim 1, characterized in that, The numerical optimization algorithm used is either a genetic algorithm or a particle swarm optimization algorithm.

7. The method according to claim 1, characterized in that, The acoustic simulation calculations employ the boundary element method.

8. A loudspeaker array structure based on acoustic path difference compensation, characterized in that, include: Box; Multiple loudspeakers are mounted on the enclosure, and the multiple loudspeakers are divided into four channel groups: left channel group, right channel group, reverse left channel group, and reverse right channel group. In the power amplifier circuit, the right channel group and the reverse right channel group are connected in series in opposite phase and then connected to the right channel power amplifier, and the left channel group and the reverse left channel group are connected in series in opposite phase and then connected to the left channel power amplifier.

9. The loudspeaker array structure according to claim 8, characterized in that, It also includes an independent subwoofer, which is connected to the power amplifier circuit, and the crossover point of the independent subwoofer is set between 200Hz and 300Hz.

10. An electronic device, characterized in that, Includes the speaker array structure as described in any one of claims 8 or 9, wherein the electronic device is any one of a speaker, television, smart display device, soundbar, portable tablet computer, and vehicle audio system.

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