A professional digital sound

Abstract

The application discloses a kind of professional digital sound, it is related to digital audio technology field, space detection module, for emitting fan-shaped detection beam to the front of sound, real-time acquisition and identification target object quantity and physical position information in detection area;Space modeling module, for constructing the virtual space coordinate system containing hierarchical grid unit.The application acquires the quantity of target object in front and physical position information, constructs the virtual space coordinate system containing hierarchical grid unit and virtual sound source model, maps target object as virtual signal point and completes grid positioning;Then by analysis relying on grid attribute setting limit condition, distribution and acquisition each virtual signal point cumulative value, finally through sound field adaptive adjustment module carries out weighted operation, solves angle deviation and deflection parameter, generates control instruction and drives physical sound adaptive deflection play range, realizes the dynamic adaptive adjustment of sound field.

Description

Technical Field

[0001] This invention relates to the field of digital audio technology, specifically to a professional digital audio system. Background Technology

[0002] Digital audio systems are professional audio devices that integrate signal processing, acoustic playback, and intelligent control. They rely on built-in detection and processing units to decode, amplify, and directionally play audio signals, and are widely used in conference rooms, family living rooms, and professional venues. Digital audio systems can radiate directional sound fields through loudspeakers, providing auditory audio services to people in the space, and are a core component of modern spatial acoustic playback systems.

[0003] However, traditional digital audio systems are inadequate in terms of sound field adaptability. Most existing devices can only provide a fixed playback range or preset sound field modes, and cannot dynamically adjust the directivity of the speakers according to the actual distribution of people in the venue. When the audience is concentrated in a certain area, the speakers may still radiate sound energy evenly into open areas, resulting in wasted sound energy, insufficient local loudness, or unnecessary reflection interference, thus reducing the listening experience. Summary of the Invention

[0004] The purpose of this invention is to provide a professional digital audio system to address the shortcomings of the prior art.

[0005] To achieve the above objectives, the present invention provides the following technical solution: a professional digital audio system, comprising: The spatial detection module is used to emit a fan-shaped detection beam in front of the speaker to collect and identify the number and physical location information of target objects in the detection area in real time. The spatial modeling module is used to construct a virtual spatial coordinate system containing hierarchical grid cells, and to establish a virtual sound source model corresponding to the physical sound playback range. It maps the target object to a virtual signal point in the virtual space and positions the virtual signal point to the corresponding hierarchical grid cell. The parameter analysis module is used to establish a data transmission link between virtual signal points and virtual sound source models, and to impose constraints on virtual signal points based on the properties of hierarchical grid cells. It assigns a cumulative value to each virtual signal point according to the number of constraints and obtains the cumulative value through the data transmission link. The sound field adaptive adjustment module is used to calculate the skew parameters of the virtual sound source model based on the cumulative value of the virtual signal points, and generate control commands based on the skew parameters to drive the physical sound system to adjust the playback range of the loudspeakers.

[0006] In a preferred embodiment, the space detection module includes: The radar unit is used to emit a fan-shaped detection beam in front of the sound source, receive the echo signal reflected by the target object in the detection area, and obtain the target object's distance, azimuth angle and radial velocity information. The signal processing subunit is used to process the echo signal and count the number of target objects within the detection area.

[0007] In a preferred embodiment, the spatial modeling module includes: Virtual space construction unit is used to establish a three-dimensional coordinate system with the sound center of the physical sound source as the origin, and divide the three-dimensional coordinate system into several hierarchical grid units of the same size; The virtual sound source modeling unit is used to obtain the playback range of the physical sound source and construct a virtual sound source model representing the acoustic playback range of the physical sound source in a three-dimensional coordinate system. The object mapping unit is used to convert the physical location information of the target object into coordinate data in a three-dimensional coordinate system, generate corresponding virtual signal points, and assign the virtual signal points to the corresponding hierarchical grid units according to the coordinate data.

[0008] In a preferred embodiment, the parameter analysis module includes: The link establishment unit is used to establish a data transmission link between the virtual signal point and the center point of the virtual sound source model. The data transmission link serves as the transmission path for the acoustic signal. The constraint condition determination unit is used to add grid attributes to the hierarchical grid cell where the virtual signal point is located. The grid attributes include the distance threshold between the virtual signal point and the virtual sound source model, the importance level of the personnel, and the intensity of environmental interference. The numerical allocation unit is used to count the grid attributes of each virtual signal point, allocate corresponding incremental values ​​according to the number of grid attributes, and accumulate the incremental values ​​to obtain the cumulative value. The data acquisition unit is used to read the cumulative value corresponding to each virtual signal point in real time through the data transmission link.

[0009] In a preferred embodiment, the sound field adaptive adjustment module includes: The weighted calculation unit is used to perform weighted calculations on the cumulative values ​​of all virtual signal points to determine the coordinates of the weighted center point of the virtual sound source model; The skew parameter calculation unit is used to calculate the angular deviation between the weighted center point coordinates and the physical sound source center axis, and convert the angular deviation into the skew parameters of the virtual sound source model. The instruction generation unit is used to generate control instructions based on the skew parameters. The drive execution unit is used to adjust the virtual sound source model in the three-dimensional coordinate system by using control commands as drive signals, so that the playback range of the physical sound is deflected toward the weighted center point coordinates.

[0010] In a preferred embodiment, it further includes a speaker housing, a speaker, and a motor fixedly mounted at the bottom of the speaker housing cavity, characterized in that: A mounting bracket is fixedly installed inside the speaker housing. An L-shaped rotating shaft is rotatably provided in the middle of the mounting bracket. A connecting component for connecting the speaker is installed at the upper end of the rotating shaft. The lower end of the rotating shaft is fixedly connected to the output end of a motor. The motor is used to receive control commands from the drive execution unit.

[0011] In a preferred embodiment, the connecting assembly includes a connecting ring fixedly mounted on the upper end of the rotating shaft and a fixing ring fixedly mounted on the speaker. A plurality of straight rods are equidistantly mounted on the fixing ring. The straight rods are inserted into the connecting ring, and a nut is threaded at the threaded end of each straight rod. A buffer is provided on each straight rod.

[0012] In a preferred embodiment, the buffer includes two buffer rings, and a retaining ring is fixedly installed on the straight rod, with the two buffer rings located on both sides of the connecting ring.

[0013] The technical effects and advantages provided by the present invention in the above technical solution are as follows: This invention collects information on the number and physical location of target objects in front, constructs a virtual spatial coordinate system and a virtual sound source model with hierarchical grid units, maps target objects to virtual signal points, and completes grid positioning. Then, based on grid attributes, it sets constraints, allocates and obtains the cumulative values ​​of each virtual signal point. Finally, through a sound field adaptive adjustment module, it performs weighted calculations, solves for angle deviation and skew parameters, and generates control commands to drive the physical speakers to adaptively deflect the playback range, achieving dynamic adaptive adjustment of the sound field. The speakers can sense the distribution of people in real time, accurately calculate the sound source deflection angle using hierarchical grids and weighted algorithms, and drive the loudspeakers to directionally focus audio energy on the target area, significantly improving sound reinforcement efficiency and auditory experience. Attached Figure Description

[0014] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the drawings used in the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments recorded in this invention. For those skilled in the art, other drawings can be obtained based on these drawings.

[0015] Figure 1 This is a system flowchart of the present invention.

[0016] Figure 2 This is a logic block diagram of the present invention.

[0017] Figure 3 This is a schematic diagram of the speaker housing structure of the present invention.

[0018] Figure 4 This is a cross-sectional view of the speaker housing of the present invention.

[0019] Figure 5 This is a schematic diagram of the connection component structure of the present invention.

[0020] The numbers in the diagram are: 1. Speaker housing; 11. Speaker; 12. Motor; 13. Mounting bracket; 14. Shaft; 2. Connecting assembly; 21. Fixing ring; 22. Straight rod; 23. Connecting ring; 24. Nut; 3. Retaining ring; 31. Buffer ring. Detailed Implementation

[0021] To make the objectives, technical solutions, and advantages of the embodiments of the present invention clearer, the technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0022] Example 1, please refer to Figure 1 As shown in this embodiment, a professional digital audio system includes: The spatial detection module is used to emit a fan-shaped detection beam in front of the speaker to collect and identify the number and physical location information of target objects in the detection area in real time. The spatial modeling module is used to construct a virtual spatial coordinate system containing hierarchical grid cells, and to establish a virtual sound source model corresponding to the physical sound playback range. It maps the target object to a virtual signal point in the virtual space and positions the virtual signal point to the corresponding hierarchical grid cell. The parameter analysis module is used to establish a data transmission link between virtual signal points and virtual sound source models, and to impose constraints on virtual signal points based on the properties of hierarchical grid cells. It assigns a cumulative value to each virtual signal point according to the number of constraints and obtains the cumulative value through the data transmission link. The sound field adaptive adjustment module is used to calculate the skew parameters of the virtual sound source model based on the cumulative value of the virtual signal points, and generate control commands based on the skew parameters to drive the physical sound system to adjust the playback range of the loudspeakers.

[0023] Furthermore, the spatial detection module uses a fan-shaped beam to scan the area in front, capturing the physical coordinates of human targets. Next, the spatial modeling module maps these coordinates onto a hierarchical grid in the virtual space, forming a visualized signal point distribution map. The parameter analysis module evaluates the grid level and distance constraints of each signal point, assigning each signal point a cumulative value representing its importance. Finally, the sound field adaptive adjustment module analyzes the centroid of these values, automatically calculating the angle parameters that the virtual sound source needs to deflect, generating control commands to drive the physical speakers to adjust the playback direction, thereby achieving the technical effect of automatically concentrating sound energy towards densely populated areas or specific level areas.

[0024] In one embodiment, the space exploration module includes: The radar unit is used to emit a fan-shaped detection beam in front of the sound source, receive the echo signal reflected by the target object in the detection area, and obtain the target object's distance, azimuth angle and radial velocity information. The signal processing subunit is used to process the echo signal and count the number of target objects within the detection area.

[0025] Furthermore, the radar unit activates, emitting a fan-shaped detection beam in front of the speaker according to preset parameters. The beam covers the speaker's preset playback detection range, ensuring comprehensive capture of target objects within the range. When the detection beam contacts a target object within the detection area, it is reflected by the target, forming an echo signal. The radar unit receives this echo signal in real time and, through signal analysis, simultaneously acquires the target object's distance (the actual distance between the target and the speaker), azimuth angle (the target's horizontal orientation relative to the speaker), and radial velocity (the target's speed relative to the speaker), completing the acquisition of basic target information. Subsequently, the radar unit synchronously transmits the acquired raw echo signal and extracted basic target information to the signal processing subunit. The signal processing subunit optimizes the raw echo signal, including filtering, noise reduction, signal amplification, and analysis, removing environmental interference, clutter, and other invalid signals, retaining valid signals related to the target object. Simultaneously, based on the analyzed valid signals, it counts the specific number of target objects within the detection area, clarifying the target distribution scale.

[0026] In one embodiment, the spatial modeling module includes: Virtual space construction unit is used to establish a three-dimensional coordinate system with the sound center of the physical sound source as the origin, and divide the three-dimensional coordinate system into several hierarchical grid units of the same size; The virtual sound source modeling unit is used to obtain the playback range of the physical sound source and construct a virtual sound source model representing the acoustic playback range of the physical sound source in a three-dimensional coordinate system. The object mapping unit is used to convert the physical location information of the target object into coordinate data in a three-dimensional coordinate system, generate corresponding virtual signal points, and assign the virtual signal points to the corresponding hierarchical grid units according to the coordinate data.

[0027] Furthermore, the virtual space construction unit establishes a three-dimensional coordinate system with the sound center of the physical speaker as the origin, consistent with the scale of the actual physical space. The coordinate axes of the three-dimensional coordinate system are synchronized with the physical space to ensure that the virtual coordinates accurately correspond to the physical location. Subsequently, according to preset specifications, the three-dimensional coordinate system is evenly divided into several hierarchical grid units of the same size. The specifications of the grid units are set according to the speaker's playback range and detection accuracy requirements. Then, the virtual sound source modeling unit works synchronously, first acquiring the actual playback range parameters of the physical speaker (including playback angle, coverage distance, acoustic radiation characteristics, etc.), and then constructing a virtual sound source model in the three-dimensional coordinate system established by the virtual space construction unit based on the actual playback range parameters of the physical speaker. The acoustic playback range and radiation characteristics of the virtual sound source model are perfectly matched with the physical speaker, achieving accurate mapping of the physical speaker in the virtual space. Finally, the object mapping unit receives the target physical location information transmitted by the space detection module. Through a coordinate transformation algorithm, it converts the physical location (distance, azimuth, etc.) of each target object into specific coordinate data in a three-dimensional coordinate system. Each target object generates a unique virtual signal point, and the coordinates of the virtual signal point correspond one-to-one with the physical location of the target object. Subsequently, the object mapping unit determines the hierarchical grid cell to which the virtual signal point belongs based on its coordinate data, and accurately assigns the virtual signal point to the corresponding grid cell, thus completing the conversion and positioning of the physical target into a virtual signal point.

[0028] In one embodiment, the parameter analysis module includes: The link establishment unit is used to establish a data transmission link between the virtual signal point and the center point of the virtual sound source model (to build a stable information transmission channel between the virtual target node and the virtual sound source center, providing path support for parameter transmission and data interaction between the two, and ensuring the real-time performance and accuracy of data transmission). The data transmission link serves as the transmission path for acoustic signals. The constraint condition determination unit is used to add grid attributes to the hierarchical grid cell where the virtual signal point is located. The grid attributes include the distance threshold between the virtual signal point and the virtual sound source model, the importance level of the personnel, and the intensity of environmental interference. The numerical allocation unit is used to statistically analyze the grid attributes of each virtual signal point, allocate corresponding incremental values ​​according to the number of grid attributes, and accumulate the incremental values ​​to obtain a cumulative value (allocate corresponding incremental values ​​for different grid attribute value ranges, such as larger increments for closer distances, longer stays, and higher densities; then arithmetically sum all the incremental values ​​corresponding to the virtual signal point to generate a cumulative value that comprehensively reflects the importance of the target, thereby quantifying the urgency of the target's need for sound field control). The data acquisition unit is used to read the cumulative value corresponding to each virtual signal point in real time through the data transmission link.

[0029] Furthermore, the link establishment unit first establishes an independent data transmission link between each virtual signal point and the center point of the virtual sound source model. The data transmission link serves as a dedicated transmission path for acoustic signals and quantization data, ensuring that the relevant parameters of the virtual signal point can be transmitted to the virtual sound source model in real time and accurately. Subsequently, the constraint determination unit supplements the attributes of the hierarchical grid cell containing each virtual signal point, adding three core grid attributes to the hierarchical grid cell: First, the distance threshold between the virtual signal point and the virtual sound source model, used to define the proximity relationship between the target and the sound source. Only when the distance between the virtual signal point and the virtual sound source model is less than the distance threshold corresponding to the grid cell will the virtual signal point be determined as a "valid target" and participate in the subsequent weight calculation; otherwise, it will be regarded as an invalid or low-priority target, and its influence will be greatly weakened or even ignored. Second, the importance level of the personnel. Based on the coordinates of the virtual signal point in the hierarchical grid cell, the degree of deviation from the central axis of the virtual sound source model is determined. The closer the target is to the central axis (i.e., directly facing the speaker), the higher its priority (for example, targets located in the front center area are marked as "high priority," and targets located at the edge corners are marked as "low priority"). Third, the environmental interference intensity. Based on the preset or detected echo characteristics, if an obstacle is detected in the hierarchical grid cell, the weight of the virtual signal point in the hierarchical grid cell with the highest environmental interference intensity will be reduced. These grid attributes together constitute the constraint conditions for the virtual signal points, providing a clear basis for subsequent numerical allocation. Next, the numerical allocation unit begins its work, counting the number of grid attributes corresponding to each virtual signal point one by one. According to preset rules (for example, virtual signal points covered by the sound propagation range of the virtual sound source model are assigned values ​​as "effective targets", and those outside the sound propagation range of the virtual sound source model are not assigned values), the distance threshold between the virtual signal points and the virtual sound source model is then used. Starting from the virtual sound source model, the hierarchical grid units are horizontally divided into three ranges based on their distance from the virtual sound source model. The range closest to the virtual sound source model is the first range, and so on. Virtual signal points in the first range are assigned a value of 0.8, the second range is assigned a value of 1, and the third range is assigned a value of 0.6.The importance level of personnel is determined by dividing the hierarchical grid unit vertically into three ranges along the central axis of the virtual sound source model. The range closest to the virtual sound source model is the first range, and so on. The virtual signal points in the first range are assigned a value of 1, the second range is assigned a value of 0.8, and the third range is assigned a value of 0.6. Finally, the environmental interference intensity is divided into three ranges, such as when the person is 80% blocked by an obstacle, half blocked, or not blocked. The first range, where the person is not blocked, is assigned a value of 1, the second range, where half is blocked, is assigned a value of 0.8, and the third range, where 80% is blocked, is assigned a value of 0.6. After summing the numbers within the ranges of the distance threshold attribute and the personnel importance level attribute of the virtual signal point and virtual sound source model, and then subtracting the numbers within the three ranges of environmental interference intensity, a corresponding incremental value is assigned to each grid attribute (e.g., a fixed increment is assigned for each grid attribute). Then, all the incremental values ​​of each virtual signal point are accumulated to obtain the cumulative value of the virtual signal point, which directly corresponds to the priority of the virtual signal point (i.e., the physical target)'s sound field adjustment requirement. The larger the cumulative value, the more prominent and higher the priority of the target's sound field requirement. Finally, the cumulative value corresponding to each virtual signal point is read in real time through the transmission link.

[0030] In one embodiment, the sound field adaptive adjustment module includes: The weighted calculation unit is used to perform weighted calculations on the cumulative values ​​of all virtual signal points to determine the coordinates of the weighted center point of the virtual sound source model; The skew parameter calculation unit is used to calculate the angular deviation between the weighted center point coordinates and the physical sound source center axis, and convert the angular deviation into the skew parameters of the virtual sound source model. The instruction generation unit is used to generate control instructions based on the skew parameters (converting the quantized skew parameters into control signals that can be recognized and executed by the physical audio system). The drive execution unit is used to adjust the virtual sound source model in the three-dimensional coordinate system by using control commands as drive signals, so that the playback range of the physical sound is deflected toward the weighted center point coordinates.

[0031] Furthermore, the weighted calculation method involves using the cumulative value corresponding to each virtual signal point as the weight, and then taking a weighted average of the coordinates of all virtual signal points according to their respective weights. Finally, the coordinates of the weighted center point with the strongest overall requirement are calculated and used as the target direction for sound field deflection. The calculation formula is: There are n virtual signal points in total, and the spatial coordinates of the i-th virtual signal point are: The weight corresponding to the cumulative value of the i-th virtual signal point: Weighted center point coordinates of virtual sound source model .

[0032]

[0033] The method for determining the angular deviation is as follows: coordinates of the reference point of the physical sound center axis. axial vector with the central axis of the sound system as the reference. The weighted center point relative to the reference point vector: The angle between the vectors is the angular deviation. : ; Where the dot product is: ; Length of the module: , .

[0034] The cumulative value of each virtual signal point is used as a weighting factor (the larger the cumulative value, the higher the weight, and the higher the priority of the sound field requirement of the corresponding physical target). Combined with the coordinate data of each virtual signal point in the three-dimensional coordinate system, a preset weighting algorithm is used to calculate the weighted center point coordinates of all virtual signal points. The weighted center point coordinates accurately represent the concentrated area of ​​sound field requirements of all physical targets and are the core target for sound field deflection, ensuring that sound field adjustment can prioritize covering the area with the most concentrated requirements. Subsequently, the skew parameter calculation unit intervenes, first obtaining the center axis parameters of the physical speaker (i.e., the reference direction axis of the physical speaker's default playback), and then calculating the angular deviation between the weighted center point coordinates obtained by the weighting calculation unit and the center axis. Through a preset parameter conversion algorithm, the angular deviation is converted into the skew parameters of the virtual sound source model. The skew parameters clarify the angle and direction that the virtual sound source model needs to be deflected, realizing the quantitative conversion from "angle deviation" to "adjustment parameters" and providing a precise basis for the generation of subsequent control commands. Next, the instruction generation unit generates corresponding control instructions based on the skew parameters output by the skew parameter calculation unit, according to the control protocol of the physical speaker. These control instructions include core execution parameters such as skew angle, deflection direction, and adjustment speed, which can be directly recognized by the physical speaker's drive unit, bridging the gap between virtual parameters and physical execution, ensuring that adjustment parameters are effectively translated into action. Finally, the drive execution unit receives the control instructions output by the instruction generation unit and uses them as drive signals to synchronously act on the virtual sound source model and the physical speaker in the three-dimensional coordinate system. On one hand, it adjusts the virtual sound source model's posture in the three-dimensional coordinate system according to the control instructions, causing its playback range to deflect towards the weighted center point coordinates, maintaining synchronization between the virtual and physical spaces. On the other hand, it drives the motor in the physical speaker that rotates the speaker, adjusting the playback angle according to the control instructions, so that the actual sound field playback range of the physical speaker accurately points to the physical area corresponding to the weighted center point coordinates.

[0035] In one embodiment, the system further includes a speaker housing 1, a speaker 11, and a motor 12 fixedly mounted at the bottom of the inner cavity of the speaker housing 1. A mounting bracket 13 is fixedly installed inside the speaker housing 1. An L-shaped rotating shaft 14 is rotatably provided in the middle of the mounting bracket 13. A connecting component 2 for connecting the speaker 11 is installed on the upper end of the rotating shaft 14. The lower end of the rotating shaft 14 is fixedly connected to the output end of the motor 12. The motor 12 is used to receive control commands sent from the drive execution unit.

[0036] Furthermore, the mounting bracket 13 is equipped with a bearing, and the rotating shaft 14 is installed in the inner ring of the bearing. The rotating shaft 14 is in a vertical position in the speaker housing 1. After the control command is sent to the motor 12, the motor 12 drives the rotating shaft 14 to rotate. The rotating shaft 14 drives the connecting component 2, allowing the speaker 11 to rotate in the horizontal direction, thereby achieving angle adjustment.

[0037] In one embodiment, the connecting assembly 2 includes a connecting ring 23 fixedly mounted on the upper end of the rotating shaft 14 and a fixing ring 21 fixedly mounted on the speaker 11. A plurality of straight rods 22 are equidistantly mounted on the fixing ring 21. The straight rods 22 are inserted into the connecting ring 23. Nuts 24 are threaded at the threaded ends of the straight rods 22. A buffer is provided on the straight rods 22.

[0038] Furthermore, the straight rod 22 is first inserted into the connecting ring 23, and the nut 24 is installed at the threaded end of the straight rod 22 to fix the connecting ring 23, thereby reducing the vibration of the speaker 11 through the buffer.

[0039] In one embodiment, the buffer includes two buffer rings 31, and a retaining ring 3 is fixedly installed on the straight rod 22. The two buffer rings 31 are located on both sides of the connecting ring 23. Furthermore, the buffer ring 31 is made of rubber, with one located between the retaining ring 3 and one side of the buffer ring 31, and the other located between the nut 24 and the other side of the buffer ring 31. The rubber buffer ring 31 absorbs the vibration of the speaker 11.

[0040] The above description is merely a specific embodiment of this application, but the scope of protection of this application is not limited thereto. Any variations or substitutions that can be easily conceived by those skilled in the art within the scope of the technology disclosed in this application should be included within the scope of protection of this application. Therefore, the scope of protection of this application should be determined by the scope of the claims.

Claims

1. A professional digital audio system, characterized in that, include: The spatial detection module is used to emit a fan-shaped detection beam in front of the speaker to collect and identify the number and physical location information of target objects in the detection area in real time. The spatial modeling module is used to construct a virtual spatial coordinate system containing hierarchical grid cells, and to establish a virtual sound source model corresponding to the physical sound playback range. It maps the target object to a virtual signal point in the virtual space and positions the virtual signal point to the corresponding hierarchical grid cell. The parameter analysis module is used to establish a data transmission link between virtual signal points and virtual sound source models, and to impose constraints on virtual signal points based on the properties of hierarchical grid cells. It assigns a cumulative value to each virtual signal point according to the number of constraints and obtains the cumulative value through the data transmission link. The sound field adaptive adjustment module is used to calculate the skew parameters of the virtual sound source model based on the cumulative value of the virtual signal points, and generate control commands based on the skew parameters to drive the physical sound system to adjust the playback range of the loudspeakers.

2. A professional digital audio system according to claim 1, characterized in that, The space detection module includes: The radar unit is used to emit a fan-shaped detection beam in front of the sound source, receive the echo signal reflected by the target object in the detection area, and obtain the target object's distance, azimuth angle and radial velocity information. The signal processing subunit is used to process the echo signal and count the number of target objects within the detection area.

3. A professional digital audio system according to claim 1, characterized in that, The spatial modeling module includes: Virtual space construction unit is used to establish a three-dimensional coordinate system with the sound center of the physical sound source as the origin, and divide the three-dimensional coordinate system into several hierarchical grid units of the same size; The virtual sound source modeling unit is used to obtain the playback range of the physical sound source and construct a virtual sound source model representing the acoustic playback range of the physical sound source in a three-dimensional coordinate system. The object mapping unit is used to convert the physical location information of the target object into coordinate data in a three-dimensional coordinate system, generate corresponding virtual signal points, and assign the virtual signal points to the corresponding hierarchical grid units according to the coordinate data.

4. A professional digital audio system according to claim 1, characterized in that, The parameter analysis module includes: The link establishment unit is used to establish a data transmission link between the virtual signal point and the center point of the virtual sound source model. The data transmission link serves as the transmission path for the acoustic signal. The constraint condition determination unit is used to add grid attributes to the hierarchical grid cell where the virtual signal point is located. The grid attributes include the distance threshold between the virtual signal point and the virtual sound source model, the importance level of the personnel, and the intensity of environmental interference. The numerical allocation unit is used to count the grid attributes of each virtual signal point, allocate corresponding incremental values ​​according to the number of grid attributes, and accumulate the incremental values ​​to obtain the cumulative value. The data acquisition unit is used to read the cumulative value corresponding to each virtual signal point in real time through the data transmission link.

5. A professional digital audio system according to claim 1, characterized in that, The sound field adaptive adjustment module includes: The weighted calculation unit is used to perform weighted calculations on the cumulative values ​​of all virtual signal points to determine the coordinates of the weighted center point of the virtual sound source model; The skew parameter calculation unit is used to calculate the angular deviation between the weighted center point coordinates and the physical sound source center axis, and convert the angular deviation into the skew parameters of the virtual sound source model. The instruction generation unit is used to generate control instructions based on the skew parameters. The drive execution unit is used to adjust the virtual sound source model in the three-dimensional coordinate system by using control commands as drive signals, so that the playback range of the physical sound is deflected toward the weighted center point coordinates.

6. A professional digital audio system according to claim 5, further comprising an audio housing 1, a speaker 11, and a motor 12 fixedly installed at the bottom of the inner cavity of the audio housing 1, characterized in that: A mounting bracket 13 is fixedly installed inside the speaker housing 1. An L-shaped rotating shaft 14 is rotatably provided in the middle of the mounting bracket 13. A connecting component 2 for connecting the speaker 11 is installed on the upper end of the rotating shaft 14. The lower end of the rotating shaft 14 is fixedly connected to the output end of the motor 12. The motor 12 is used to receive control commands sent from the drive execution unit.

7. A professional digital audio system according to claim 6, characterized in that: The connecting assembly 2 includes a connecting ring 23 fixedly installed on the upper end of the rotating shaft 14 and a fixing ring 21 fixedly installed on the speaker 11. A plurality of straight rods 22 are equidistantly installed on the fixing ring 21. The straight rods 22 are inserted into the connecting ring 23. Nuts 24 are threaded at the threaded ends of the straight rods 22. A buffer is provided on the straight rods 22.

8. A professional digital audio system according to claim 7, characterized in that: The buffer component includes two buffer rings 31, and a retaining ring 3 is fixedly installed on the straight rod 22. The two buffer rings 31 are located on both sides of the connecting ring 23.

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