Wide-angle adaptive directional loudspeaker system and electromechanical cooperative control method thereof

Abstract

The invention relates to the technical field of loudspeaker manufacturing, in particular to a wide-angle self-adaptive directivity loudspeaker system and an electromechanical cooperative control method thereof, and the wide-angle self-adaptive directivity loudspeaker system achieves large-range sound directivity tracking through cooperative work of mechanical steering and electronic beam forming, the wide-angle self-adaptive directivity loudspeaker system is provided with a box body, and the top of the box body is provided with a binocular high-definition camera. The box body is further connected with a mechanical steering mechanism, a loudspeaker unit array, a frequency division and signal processing unit, an MEMS microphone and a control unit are arranged in the box body, the design scheme is simple and feasible, easy to implement and suitable for various scenes, through frequency division processing and optimization design, the system structure is more compact, the cost is lower, the application range is expanded, and the application range is wider. And the limitation of a complex directivity optimization method in the prior art is overcome.

Description

Technical Field

[0001] This invention relates to the field of loudspeaker manufacturing technology, and more particularly to a wide-angle adaptive directivity loudspeaker system and its electromechanical co-control method that achieves wide-range sound directivity tracking through the coordinated operation of mechanical steering and electronic beamforming. Background Technology

[0002] With the rapid development of technology, loudspeaker systems are increasingly widely used in various fields. However, existing directional loudspeaker solutions still have many problems. First, traditional loudspeaker arrays struggle to precisely control the direction of sound propagation, resulting in excessively wide or narrow sound diffusion, affecting the listening experience. Second, inconsistent directional control across different frequencies leads to overly concentrated mid-to-high frequency sounds or excessively diffused low-frequency sounds, impacting sound quality. Furthermore, existing loudspeaker array directional optimization methods are complex, making directional control difficult. To address these issues, the industry has been exploring new technological solutions. Some solutions attempt to optimize directionality and frequency response by adjusting signals, while others focus on automatically adjusting directionality based on environmental changes or audience position. However, these solutions often suffer from operational complexity, high cost, and limited applicability, hindering their widespread adoption in practical applications. Therefore, the market urgently needs a loudspeaker system capable of precise directional control, uniform frequency response, adaptive adjustment capabilities, and effective optimization of low-frequency and mid-to-high frequency sound effects. Such a system should be able to automatically adjust directionality and frequency response based on different environments and audience positions, while also possessing strong practicality and feasibility, suitable for various scenarios.

[0003] To address issues such as inaccurate directional control, uneven frequency response, and insufficient adaptive capability in loudspeaker systems, several patent documents have been published. CN109040908A discloses a directional ring-shaped loudspeaker array and its control method. This invention solves the problem of main channel sound reinforcement when the movie screen is made of non-acoustic material by distributing multiple loudspeaker sub-arrays around a non-acoustic material screen, making it feasible to use non-acoustic material screens as movie screens and enabling adjustable directional control of the loudspeaker array. However, this invention still has room for optimization in the structural design of the loudspeaker array, making it difficult to achieve a lighter and more compact array structure.

[0004] CN106031195A discloses a loudspeaker system and a loudspeaker with controlled directionality. The invention includes multiple sound transducer units distributed on two or more surfaces of a body. Each sound transducer unit is controlled by a device for frequency-dependent level / gain adjustment and delay / phase adjustment for each independent sound transducer unit. However, this invention still requires improvement in the physical layout and acoustic performance of the complex gain and filtering devices.

[0005] CN105190743A describes a directivity adjustment device that maintains a constant direct-to-reverberation energy ratio based on a listener's detected position relative to a speaker array, including a distance estimator, a directivity compensator, and an array processor. The invention may include a distance estimator detecting the distance between the speaker array and the listener, and based on the detected distance, the directivity compensator calculating a directivity index derived from adjusting the beam generated by the speaker array to maintain a predefined direct-to-reverberation energy ratio. However, this invention struggles to adjust the directivity in real-time according to the listener's position and sound playback conditions to achieve optimal sound quality.

[0006] In summary, existing technologies still have the following drawbacks: 1. Inaccurate directivity control: Existing speaker arrays struggle to precisely control the direction of sound propagation, resulting in excessively large or small sound diffusion ranges, affecting the listening experience. Traditional speaker arrays require changes in system placement or orientation to adjust sound wave directivity or enable directional broadcasting, which is difficult and time-consuming for speaker systems placed at high locations. 2. Uneven frequency response: Inconsistent directivity control across different frequencies leads to overly concentrated mid-to-high frequency sounds or excessively diffused low-frequency sounds, affecting sound quality. For example, existing technologies require lengthening the sound tube to impart a phase difference to bass frequencies, resulting in larger speaker devices. 3. Insufficient adaptability: Existing technologies cannot automatically adjust directivity based on environmental changes or listener positions, requiring manual adjustment, which is complex. Once most placement methods are set up, their effective listening area is fixed; users must remain within that area to receive sound, and once they leave the area, the sound effect deteriorates until it becomes ineffective. 4. Insufficient Low-Frequency and Mid-High-Frequency Optimization: Poor directivity control in low and mid-high frequencies results in muddy low-frequency sound and harsh mid-high-frequency sound, affecting sound quality. Existing stereo widening solutions introduce frequency response coloration, low-frequency and center image attenuation, and transient tailing. Furthermore, they are lengthy to calibrate and lack adaptive compensation for wiring / phase errors, leading to a degraded user experience. 5. Complex Directivity Optimization Methods: Existing speaker array directivity optimization methods are complex, making directivity difficult to control. This complexity increases the difficulty and cost of system implementation, limiting its application in constrained environments.

[0007] More importantly, existing technologies, whether based on fixed array-based electronic beamforming (such as CN106031195A) or facial recognition-based dynamic beam following (such as WO2020228608A1), are limited to the signal processing level. These purely electronic solutions have inherent physical limitations: when the horizontal deflection angle of electronic beamforming is too large (usually exceeding 30°), the beam sidelobes deteriorate sharply, leading to a decrease in sound quality and inaccurate directivity. Therefore, existing technologies struggle to achieve accurate and stable sound tracking when the listener moves freely at large horizontal angles, especially in three-dimensional spaces with varying heights. The market urgently needs a system solution that can overcome the physical limitations of electronic beamforming and achieve truly comprehensive, wide-range adaptive directivity. Summary of the Invention:

[0008] This invention aims to solve the problem that existing pure electronic beamforming technology cannot effectively cope with the loss of sound directivity and deterioration of sound quality caused by the large-scale horizontal movement (>25°), rapid movement (>1m / s), and changes in pitch direction of the listener. It provides a wide-angle adaptive directivity loudspeaker system and its electromechanical co-control method that can achieve full-space horizontal and pitch, fast response, and high-precision sound tracking.

[0009] This invention achieves its purpose through the following measures:

[0010] A wide-angle adaptive directional loudspeaker system is provided, comprising a cuboid enclosure, characterized in that a binocular high-definition camera is installed on the top of the enclosure, the enclosure is also connected to a mechanical steering mechanism, and the enclosure contains a loudspeaker unit array, a frequency division and signal processing unit, a MEMS microphone and a control unit, wherein the control unit is connected to the binocular high-definition camera, the microphone and the frequency division and signal processing unit respectively, and the output terminal of the frequency division and signal processing unit is connected to the loudspeaker array unit.

[0011] The binocular high-definition camera is installed on the top of the speaker to take a picture of the listener's position. The image processing software is used to analyze the two pictures taken by the binocular camera to obtain the listener's angle θ relative to the central axis and the distance relative to the sound source.

[0012] The mechanical steering mechanism is also connected to the control unit and is located outside the entire enclosure. It is used to adjust the orientation of the entire enclosure and is suitable for situations where the listener's range of motion deviates from the positive angle of the enclosure by more than 30°. In order to avoid exceeding the range due to mechanical response delay, a margin of 5° is left. Mechanical adjustment will begin for a range exceeding 25°.

[0013] The loudspeaker unit array consists of eight loudspeaker units, including two low-frequency loudspeaker units located at both ends of the array, employing a coil-piezoelectric speaker configuration suitable for frequencies above 10kHz; and six mid-high frequency loudspeaker units arranged in a ring in the middle, also employing a coil-piezoelectric speaker configuration suitable for frequencies above 10kHz. The ring diameter is 400±50mm, and the diaphragm area of ​​the six mid-high frequency loudspeaker units is 150±20mm². 2 The volume of the acoustic cavity is 40±5cm. 3 The sensitivity is 105±5dB. The frequency division and signal processing unit adopts a collaborative processing architecture of digital signal processor (DSP) and field-programmable gate array (FPGA). The DSP is configured to perform digital frequency division processing, parametric equalization, and beamforming of the audio signal. The frequency division point is obtained by the center position of the frequency response curves of the mid-high frequency speaker and the low frequency speaker. The low-frequency signal after frequency division is sent to the two low-frequency speaker units. The low-frequency signal refers to the signal with a frequency below 500Hz±50Hz. The mid-high frequency signal, i.e., the signal with a frequency above 500Hz±50Hz, is sent to the six mid-high frequency speaker units. The FPGA is connected to the DSP to form a digital directional adjustment of the speaker sound direction. It receives parametric equalization and beamforming coefficient instructions from the DSP. Parametric equalization is used to calibrate each speaker, and beamforming is used to adjust the phase and delay of the signal sent to the six mid-high frequency speaker units in real time, thereby realizing directional adjustment in a direction less than 25°.

[0014] The microphone is a MEMS microphone positioned at the center of the six mid-to-high frequency speaker units inside the speaker cabinet. It is used to sense the direct sound emitted by each speaker unit in a time-division manner, that is, the six speakers emit sound at different times for periodic acoustic calibration. The input of the control unit is connected to an environmental perception unit containing a MEMS microphone and a binocular camera, and the output is connected to a digital signal processor (DSP) in the frequency division and signal processing unit. The control unit receives and fuses the data from the environmental perception unit, dynamically calculates the parametric equalization and beamforming parameters required for each speaker unit in the speaker array based on the fused data, and sends the parameters to the DSP.

[0015] The housing of the present invention is provided with 12 through holes with a diameter of 8±1mm. The through holes are distributed on the left, right and rear sides of the housing, with 4 holes on each side, which are used to balance the air pressure inside and outside the housing and optimize the low frequency radiation characteristics.

[0016] The mechanical steering mechanism of this invention includes a speaker connector and a base. A universal rotating unit is mounted on the base, and its upper surface is fixedly connected to the bottom of the cabinet via the speaker connector. The universal rotating unit is cylindrical, and its outer shell has two pitch drive shafts arranged radially opposite each other. The open end of a U-shaped bracket is connected to the two pitch drive shafts, and the closed end of the U-shaped bracket is connected to the output shaft of a first stepper motor. The first stepper motor is fixed to the base via a support plate. The outer shell of the universal rotating unit also has a left-right undulating drive shaft located radially perpendicular to the pitch axis. The left-right undulating drive shaft is connected to the output shaft of a second stepper motor via a V-shaped rotating bracket. The V-shaped apex of the rotating bracket is located directly below the universal rotating unit, and one end of the rotating bracket is connected to the output shaft of the second stepper motor. The second stepper motor is fixed to the base via another support plate, and its output shaft is perpendicular to the first stepper motor's output shaft.

[0017] This invention also proposes an electromechanical cooperative control method for a wide-angle adaptive directivity loudspeaker system as described above. The method is characterized by setting a first preset threshold and a second preset threshold for angle adjustment, where the angle of the first preset threshold is greater than the angle of the second preset threshold. The control unit is configured to perform the following electromechanical cooperative control: when the angle between the listener's position and the normal to the front of the speaker is greater than the first preset threshold, the mechanical steering mechanism of the control unit adjusts the orientation of the speaker enclosure, and simultaneously controls the crossover and signal processing unit to perform electronic beamforming compensation on the loudspeaker unit array; when the angle between the listener's position and the normal to the front of the speaker is reduced to below the second preset threshold, the mechanical steering mechanism is stopped, and electronic beamforming tracking is performed solely by the crossover and signal processing unit.

[0018] It also includes setting a moving speed threshold. When the listener's moving speed is greater than the moving speed threshold, the mechanical steering mechanism and the crossover and signal processing unit work simultaneously. The crossover and signal processing unit calculates and compensates for the insufficient response of the mechanical steering mechanism to the speaker position adjustment in real time.

[0019] When the angle difference between the listener's position and the normal of the speaker's front is ≤25°, the system only uses electronic pointing compensation adjustment. The FPGA performs real-time phase and delay adjustment on the mid-to-high frequency speaker unit signal to achieve fast, silent, and precise micro-tracking. The first preset threshold can be 25°. When the angle difference is greater than 25°, the system first activates the mechanical steering mechanism for coarse positioning of the speaker enclosure, while simultaneously superimposing electronic pointing compensation adjustment to compensate for the delay of mechanical movement and achieve a smooth transition. The goal of the mechanical adjustment is to reduce the angle difference to within 5° (the second preset threshold). After that, the mechanical adjustment is stopped and the electronic system is completely handed over for fine tracking again. When the listener moves quickly, i.e., the movement speed is >1m / s, the mechanical and electronic systems work simultaneously. The electronic system calculates and compensates for the insufficient response of the mechanical system in real time.

[0020] This invention includes the following steps:

[0021] Step 1: Initialize system configuration and set system operating parameters;

[0022] Step 2: Environmental perception and data collection: During speaker idle time, the MEMS microphone is used to sense the sound intensity and phase difference of each speaker in a time-division manner, and the binocular high-definition camera is used to detect the angle of the listener relative to the normal of the front of the speaker and the distance relative to the sound source in real time.

[0023] Step 3: Parametric Equalization Calculation: The Digital Signal Processor (DSP) uses a high-precision parametric equalizer to calibrate the data collected by the microphone in Step 2 across the entire frequency band, from 20Hz to 20kHz. First, the frequency response curve is measured using the microphone. For deviations exceeding ±2dB, boosting or attenuation is performed to compensate for the speaker's own deviation and to solve the problem of uneven and non-uniform frequency response caused by standing waves and resonance. The difference in sound intensity caused by the original sound or time-lapsed sound between different speakers is compensated to make the deviation less than ±1dB. The phase of different speakers is calibrated to make the deviation less than 0.05ms.

[0024] Step 4: If the angle of change of the listener's position relative to the normal of the front of the speaker is within 25°, then perform electronic directivity compensation adjustment: Perform beamforming calculation, calculate beamforming parameters through DSP based on environmental data: Calculate the time difference required for sound to reach the listener from each speaker unit based on the listener's position parameters, including angle and distance, compensate for the delay of the electrical signals of each speaker unit through delay circuit or digital delay algorithm, calculate the phase difference of each speaker unit based on the listener's position and environmental parameters, and calculate the beamforming parameters required for each speaker unit based on these time and phase differences, so that the sound waves from different speakers at the listener's position are superimposed and enhanced, thereby achieving sound directivity; If the angle is greater than 25°, proceed to step 6;

[0025] Step 5: Fine-tune the directivity of the loudspeakers: Based on the beamforming parameters, control the loudspeaker array to split the audio signal into low-frequency and mid-high-frequency signals through a crossover. The low-frequency signal is sent to a low-frequency array consisting of two loudspeaker units, and the mid-high-frequency signal is sent to a mid-high-frequency array consisting of six loudspeaker units. Through high-speed parallel processing by FPGA, real-time delay and phase adjustment of the mid-high-frequency signals are achieved. The delay adjustment range is 0.1ms to 2ms, and the phase adjustment range is -220° to 220°.

[0026] Step 6: If the change in the listener's direction exceeds the adjustment range of 25°, mechanical adjustment is activated first, followed by electronic pointing compensation adjustment. The mechanical adjustment calculates the rotation of the two stepper motor shafts and then drives the stepper motors through the control unit to drive the angle adjustment mechanism to achieve mechanical direction adjustment of the enclosure, thus achieving a wide range of adjustment, which can at least meet the requirements of ±60°.

[0027] Step 7: Return to step 2 and start the next loop.

[0028] In step 6 of this invention, the target of mechanical adjustment is to keep the angle within 5°. During the mechanical movement, the binocular camera collects the listener's angle in real time, and the control part simultaneously determines whether the current listener's position is within the 5° range. If not, the mechanical part continues to rotate for compensation, and the electronic pointing control continues to perform compensation. This cycle repeats until the angle is less than 5°, at which point the mechanical compensation stops. At this point, the angle compensated by the electronic pointing compensation is equal to the angle between the speaker plane normal and the line connecting the speaker and the listener before the mechanical compensation was started - the angle adjusted by the mechanical part.

[0029] In step 6 of this invention, if the listener's movement includes not only a wide range but also rapid movement, with a speed greater than 1 m / s, and the mechanical response speed cannot keep up with the listener's rapid movement, electronic pointing must be superimposed simultaneously with the mechanical movement. That is, during the mechanical rotation, the mechanical control part transmits the horizontal (left-right undulation) and vertical (tilt) rotation angles to the beamforming control part in real time (the first stepper motor drives the tilting motion to adapt to changes in the listener's distance and height changes caused by standing or sitting postures; the second drive shaft generates left-right undulation motion, which, in conjunction with the first drive shaft, generates pointing in any direction facing the listener); during electronic pointing compensation adjustment, the beamforming... The beamforming control unit calculates the difference between the angle of mechanical change and the angle of the listener's actual movement obtained by the binocular camera as the compensation angle for electronic pointing. This is then fused with the listener's distance data to recalculate the beamforming parameters and adjust the beamforming in real time, ensuring that the sound is always accurately pointed at the listener. When the mechanical adjustment stops, the electronic pointing compensation operation continues. This can meet the pointing requirements of situations where the listener moves quickly and over a wide area. While the mechanical mechanism moves smoothly to achieve wide coverage, the electronic system responds at high speed to process dynamic details, thus achieving a tracking range and response speed far exceeding the limits of a single technology without sacrificing sound quality.

[0030] Step 4 of this invention is achieved through the following measures:

[0031] Step 4-1: Use DSP to calculate beamforming. Calculate beamforming parameters based on environmental data. Calculate the time difference required for sound to reach the listener from each speaker unit based on the listener's position (e.g., x = 2.0m, y = 1.4m). The velocity constant c0 = 340m / s. Assuming the distance between two adjacent speaker units is d, we want to point the main sound beam in a direction that forms an angle θ with the array normal. Then, the path length the sound travels from these two units to the listener's position will have a difference (i.e., path difference) ΔL = d * sin(θ). To make these two sound waves superimpose (enhance) in the θ direction, this path difference must be compensated. The wavelength of the sound wave λ = speed of sound c / frequency f. The time difference corresponding to this path difference ΔL is Δt = ΔL / c.

[0032] Step 4-2: The FPGA needs to send a signal Δt time in advance for the unit that is farther away from the listener, and use a digital delay algorithm to compensate for the delay of the electrical signal of each speaker unit.

[0033] Step 4-3: Make the phase of the signal from the speaker unit farther away from the listener lead by a specific angle, i.e., the phase difference. The conversion relationship between them is: phase difference φ=(ΔL / λ)*360°. Calculate the phase difference of each speaker unit according to the listener's position and environmental parameters.

[0034] Step 4-4: Based on the time difference and phase difference, calculate the beamforming parameters required for each speaker unit and obtain the phase compensation value for each unit.

[0035] This invention, by separating mechanical and digital directional adjustment of the speaker, achieves a faster system response and a smooth experience of "mechanical positioning and electronic fine-tuning," ensuring the listener is always in the optimal position within the sound field. The organic combination of digital and mechanical directional adjustment enables rapid and precise adjustment of mid-to-high frequencies within a small angle range, and effective expansion across a large angle range across the entire frequency band. Mechanical adjustment overcomes the physical limitations of electronic beamforming at large angles; electronic fine-tuning (digital directional adjustment) compensates for the shortcomings of mechanical mechanisms in terms of precision and speed, and avoids lifespan and noise issues caused by frequent, minute movements of mechanical components. For situations where the listener moves over a large area (over 25°) and moves rapidly (>1m / s), both digital and mechanical directional adjustment must work simultaneously to achieve accurate directional control.

[0036] This invention achieves a breakthrough in wide-range directional capability: through a mechanical steering mechanism, this invention effectively solves the problem of sound directionality control in the pitch direction for the first time in a practical product, enabling the sound to follow the listener's posture, such as standing or sitting, which is difficult to achieve with pure electronic beamforming solutions. Through electromechanical synergy, it achieves performance far exceeding the limits of a single technology: mechanical adjustment overcomes the physical limits of electronic beamforming at large angles, achieving wide-angle coverage of over ±60°; electronic fine-tuning compensates for the shortcomings of mechanical mechanisms in terms of precision and speed. Together, the two achieve uniform coverage with sound pressure level fluctuations of less than ±2dB within a horizontal and pitch range of ±60° at a distance of 4 meters, and a tracking delay of less than 100 milliseconds.

[0037] The number of speakers mentioned in this invention is a recommended number to achieve better results. It can be increased or decreased appropriately according to actual requirements. After the increase or decrease, the size and arrangement of individual speaker units also need to be adjusted appropriately. Among them, the mid-to-high frequency speaker units always maintain a ring structure, which makes it easy to achieve arbitrary directionality. The more speaker units there are, the more complex the control becomes, but the better the directivity. Attached Figure Description

[0038] Appendix Figure 1 This is a diagram showing one usage state of the loudspeaker system in this invention.

[0039] Appendix Figure 2 This is a schematic diagram of the layout of the through hole on the speaker system housing in this invention.

[0040] Appendix Figure 3 This is a schematic diagram of the mechanical steering mechanism in this invention, wherein... Figure 3(a) is a schematic diagram of the initial state, (b) is a schematic diagram of the first stepper motor driving the rotating unit in the tilting state, and (c) shows the second stepper motor driving the rotating unit in the left-right undulating state. Combined with the tilting state in (b), the box can be rotated in all directions in the horizontal and vertical directions, which is the key mechanical basis for realizing the sound directionality of the whole space.

[0041] Appendix Figure 4 This is a schematic diagram of the loudspeaker system in this invention.

[0042] Appendix Figure 5 These are the frequency response curves before and after equalization in this embodiment of the invention.

[0043] Appendix Figure 6 This is an embodiment of the present invention, a phase and delay calculation table (distance 4m).

[0044] Appendix Figure 7 This is a flowchart of the present invention.

[0045] Attached reference numerals: 1. Speaker connector; 2. Base; 3. Universal rotating unit; 4. Cabinet; 5. U-shaped bracket; 6. First stepper motor; 7. Support plate; 8. Second stepper motor; 9. Rotating bracket. Detailed Implementation

[0046] The present invention will be further described below with reference to the accompanying drawings and embodiments.

[0047] This invention addresses the inherent limitations of existing technologies for adjusting sound wave directionality using single techniques: purely mechanical adjustments are slow, noisy, and prone to wear, making smooth sound field fine-tuning difficult; purely electronic beamforming is inefficient at low frequencies and wide angles, and has physical limitations on array size, with directional control capabilities decreasing sharply as frequency decreases and angle increases. Those skilled in the art, when faced with large-angle coverage requirements, tend to design larger arrays or use multiple fixed speakers rather than considering modifying the mechanical structure and deeply coupling it with real-time signal processing algorithms. EAW's adaptive technology also primarily focuses on electronic beam adjustment, without mentioning its synergy with mechanical steering. While EAW's adaptive technology is indeed industry-leading in low-frequency control, it cannot "completely solve" the low-frequency directivity problem. It primarily maximizes control capabilities within physical limits through algorithms and array design, significantly improving low-frequency coverage and suppression under ideal conditions. However, in complex environments, with limited arrays, or with insufficient tuning, issues such as uneven energy distribution and obstruction interference may still occur.

[0048] This invention proposes an adaptive directional loudspeaker system, which mainly includes a loudspeaker array, a crossover and signal processing unit, an environmental sensing unit, and an intelligent control unit, as shown in the attached figure. Figure 4As shown, the loudspeaker array consists of eight loudspeaker units. Two low-frequency loudspeakers are placed at both ends, using a coil piezoelectric method. The other six mid-high frequency loudspeakers are located in the middle, arranged in a circular array using a combination of coil and piezoelectric methods. The piezoelectric method is suitable for frequencies above 10kHz. The six mid-high frequency loudspeaker units are evenly distributed on a circle with a diameter of 400mm. This arrangement facilitates beamforming and control in the 180° horizontal direction. The diaphragm area is 150mm². 2 The volume of the acoustic cavity is 40cm. 3 Sensitivity 105dB;

[0049] The crossover uses digital crossover (DSP implementation). Its crossover point is obtained by centering the frequency response curves of the mid-high frequency speaker and the low-frequency speaker. For example, the audio signal is crossovered at 500Hz. The low-frequency signal (below 500Hz) is sent to a low-frequency array consisting of two speaker units. These two units are driven by a power amplifier after signal preprocessing (Butterworth low-pass filter, cutoff frequency 500Hz). They are specifically responsible for powerful low-frequency reproduction. The mid-high frequency signal (above 500Hz) is sent to a mid-high frequency array consisting of six speaker units. These six units are controlled by an FPGA. By adjusting the phase and delay of the signals of each unit in real time, precise electronic beamforming is achieved to control the directivity of the mid-high frequency sound waves.

[0050] The core of the signal processing unit is a collaborative processing architecture of one DSP and one FPGA. The DSP is responsible for running upper-level algorithms, such as frequency division, parametric equalization, calculation of beamforming coefficients (the DSP calculates the initial phase and amplitude weights required by each of the six units according to the desired beam shape (such as narrower or wider) and direction, which is equivalent to setting the "initial shape and direction" of the beam), environmental perception information fusion, system control logic, etc.

[0051] In a digital signal processor (DSP), the time difference required for sound to reach each speaker unit is first calculated based on the listener's location (provided by the environmental sensing unit). Then, the electrical signal sent to each unit is precisely delayed and compensated by a delay circuit or digital delay algorithm. By changing the phase difference, the main lobe direction of the signal beam of the speaker unit can be deflected, achieving dynamic scanning or tracking. Finally, the sound waves emitted by all units are superimposed in phase in the target direction (the listener's location), resulting in the strongest energy and forming the "main beam." In other directions, they are partially canceled out due to different phases, resulting in weaker energy. The FPGA receives coefficient instructions from the DSP and performs precise real-time delay and phase adjustment on the six mid-to-high frequency signals at high speed and in parallel. This is the key to achieving beamforming.

[0052] The environmental perception unit acts as the system's "eyes and ears," including: a high-definition camera for visually identifying the location and number of listeners. Once a listener moves, the system immediately recalculates the delay parameters required for each speaker unit and dynamically adjusts them to ensure the main beam always "follows" the listener. This not only significantly improves sound clarity and immersion but also effectively reduces noise interference from the surrounding environment. The microphone is a MEMS microphone positioned at the center of the six mid-to-high frequency speaker units inside the speaker enclosure. It is used for time-division (the six speakers emit sound separately in time) sensing of the direct sound emitted by each speaker unit, enabling periodic, time-varying acoustic calibration.

[0053] The intelligent control unit is the "brain" of the system. It receives all data from the environmental sensing unit and performs fusion processing based on intelligent algorithms. This module can record optimization parameters under different environments, enabling the system to better adapt to specific environments. It dynamically calculates the optimal beamforming parameters (i.e., delay and phase values ​​of each unit) and frequency response equalization parameters of the speaker array under the current environment.

[0054] Hybrid directional control includes electronic and mechanical adjustment. The control signals generated by electronic adjustment are primarily sent to the signal processing unit (FPGA), which achieves precise and rapid electronic beam steering by altering the phase and delay of the electrical signals. The mechanical adjustment control unit is also connected to a stepper motor drive system. The speaker is connected to a miniature stepper motor, used to fine-tune the physical pitch and horizontal heading angles according to the control signal commands. This mechanical adjustment is mainly used to assist electronic beamforming, compensate for potential performance degradation of electronic methods at extreme angles, or achieve a greater range of coverage adjustments; it is an effective supplement to electronic adjustment.

[0055] As attached Figure 3As shown, the mechanical steering mechanism includes a speaker connector 1 and a base 2. A universal rotating unit 3 is provided on the base 2. The upper surface of the universal rotating unit 3 is fixedly connected to the bottom of the cabinet 4 via the speaker connector 1. The universal rotating unit 3 is cylindrical. Two pitch drive shafts are arranged in the same radial direction on opposite sides of the outer shell of the universal rotating unit 3. The open end of the U-shaped bracket 5 is connected to the two pitch drive shafts respectively. The closed end of the U-shaped bracket 5 is connected to the output shaft of the first stepper motor 6. The first stepper motor 6 is fixed on the base 2 via a support plate 7. The outer shell of the universal rotating unit 3 is also provided with a left and right undulating drive shaft. The left and right undulating drive shaft is located in the radial direction perpendicular to the pitch axis. The left and right undulating drive shaft is connected to the output shaft of the second stepper motor 8 via a V-shaped rotating bracket 9. The V-shaped apex of the rotating bracket 9 is located directly below the universal rotating unit 3. One end of the rotating bracket 9 is connected to the output shaft of the second stepper motor 8. The second stepper motor 8 is fixed on the base 2 via another supporting plate, and the output shaft of the second stepper motor 8 is perpendicular to the output shaft of the first stepper motor 6.

[0056] The system also includes a through-hole design, with 12 through-holes of 8mm diameter on the speaker array housing, four on the left, four on the right, and four at the rear. See details... Figure 2 .

[0057] These through-holes are an application of a precisely acoustically designed Helmholtz resonator, whose primary function is to balance the air pressure inside and outside the enclosure and optimize low-frequency radiation characteristics. Specifically, when the speaker units (especially the low-frequency units) operate at high power, the intense vibration of the diaphragm causes a rapid change in air pressure inside the enclosure. These through-holes provide a controlled flow path for the air inside and outside the enclosure, effectively releasing excess pressure, reducing the constraint on diaphragm movement, thereby reducing nonlinear distortion under large dynamic range signals and improving transient response. At the same time, the carefully designed aperture and layout help to disperse high-order standing waves formed inside the enclosure and make the low-frequency response smoother, improving sound clarity and low-frequency efficiency. The through-hole design of this system does not exist in isolation; it works in conjunction with adaptive beamforming and mechanical steering: the optimized low-frequency response provides a purer, lower-distortion original sound signal for electronic beamforming; the balanced air pressure allows the speaker units (especially those responsible for low frequencies) to operate in a more linear range, which is crucial for maintaining sonic consistency during a wide range of mechanical steering and dynamic beam tracking.

[0058] This system achieves hybrid directional control by combining electronic beamforming (primary) and mechanical angle adjustment (secondary), offering both flexibility and high performance. Simultaneously, through multimodal environmental perception (visual + acoustic) and intelligent algorithms, it achieves adaptive optimization in complex indoor environments, significantly improving speech clarity and listening experience.

[0059] Example 1:

[0060] This example provides an application of the adaptive directivity loudspeaker system described above, as shown in the attached figure. Figure 7 As shown, it includes the following steps:

[0061] Step 1: Initialize system configuration and set system operating parameters; after system initialization, enter a continuous environmental perception loop, with the key decision point being the judgment of angle difference (Δθ);

[0062] Step 2: Environmental perception and data collection: During speaker idle time, the MEMS microphone is used to sense the sound intensity and phase difference of each speaker in a time-division manner, and the binocular high-definition camera is used to detect the angle of the listener relative to the normal of the front of the speaker and the distance relative to the sound source in real time.

[0063] Step 3: Parametric Equalization Calculation: The Digital Signal Processor (DSP) uses a high-precision parametric equalizer to calibrate the data collected by the microphone in Step 2 across the entire frequency range (20Hz to 20kHz). First, the frequency response curve is measured using the microphone. Deviations exceeding ±2dB are boosted or attenuated to compensate for speaker inherent biases and resolve issues of uneven or non-uniform frequency response caused by standing waves and resonance. The difference in sound intensity between different speakers, whether due to natural sound or time-dependent attenuation, is compensated to reduce the deviation to within ±1dB. The phase of different speakers is calibrated to reduce the deviation to less than 0.05ms. (See attached diagram.) Figure 5 As shown;

[0064] Step 4: If the angle of change of the listener's position relative to the normal of the front of the speaker is within 25°, then perform electronic directivity compensation adjustment: Perform beamforming calculation, calculate beamforming parameters through DSP based on environmental data: Calculate the time difference required for sound to reach the listener from each speaker unit based on the listener's position parameters, including angle and distance, compensate for the delay of the electrical signals of each speaker unit through delay circuit or digital delay algorithm, calculate the phase difference of each speaker unit based on the listener's position and environmental parameters, and calculate the beamforming parameters required for each speaker unit based on these time and phase differences, so that the sound waves from different speakers at the listener's position are superimposed and enhanced, thereby achieving sound directivity; If the angle is greater than 25°, proceed to step 6;

[0065] Specifically, the steps include:

[0066] Step 4-1: Use DSP to calculate beamforming. Calculate beamforming parameters based on environmental data. Calculate the time difference required for sound to reach the listener from each speaker unit based on the listener's position (e.g., x = 2.0m, y = 1.4m). The velocity constant c0 = 340m / s. Assuming the distance between two adjacent speaker units is d, we want to point the main sound beam in a direction that forms an angle θ with the array normal. Then, the path length the sound travels from these two units to the listener's position will have a difference (i.e., path difference) ΔL = d * sin(θ). To make these two sound waves superimpose (enhance) in the θ direction, this path difference must be compensated. The wavelength of the sound wave λ = speed of sound c / frequency f. The time difference corresponding to this path difference ΔL is Δt = ΔL / c.

[0067] Step 4-2: The FPGA needs to send a signal Δt time in advance for the unit that is farther away from the listener, and use a digital delay algorithm to compensate for the delay of the electrical signal of each speaker unit.

[0068] Step 4-3: Make the phase of the signal from the speaker unit farther away from the listener lead by a specific angle, i.e., the phase difference. The conversion relationship between them is: phase difference φ=(ΔL / λ)*360°. Calculate the phase difference of each speaker unit according to the listener's position and environmental parameters.

[0069] Step 4-4: Based on the time difference and phase difference, calculate the beamforming parameters required for each loudspeaker unit and obtain the phase compensation value for each unit;

[0070] Step 5: Fine-tune the directivity of the loudspeakers: Based on the beamforming parameters, control the loudspeaker array to split the audio signal into low-frequency and mid-high-frequency signals through a crossover. The low-frequency signal is sent to a low-frequency array consisting of two loudspeaker units, and the mid-high-frequency signal is sent to a mid-high-frequency array consisting of six loudspeaker units. Through high-speed parallel processing by FPGA, real-time delay and phase adjustment of the mid-high-frequency signals are achieved. The delay adjustment range is 0.1ms to 2ms, and the phase adjustment range is -220° to 220°.

[0071] Step 6: If the change in the listener's direction exceeds the adjustment range of 25°, mechanical adjustment is activated first, followed by electronic pointing compensation adjustment. The mechanical adjustment calculates the rotation of the two stepper motor shafts and then drives the stepper motors through the control unit to drive the angle adjustment mechanism to achieve mechanical direction adjustment of the enclosure, thus achieving a wide range of adjustment, which can at least meet the requirements of ±60°.

[0072] Step 7: Return to step 2 and start the next loop.

[0073] In step 6 of this example, the goal of mechanical adjustment is to keep the angle within 5°. During the mechanical movement, the binocular camera collects the listener's angle in real time, and the control unit simultaneously determines whether the current listener's position is within the 5° range. If not, the mechanical part continues to rotate to compensate, and the electronic pointing control continues to perform compensation. This cycle repeats until the angle is less than 5°, at which point the mechanical compensation stops. At this point, the angle compensated by the electronic pointing compensation is equal to the angle between the speaker plane normal and the line connecting the speaker and the listener before the mechanical compensation starts - the angle adjusted by the mechanical part.

[0074] In step 6 of this example, if the listener's movement includes not only large-scale movement but also rapid movement (greater than 1 m / s), and the mechanical response cannot keep up with the listener's rapid movement, electronic pointing must be superimposed simultaneously with the mechanical movement. That is, during mechanical rotation, the mechanical control unit transmits the horizontal (left-right undulation) and vertical (tilt) rotation angles to the beamforming control unit in real time (the first stepper motor drives the tilting motion to adapt to changes in the listener's distance and height changes due to standing or sitting postures; the second drive shaft generates left-right undulation motion, which, in conjunction with the first drive shaft, produces pointing in any direction facing the listener). During electronic pointing compensation adjustment, the beamforming... The beamforming control unit calculates the difference between the angle of mechanical change and the angle of the listener's actual movement obtained by the binocular camera as the compensation angle for electronic pointing. This is then fused with the listener's distance data to recalculate the beamforming parameters and adjust the beamforming in real time, ensuring that the sound is always accurately pointed at the listener. When the mechanical adjustment stops, the electronic pointing compensation operation continues. This can meet the pointing requirements of situations where the listener moves quickly and over a wide area. While the mechanical mechanism moves smoothly to achieve wide coverage, the electronic system responds at high speed to process dynamic details, thus achieving a tracking range and response speed far exceeding the limits of a single technology without sacrificing sound quality.

[0075] Compared with existing technologies, the present invention has the following advantages: 1. By combining mechanical and electronic adjustments, the present invention achieves precise control of sound directivity, overcoming the problem of inaccurate directivity in traditional speaker arrays. The system can adjust the direction and range of sound propagation as needed, effectively solving the problem of poor listening experience caused by excessively large or small sound diffusion range; 2. The present invention uses frequency division processing technology to process low-frequency and mid-to-high-frequency sounds separately and optimize them respectively, achieving a uniform frequency response. By setting multiple through holes in the speaker housing, the sound of the speaker unit is reversed, effectively optimizing the sound directivity of the low-frequency range. Simultaneously, by using a mid-to-high frequency speaker array and beamforming technology, precise directivity control of mid-to-high frequency sounds is achieved, effectively solving the problems of overly concentrated mid-to-high frequency sounds or excessively diffused low-frequency sounds. 3. This invention constructs a complete adaptive system capable of automatically adjusting directivity and frequency response according to environmental changes or the listener's position. Through environmental perception and intelligent algorithms, it monitors the listener's position in real time and dynamically adjusts the directivity of the speaker array as needed, greatly improving the system's adaptability and flexibility, and solving the problem of complex manual adjustment operations in existing technologies. 4. This invention optimizes low-frequency and mid-to-high frequency sounds separately. By using low-frequency speaker arrays and signal preprocessing technology, the effect of low-frequency sound is improved. At the same time, by utilizing mid-to-high frequency enhancement technology, parametric equalization, and beamforming principles, the performance of mid-to-high frequency sound is significantly improved, effectively solving problems such as muddy low-frequency sound and harsh mid-to-high frequency sound. 5. The design scheme of this invention is simple, feasible, and easy to implement, and can be applied to various scenarios. Through frequency division processing and optimized design, the system structure is more compact and the cost is lower, expanding the application range and overcoming the limitations of complex directivity optimization methods in existing technologies.

Claims

1. A wide-angle adaptive directivity loudspeaker system, comprising a cuboid enclosure, characterized in that, The top of the enclosure is equipped with a binocular high-definition camera. The enclosure is also connected to a mechanical steering mechanism. Inside the enclosure are a speaker unit array, a frequency division and signal processing unit, a MEMS microphone, and a control unit. The control unit is connected to the binocular high-definition camera, the microphone, and the frequency division and signal processing unit, respectively. The output of the frequency division and signal processing unit is connected to the speaker array unit. The binocular high-definition camera is installed on the top of the speaker to take a picture of the listener's position. The image processing software is used to analyze the two pictures taken by the binocular camera to obtain the listener's angle θ relative to the central axis and the distance relative to the sound source. The mechanical steering mechanism is also connected to the control unit and is located outside the entire enclosure. It is used to adjust the orientation of the entire enclosure, making it suitable for situations where the listener's range of motion exceeds 25°. The loudspeaker unit array consists of eight loudspeaker units, including two low-frequency loudspeaker units located at both ends of the array, employing a coil loudspeaker configuration; and six mid-high frequency loudspeaker units arranged in a ring in the middle, employing a coil-plus-piezoelectric loudspeaker configuration. The piezoelectric configuration is suitable for frequencies above 10kHz. The ring diameter is 400±50mm, and the diaphragm area of ​​the six mid-high frequency loudspeaker units is 150±20mm². 2 The volume of the acoustic cavity is 40±5cm. 3 The sensitivity is 105±5dB; The frequency division and signal processing unit adopts a collaborative processing architecture of a digital signal processor (DSP) and a field-programmable gate array (FPGA). The DSP is configured to perform digital frequency division processing, parametric equalization, and beamforming of the audio signal. The frequency division point is obtained by the center position of the frequency response curves of the mid-high frequency speaker and the low-frequency speaker. The low-frequency signal after frequency division is sent to the two low-frequency speaker units, where the low-frequency signal refers to the signal with a frequency below 500Hz±50Hz. The mid-high frequency signal, i.e., the signal with a frequency above 500Hz±50Hz, is sent to the six mid-high frequency speaker units. The FPGA is connected to the DSP to form a digital directional adjustment for the sound direction of the speakers. It receives parametric equalization and beamforming coefficient instructions from the DSP. Parametric equalization is used to calibrate each speaker, and beamforming is used to adjust the phase and delay of the signal sent to the six mid-high frequency speaker units in real time, thereby achieving directional adjustment in a direction less than 25°. For large-scale applications and situations where the listener moves quickly, mechanical and electronic directional coupling control is necessary to achieve the desired effect. The microphone is a MEMS microphone positioned at the center of the six mid-to-high frequency speaker units inside the speaker cabinet. It is used to sense the direct sound emitted by each speaker unit in a time-division manner, that is, the six speakers emit sound at different times for periodic acoustic calibration. The input of the control unit is connected to an environmental perception unit containing a MEMS microphone and a binocular camera, and the output is connected to a digital signal processor (DSP) in the frequency division and signal processing unit. The control unit receives and fuses the data from the environmental perception unit, dynamically calculates the parametric equalization and beamforming parameters required for each speaker unit in the speaker array based on the fused data, and sends the parameters to the DSP.

2. The wide-angle adaptive directivity loudspeaker system according to claim 1, characterized in that, The shell has 12 through holes with a diameter of 8±1mm. The through holes are distributed on the left, right and rear sides of the shell, with 4 holes on each side. They are used to balance the air pressure inside and outside the box and optimize the low-frequency radiation characteristics.

3. The wide-angle adaptive directivity loudspeaker system according to claim 1, characterized in that, The mechanical steering mechanism includes a speaker connector and a base. A universal rotating unit is mounted on the base, and its upper surface is fixedly connected to the bottom of the cabinet via the speaker connector. The universal rotating unit is cylindrical, and its outer casing has two pitch drive shafts arranged radially opposite each other. The open end of a U-shaped bracket is connected to the two pitch drive shafts, and the closed end of the U-shaped bracket is connected to the output shaft of the first stepper motor. The first stepper motor is fixed to the base via a support plate. The outer casing of the universal rotating unit also has a left-right undulating drive shaft located radially perpendicular to the pitch axis. The left-right undulating drive shaft is connected to the output shaft of the second stepper motor via a V-shaped rotating bracket. The V-shaped apex of the rotating bracket is located directly below the universal rotating unit, and one end of the rotating bracket is connected to the output shaft of the second stepper motor. The second stepper motor is fixed to the base via another support plate, and its output shaft is perpendicular to the first stepper motor's output shaft.

4. An electromechanical coordinated control method for a wide-angle adaptive directivity loudspeaker system as described in any one of claims 1-3, characterized in that, The control unit is configured to perform the following electromechanical coordinated control: when the angle between the listener's position and the normal to the front of the speaker is greater than the first preset threshold, the mechanical steering mechanism of the control unit adjusts the orientation of the speaker enclosure, and simultaneously controls the crossover and signal processing unit to perform electronic beamforming compensation on the speaker unit array; when the angle between the listener's position and the normal to the front of the speaker is reduced to below the second preset threshold, the mechanical steering mechanism is stopped, and electronic beamforming tracking is performed only by the crossover and signal processing unit; the control unit also includes setting a movement speed threshold, when the listener's movement speed is greater than the movement speed threshold, the mechanical steering mechanism and the crossover and signal processing unit work simultaneously, and the crossover and signal processing unit calculates and compensates for the insufficient response of the mechanical steering mechanism to the speaker position adjustment in real time.

5. The electromechanical coordinated control method for a wide-angle adaptive directivity loudspeaker system according to claim 4, characterized in that, When the angle difference between the listener's position and the normal of the speaker's front is ≤25°, the system only uses electronic pointing compensation adjustment. The FPGA performs real-time phase and delay adjustment on the mid-to-high frequency speaker unit signal to achieve fast, silent, and precise micro-tracking, i.e., the first preset threshold is 25°. When the angle difference is greater than 25°, the system first activates the mechanical steering mechanism for coarse positioning of the speaker enclosure, while simultaneously superimposing electronic pointing compensation adjustment to compensate for the delay of mechanical movement and achieve a smooth transition. The goal of mechanical adjustment is to reduce the angle difference to within 5°, i.e., within the second preset threshold. After that, mechanical adjustment is stopped and the electronic system is completely handed over for fine tracking again. When the listener moves quickly, i.e., the movement speed is >1m / s, the mechanical and electronic systems work simultaneously. The electronic system calculates and compensates for the insufficient response of the mechanical system in real time.

6. The electromechanical coordinated control method for a wide-angle adaptive directivity loudspeaker system according to claim 5, characterized in that, Includes the following steps: Step 1: Initialize system configuration and set system operating parameters; Step 2: Environmental perception and data collection: During speaker idle time, the MEMS microphone is used to sense the sound intensity and phase difference of each speaker in a time-division manner, and the binocular high-definition camera is used to detect the angle of the listener relative to the normal of the front of the speaker and the distance relative to the sound source in real time. Step 3: Parametric Equalization Calculation: The Digital Signal Processor (DSP) uses a high-precision parametric equalizer to calibrate the data collected by the microphone in Step 2 across the entire frequency band, from 20Hz to 20kHz. First, the frequency response curve is measured using the microphone. For deviations exceeding ±2dB, boosting or attenuation is performed to compensate for the speaker's own deviation and to solve the problem of uneven and non-uniform frequency response caused by standing waves and resonance. The difference in sound intensity caused by the original sound or time-lapsed sound between different speakers is compensated to make the deviation less than ±1dB. The phase of different speakers is calibrated to make the deviation less than 0.05ms. Step 4: If the angle of change of the listener's position relative to the normal of the front of the speaker is within 25°, then perform electronic directivity compensation adjustment: Perform beamforming calculation, calculate beamforming parameters through DSP based on environmental data: Calculate the time difference required for sound to reach the listener from each speaker unit based on the listener's position parameters, including angle and distance, compensate for the delay of the electrical signals of each speaker unit through delay circuit or digital delay algorithm, calculate the phase difference of each speaker unit based on the listener's position and environmental parameters, and calculate the beamforming parameters required for each speaker unit based on these time and phase differences, so that the sound waves from different speakers at the listener's position are superimposed and enhanced, thereby achieving sound directivity; If the angle is greater than 25°, proceed to step 6; Step 5: Fine-tune the directivity of the loudspeakers: Based on the beamforming parameters, control the loudspeaker array to split the audio signal into low-frequency and mid-high-frequency signals through a crossover. The low-frequency signal is sent to a low-frequency array consisting of two loudspeaker units, and the mid-high-frequency signal is sent to a mid-high-frequency array consisting of six loudspeaker units. Through high-speed parallel processing by FPGA, real-time delay and phase adjustment of the mid-high-frequency signals are achieved. The delay adjustment range is 0.1ms to 2ms, and the phase adjustment range is -220° to 220°. Step 6: If the change in the listener's direction exceeds the adjustment range of 25°, mechanical adjustment is activated first, followed by electronic pointing compensation adjustment. The mechanical adjustment calculates the rotation of the two stepper motor shafts and then drives the stepper motors through the control unit to drive the angle adjustment mechanism to achieve mechanical direction adjustment of the enclosure, thus achieving a wide range of adjustment, which can at least meet the requirements of ±60°. Step 7: Return to step 2 and start the next loop.

7. The electromechanical coordinated control method for a wide-angle adaptive directivity loudspeaker system according to claim 6, characterized in that, In step 6, the goal of mechanical adjustment is to keep the angle within 5°. During the mechanical movement, the binocular camera collects the listener's angle in real time, and the control unit simultaneously determines whether the current listener's position is within the 5° range. If not, the mechanical part continues to rotate for compensation, and the electronic pointing control continues to perform compensation. This cycle repeats until the angle is less than 5°, at which point the mechanical compensation stops. At this point, the angle compensated by the electronic pointing compensation is equal to the angle between the speaker plane normal and the line connecting the speaker and the listener before the mechanical compensation was started - the angle adjusted by the mechanical part.

8. The electromechanical coordinated control method for a wide-angle adaptive directivity loudspeaker system according to claim 7, characterized in that, In step 6, if the listener's movement includes not only large-scale movement but also rapid movement (greater than 1 m / s), and the mechanical response cannot keep up with the listener's rapid movement, electronic pointing must be superimposed simultaneously with the mechanical movement. That is, during mechanical rotation, the mechanical control unit transmits the horizontal (left-right undulation) and vertical tilt angles to the beamforming control unit in real time. The first drive motor generates the tilting motion to adapt to changes in the listener's distance and height (standing or sitting posture); the second drive shaft generates the left-right undulation motion, working in conjunction with the first drive shaft to produce pointing in any direction towards the listener; during electronic pointing compensation adjustment, the beamforming... The beamforming control unit calculates the difference between the angle of mechanical change and the angle of the listener's actual movement obtained by the binocular camera as the compensation angle for electronic pointing. This is then fused with the listener's distance data to recalculate the beamforming parameters and adjust the beamforming in real time, ensuring that the sound is always accurately pointed at the listener. When the mechanical adjustment stops, the electronic pointing compensation operation continues. This can meet the pointing requirements of situations where the listener moves quickly and over a wide area. While the mechanical mechanism moves smoothly to achieve wide coverage, the electronic system responds at high speed to process dynamic details, thus achieving a tracking range and response speed far exceeding the limits of a single technology without sacrificing sound quality.

9. The electromechanical coordinated control method for a wide-angle adaptive directivity loudspeaker system according to claim 8, characterized in that, Step 4 is achieved through the following measures: Step 4-1: Use DSP to calculate beamforming. Calculate beamforming parameters based on environmental data and calculate the time difference required for sound to reach the listener from each speaker unit based on the listener's position. The velocity constant c0 = 340 m / s. Assuming the distance between two adjacent speaker units is d, and the target is to point the main sound beam in a direction that forms an angle θ with the array normal, then the path length the sound travels from these two units to the listener's position will have a difference, i.e., a path difference ΔL = d * sin(θ). To make these two sound waves superimpose in phase in the θ direction, i.e., enhance each other, this path difference must be compensated. The wavelength of the sound wave λ = the speed of sound c / frequency f. The time difference corresponding to this path difference ΔL is Δt = ΔL / c. Step 4-2: The FPGA needs to send a signal Δt time in advance for the unit farther from the listener, and perform delay compensation on the electrical signals of each speaker unit through a digital delay algorithm; Step 4-3: Make the phase of the signal of the speaker unit farther from the listener lead by a specific angle, i.e., the phase difference. The conversion relationship between them is: phase difference φ=(ΔL / λ)*360°. Calculate the phase difference of each speaker unit according to the listener's position and environmental parameters; Step 4-4: Based on the time difference and phase difference, calculate the beamforming parameters required for each speaker unit, and obtain the phase compensation value of each unit.

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