Sound control method and system

Abstract

The invention relates to the technical field of audio control, in particular to a sound equipment control method, which comprises the following steps: acquiring comprehensive condition data of a sound equipment use environment in real time by integrating various high-precision sensors, performing identification based on the comprehensive condition data, and determining a corresponding environment area, the comprehensive condition data comprises temperature, humidity, wind speed and crowd density, and the environment area comprises a sunshade area, a direct sunlight area and an air duct area; adjusting a working mode and an audio output parameter of the sound box based on the environment area, and applying the working mode and the audio output parameter to a sound box use stage; in the sound equipment using process, a corresponding expected sound field distribution diagram is generated based on a current corresponding sound equipment arrangement scheme, the expected sound field distribution diagram is sent to a user, corresponding user feedback data is obtained, and the corresponding sound equipment arrangement scheme is adjusted based on the user feedback data. According to the invention, the outdoor sound experience of a user can be improved.

Description

Technical Field

[0001] The present application relates to the technical field of audio control, and in particular, to an audio control method and system. Background Art

[0002] Most of the current audio devices on the market mainly focus on indoor usage environments and have limited support for outdoor activities. With the change of people's lifestyle, more and more people hope to enjoy high-quality music experiences during outdoor activities, such as playing background music while camping, hiking, or relaxing in the park. However, existing audio devices often lack effective countermeasures for outdoor environmental factors (such as wind noise and changes in crowd density), resulting in a poor user experience.

[0003] As can be seen from the above, how to improve the audio experience of users outdoors remains to be solved. Summary of the Invention

[0004] In order to improve the audio experience of users outdoors, the present application provides an audio control method and system.

[0005] In a first aspect, the present application provides an audio control method, adopting the following technical solution: An audio control method includes: obtaining comprehensive condition data of the audio usage environment in real time by integrating multiple high-precision sensors, where the comprehensive condition data includes temperature, humidity, wind speed, and crowd density; determining a corresponding environmental area based on the comprehensive condition data, where the environmental area includes a shaded area, a direct sunlight area, and a wind duct area; determining it as the shaded area when the temperature is lower than a first preset threshold, the wind speed is lower than a second preset threshold, and the crowd density is lower than a third preset threshold; determining it as the direct sunlight area when the temperature is higher than a fourth preset threshold, the wind speed is lower than a fifth preset threshold, and the crowd density is higher than a sixth preset threshold; determining it as the wind duct area when the wind speed is higher than a seventh preset threshold and the temperature difference in the surrounding area exceeds an eighth preset threshold; adjusting the working mode and audio output parameters of the audio according to the environmental area, and applying the working mode and audio output parameters during the audio usage stage; during the audio usage process, generating a corresponding expected sound field distribution map based on the current corresponding audio layout plan, sending the expected sound field distribution map to the user, obtaining corresponding user feedback data, and adjusting the corresponding audio layout plan based on the user feedback data; monitoring the change of the comprehensive condition data in real time, and dynamically updating the division of the environmental area and the audio parameter settings according to the real-time data.

[0006] By adopting the above technical solution, temperature, humidity, wind speed and crowd density data are collected in real time through integrated high-precision sensors. Outdoor environment areas (such as shaded areas, direct sunlight areas, air duct areas) are dynamically divided in combination with preset thresholds, and parameters such as the volume, sound effect, and wind protection mode of the sound system are automatically adjusted according to the characteristics of the areas. At the same time, the parameter settings are dynamically updated by real-time monitoring of environmental changes, and the sound system layout is further optimized in combination with the user's feedback on the expected sound field distribution map, so as to accurately adapt the sound effect in different outdoor scenarios (such as reducing wind noise, enhancing bass, adjusting volume distribution), ensuring that the sound clarity and coverage always match the environmental and crowd needs, significantly improving the auditory comfort, anti-interference ability and sound field uniformity in outdoor activities, and realizing an immersive sound effect experience.

[0007] Optionally, the method further includes: real-time monitoring of the comprehensive condition data of the usage environment, evaluating the situation characteristics corresponding to the current usage environment based on the comprehensive condition data; judging whether it is in an emergency based on the situation characteristics, if so, determining the corresponding severity of the situation, and determining the corresponding outdoor situation based on the situation characteristics; adjusting the emergency level and propagation range of the alarm based on the severity of the situation and the outdoor situation.

[0008] By adopting the above technical solution, by real-time monitoring the comprehensive condition data of the usage environment, the system can evaluate the current situation characteristics and intelligently judge whether it is in an emergency. When an emergency is detected, the system will determine the severity and the corresponding outdoor situation according to the specific situation, and then automatically adjust the emergency level and propagation range of the alarm. This method ensures that in case of an emergency, information can be quickly and accurately conveyed to the people in need, enhancing the safety guarantee of users, while minimizing unnecessary panic and interference. This intelligent situation awareness and response mechanism greatly improves the safety and organizational efficiency of outdoor activities.

[0009] Optionally, when multiple sound nodes play audio simultaneously, the method further includes: continuously obtaining the playback status of each sound node based on the built-in real-time audio stream monitoring module, where the playback status includes the transmission delay, processing time and playback progress of the audio stream; determining the time difference and phase difference corresponding to each sound node through a delay compensation algorithm based on the playback status; adjusting the playback start time and audio frame delay of each sound node based on the time difference and the phase difference.

[0010] By adopting the above technical solution, the built-in real-time audio stream monitoring module continuously tracks the playing status of each audio node, and uses the delay compensation algorithm to accurately calculate the time difference and phase difference between each node. Based on these differences, the system dynamically adjusts the playing start time and audio frame delay of each audio node to ensure perfect synchronization when multiple audio nodes play simultaneously. This method effectively eliminates the audio out-of-sync problem caused by network transmission or hardware processing, provides a seamless and consistent sound experience, and is especially suitable for large-scale outdoor activities or multi-area coverage scenarios, significantly improving the stability of the overall audio system and the user experience.

[0011] Optionally, the method further includes: real-time monitoring of the audio status of each audio node, where the audio status includes the online status, location information, and the number of audio nodes; when a change in the audio status is detected, obtaining the corresponding latest audio node layout; starting the sound field modeling algorithm based on the latest audio node layout, recalculating the sound field distribution based on the latest audio node layout, and adjusting the audio output parameters of each audio node based on the sound field distribution.

[0012] By adopting the above technical solution, by real-time monitoring the online status, location information, and the number of nodes of each audio node, the change in the audio status is automatically sensed. Once a change is detected, the system immediately obtains the latest audio node layout and starts the sound field modeling algorithm to recalculate the sound field distribution. Based on the new sound field distribution, the system intelligently adjusts the audio output parameters of each audio node to ensure that the sound coverage is always uniform and clear. This method enables the audio system to quickly adapt to node changes, maintain a consistent auditory effect, improves the flexibility and intelligence level of the audio in outdoor activities, and ensures that users can enjoy the best sound experience after any layout adjustment.

[0013] Optionally, the method further includes: setting a corresponding sound focus position in the actual environment; calculating based on the sound focus position through the acoustic modeling algorithm to determine the corresponding audio output parameters, where the audio output parameters include volume, equalizer settings, and phase difference; when the user adjusts or adds a new sound focus position, recalculating and optimizing the audio output parameters of the audio network and generating a new sound field distribution map to be displayed to the user in real time.

[0014] By adopting the above technical solution, the sound focus position is set in the actual environment, and the optimal audio output parameters (such as volume, equalizer settings, phase difference) are calculated through an acoustic modeling algorithm to ensure that the sound is concentrated and evenly covers each focus. When the user adjusts or adds a new sound focus, the system will immediately recalculate and optimize the audio output parameters of the sound network, and generate a new sound field distribution map to be displayed to the user in real time. This method not only provides a highly personalized sound configuration, but also enhances the user's control and sense of participation, ensuring that each adjustment takes effect immediately, bringing a precise and consistent sound experience to the listeners in a specific area, and greatly improving the sound effect and user satisfaction of outdoor activities.

[0015] Optionally, the method further includes: obtaining the load condition of each sound node, and retrieving the corresponding comprehensive condition data, where the load condition includes CPU usage rate, memory occupancy, and network bandwidth; determining the priority corresponding to each sound node based on the load condition and the comprehensive condition data; when it is detected that there is a change in the load condition or comprehensive condition data of a sound node, adjusting the audio stream parameters of the corresponding sound node based on the priority.

[0016] By adopting the above technical solution, by monitoring the load condition (such as CPU usage rate, memory occupancy, network bandwidth) and comprehensive condition data of each sound node, the priority of each node is dynamically determined. When it is detected that there is a change in the load or condition of any sound node, the system automatically adjusts the audio stream parameters of the corresponding node according to the priority to optimize resource allocation and performance. This method ensures the stability and efficiency of the sound system, and can maintain high-quality audio output even in a complex or changing environment, while preventing the degradation of service quality caused by single-point overload.

[0017] Optionally, the method further includes: monitoring the environmental area where the user is located in real time, and determining whether the current corresponding environmental area has changed; if it has changed, determining the corresponding current environmental area, and adjusting the corresponding working mode and audio output parameters based on the current environmental area.

[0018] By adopting the above technical solution, by monitoring the environmental area where the user is located in real time, the change of the environmental area is automatically detected and responded to. Once a change is detected, the system will immediately determine the new environmental area and adjust the working mode and audio output parameters of the sound accordingly to ensure that the sound effect is always the best. This method enables the sound system to dynamically adapt to environmental changes and provide a continuous, consistent and optimized auditory experience. Whether the user is in the shaded area, the direct sunlight area or the air duct area, they can enjoy high-quality sound that is most suitable for the current environment, greatly improving the sound flexibility and user experience in outdoor activities.

[0019] In a second aspect, the present application provides a sound control system, adopting the following technical solution: An audio control system, characterized by comprising: An environmental area determination module, which obtains comprehensive condition data of the audio usage environment in real time by integrating a variety of high-precision sensors. The comprehensive condition data includes temperature, humidity, wind speed, and crowd density; based on the comprehensive condition data, the corresponding environmental area is determined. Among them, the environmental area includes a shaded area, a direct sunlight area, and a ventilation duct area; when the temperature is lower than the first preset threshold, the wind speed is lower than the second preset threshold, and the crowd density is lower than the third preset threshold, it is used to determine as the shaded area; when the temperature is higher than the fourth preset threshold, the wind speed is lower than the fifth preset threshold, and the crowd density is higher than the sixth preset threshold, it is determined as the direct sunlight area; when the wind speed is higher than the seventh preset threshold and the temperature difference in the surrounding area exceeds the eighth preset threshold, it is used to determine as the ventilation duct area; An application module, which adjusts the working mode and audio output parameters of the audio according to the environmental area, and is used to apply the working mode and audio output parameters to the audio usage stage; An audio layout scheme adjustment module, during the audio usage process, generates a corresponding expected sound field distribution map based on the current corresponding audio layout scheme, sends the expected sound field distribution map to the user, obtains the corresponding user feedback data, and is used to adjust the corresponding audio layout scheme based on the user feedback data; real-time monitors the change of the comprehensive condition data, and dynamically updates the division of the environmental area and the audio parameter settings according to the real-time data.

[0020] In a third aspect, the present application provides an audio control method, adopting the following technical solution: An audio control method, including a processor, and a program of the audio control method described in any one of the above is running in the processor.

[0021] In a fourth aspect, the present application provides a storage medium, adopting the following technical solution: A storage medium stores a program of the audio control method described in any one of the above.

[0022] In summary, the present application includes at least one of the following beneficial technical effects: 1. Real-time collect environmental data such as temperature, humidity, wind speed, and crowd density through high-precision sensors, dynamically divide the outdoor environmental area (such as shaded areas, direct sunlight areas, wind tunnel areas) in combination with preset thresholds, and automatically adjust parameters such as the volume, sound effect, and windproof mode of the speaker according to the characteristics of the area. At the same time, monitor environmental changes in real time to dynamically update the settings. This technical solution ensures that the sound clarity, coverage, and sound effect quality always match the current scene requirements through precise environmental perception and parameter adaptation. For example, enable the windproof and noise reduction mode in the wind tunnel area to reduce wind noise interference, and optimize the volume distribution in crowded areas to avoid local overload, thereby significantly improving the auditory comfort and sound field uniformity in outdoor scenarios and achieving an immersive sound effect experience.

[0023] 2. In addition, eliminate audio out-of-sync problems through multi-node synchronization technology (such as delay compensation algorithms) to ensure seamless connection when multiple speakers play in coordination; combine dynamic sound field modeling with user feedback to adjust the speaker layout and output parameters in real time, so that the sound coverage always precisely matches the crowd distribution and focus areas; at the same time, the system optimizes resource allocation through load monitoring and priority adjustment to ensure stability in complex environments, and intelligently triggers hierarchical alarms in emergency situations to improve the safety response ability. The synergistic effect of these technologies not only solves the problem of unstable sound quality caused by environmental changes, node addition or deletion, or load fluctuations in outdoor scenarios, but also gives users active control over the sound field, realizes the deep adaptation of the sound effect to the scene, and significantly enhances the sense of participation, safety, and refinement level of the sound experience in outdoor activities. BRIEF DESCRIPTION OF THE DRAWINGS

[0024] Figure 1 is a flowchart of a speaker control method shown according to an exemplary embodiment.

[0025] Figure 2 is a structural block diagram of a speaker control method system shown according to an exemplary embodiment. DETAILED DESCRIPTION OF THE EMBODIMENTS

[0026] The following describes in detail the embodiments of the present application, and the examples of the embodiments are shown in the drawings.

[0027] In the description of this specification, the description of reference terms "certain embodiments", "one embodiment", "some embodiments", "illustrative embodiments", "examples", "specific examples", or "some examples" means that the specific features, structures, materials, or characteristics described in connection with the embodiments or examples are included in at least one embodiment or example of the present application. In this specification, the schematic representations of the above terms do not necessarily refer to the same embodiments or examples. Moreover, the specific features, structures, materials, or characteristics described can be combined in a suitable manner in any one or more embodiments or examples.

[0028] An embodiment of the present application discloses an audio control method. Refer to Figure 1 , including: S100, obtaining comprehensive condition data of the audio usage environment in real time by integrating multiple high-precision sensors. The comprehensive condition data includes temperature, humidity, wind speed, and population density; determining the corresponding environmental area based on the comprehensive condition data.

[0029] Among them, the environmental area includes the shaded area, the direct sunlight area, and the air duct area; when the temperature is lower than the first preset threshold, the wind speed is lower than the second preset threshold, and the population density is lower than the third preset threshold, it is determined as the shaded area; when the temperature is higher than the fourth preset threshold, the wind speed is lower than the fifth preset threshold, and the population density is higher than the sixth preset threshold, it is determined as the direct sunlight area; when the wind speed is higher than the seventh preset threshold and the temperature difference in the surrounding area exceeds the eighth preset threshold, it is determined as the air duct area.

[0030] First, deploy high-precision sensors (such as thermometers, hygrometers, anemometers, population density detectors, etc.) in the audio device and its surrounding environment. The sensors can be embedded in the audio device or independently deployed in the audio coverage area. The sensors continuously collect environmental data, including parameters such as temperature, humidity, wind speed, and population density, and transmit them to the central control system in real time through wireless / wired networks.

[0031] The central control system preprocesses the original data, including denoising, calibration (such as zero-point correction of the thermometer), timestamp marking, and stores it in the database; extracts key feature parameters, such as: temperature gradient, temperature difference in different regions; wind speed and wind direction, dominant wind direction and local air duct effect; population distribution pattern, density hotspots and flow trends.

[0032] In the embodiment of the present application, for the shaded area, it is determined when the sensor data meets the following conditions: temperature ≤ the first preset threshold (such as 25 °C); wind speed ≤ the second preset threshold (such as 2 m / s); population density ≤ the third preset threshold (such as 0.5 person / m²). For the direct sunlight area, it is determined when the sensor data meets the following conditions: temperature ≥ the fourth preset threshold (such as 30 °C); wind speed ≤ the fifth preset threshold (such as 3 m / s); population density ≥ the sixth preset threshold (such as 1.5 person / m²). For the air duct area, it is determined when the sensor data meets the following conditions: wind speed ≥ the seventh preset threshold (such as 5 m / s); temperature difference in the surrounding area ≥ the eighth preset threshold (such as 5 °C).

[0033] The system monitors environmental data changes in real time. When key parameters (such as temperature, wind speed) exceed the threshold range, it automatically re-divides the area boundaries and updates the feature information; for example, if the wind speed suddenly increases and the temperature difference increases, the system will trigger the determination of the air duct area. The dynamic update mechanism ensures that the area division always reflects the real-time environmental state. For example, when the afternoon sunlight intensifies, some areas may switch from the shaded area to the direct sunlight area.

[0034] S110. Adjust the working mode and audio output parameters of the audio according to the environmental area, and apply the working mode and audio output parameters during the audio usage stage.

[0035] Among them, the system selects the most suitable audio working mode for this area (such as standard mode, enhanced low-frequency mode, energy-saving mode, etc.), and adjusts the audio output parameters accordingly, including but not limited to: Volume: Adjust the volume according to the environmental noise level and crowd density; Equalization settings: Optimize the frequency response for the acoustic characteristics of different areas; Phase difference: Adjust the phase relationship between each audio node to ensure the synchronous playback effect; Compression ratio and sampling rate: Optimize the transmission efficiency of the audio stream without affecting the sound quality.

[0036] In addition, it is necessary to consider the collaborative work among multiple audio nodes to ensure the best performance of the overall audio system, which includes adjusting the distance, angle and relative position between the audio nodes to achieve uniform sound coverage and consistent auditory experience.

[0037] S120. During the audio usage process, generate the corresponding expected sound field distribution map based on the current corresponding audio layout plan, send the expected sound field distribution map to the user, obtain the corresponding user feedback data, and adjust the corresponding audio layout plan based on the user feedback data; monitor the changes of comprehensive condition data in real time, and dynamically update the area division of the environmental area and the audio parameter settings according to the real-time data.

[0038] Among them, based on the current audio layout plan, the system uses advanced acoustic modeling technology to predict the sound distribution around each audio node, and then generates a detailed expected sound field distribution map; the generated sound field distribution map not only shows the coverage range of the audio system, but also includes the spatial changes of key parameters (such as volume level). The system will convert this map into an easy-to-understand graphical interface, highlighting the sound characteristics of each area.

[0039] In addition, send the generated expected sound field distribution map to the user for viewing through a mobile application, a Web platform or other visualization interfaces. The user can put forward modification opinions or confirm the final layout plan through these channels.

[0040] In addition, the system encourages users to provide feedback, whether it is positive reviews or improvement suggestions. Such feedback can be collected through various channels such as online questionnaires, comment functions in mobile applications, and social media platforms. Based on the collected feedback data, the system identifies common problems or special requirements and adjusts the sound layout plan accordingly. For example, if most users report insufficient volume in a certain area, the system will recommend increasing the number of speakers in that area or adjusting the positions of the existing speakers.

[0041] When necessary, the system will recalculate the optimal audio output parameters and even suggest changing the positions or quantities of the speakers. For major adjustments, the system will also generate a new sound field distribution map for the user to review. Once the adjustment plan is confirmed by the user, the system will automatically implement the new configuration and continue to monitor the environmental conditions and user feedback to ensure that the adjusted sound effect meets expectations.

[0042] By performing the steps from S100 to S120 above, even during the same outdoor activity, listeners in different positions can enjoy a consistent and high-quality auditory experience. At the same time, this intelligent adjustment also helps to save energy and extend the service life of the audio equipment. The expected sound field distribution map can help users intuitively understand the effect of the current sound layout and allow them to make adjustments according to actual needs. This is especially important for large-scale events because it enables organizers to optimize the sound layout before the event, thereby enhancing the overall quality of the event and the satisfaction of participants. And by continuously absorbing user feedback and making corresponding adjustments, the audio system can gradually self-optimize and better meet user expectations.

[0043] In the embodiment of the present application, the method further includes: S1211, continuously monitor the comprehensive condition data of the usage environment, and evaluate the situational characteristics corresponding to the current usage environment based on the comprehensive condition data.

[0044] Among them, the comprehensive condition data of the environment is collected in real time and transmitted to the central control system through wireless or wired networks. The data includes but is not limited to temperature, humidity, wind speed, crowd density, air quality, etc. The central control system preprocesses the received data, such as denoising, calibration, and timestamp marking, and stores the processed data in the database. And the system keeps continuously monitoring the environmental conditions to ensure that all data is up-to-date and can quickly respond to any emergencies.

[0045] Different environmental conditions may indicate different situational characteristics. For example, a sudden increase in crowd density may indicate that the event has reached a climax; abnormal changes in temperature or air quality may suggest potential safety hazards; changes in wind speed and direction may affect the sound propagation effect.

[0046] The central control system uses machine learning algorithms or rule engines to deeply analyze the collected comprehensive condition data, extract key features, and identify the current situation pattern. The system maintains a pre-defined situation feature library covering various common and potential outdoor activity scenarios (such as normal activities, emergency evacuations, severe weather warnings, etc.). By comparing the current data with the features in the library, the most matching situation feature is determined. Based on the above analysis results, the system generates a detailed situation feature assessment report, including the main characteristics of the current environment, potential risk factors, and recommended measures to be taken.

[0047] S1212, Determine whether it is in an emergency based on the situation features. If so, determine the corresponding severity of the situation and the corresponding outdoor situation based on the situation features.

[0048] Among them, according to historical data and expert experience, the system sets emergency thresholds for various situations. For example, when the temperature exceeds a certain limit, the smoke concentration reaches a specific level, or the population density surges abnormally, an emergency alarm is triggered. The central control system continuously compares the current environmental conditions with the preset emergency thresholds in real time. Once it is found that some conditions exceed the safe range, the system immediately triggers an emergency alarm; and to reduce false alarms, the system introduces a multi-verification mechanism, such as cross-verifying the data of different sensors and combining external information sources (such as weather forecasts, official announcements) for confirmation to ensure the accuracy of the alarm.

[0049] In addition, once an emergency is confirmed, the system will evaluate the severity of the emergency according to the specific environmental conditions and situation features. Usually, a grading system is adopted, such as level one (slight), level two (moderate), level three (severe), and each level corresponds to a different emergency response level. The system further analyzes the current outdoor situation, considering factors such as terrain, building distribution, traffic conditions, etc., to determine the most suitable emergency response strategy. For example, in an open area, a more extensive evacuation path plan may be required; while in a densely built-up area, attention needs to be paid to the smoothness of the fire channels. Combining the analysis results of the severity and the outdoor situation, the system generates a detailed comprehensive assessment report to guide subsequent emergency response work.

[0050] S1213, Adjust the urgency and spread range of the alarm based on the severity of the situation and the outdoor situation.

[0051] Among them, according to the comprehensive assessment report, the system automatically adjusts the urgency of the alarm (such as volume, frequency, content, etc.) to ensure that the alarm can attract enough attention without causing excessive panic. For example, a lower volume and slower frequency can be selected during quiet hours at night; while in crowded places, the volume needs to be increased and the spread range needs to be expanded.

[0052] In addition to the audible alarm, the system can also combine various methods such as visual cues (e.g., flashing lights) and tactile cues (e.g., vibration alerts) to help users understand emergency information more quickly, especially suitable for noisy environments or people with hearing impairments; as the emergency situation develops, the system will dynamically adjust the alarm parameters based on the latest data and continuously optimize the alarm effect through user feedback. For example, if it is found that the alarm in a certain area fails to effectively convey the information, the system will immediately adjust the relevant parameters to improve the coverage effect.

[0053] By performing the steps from S1211 to S1213 above, through in-depth analysis of environmental data, the system can perceive potential risks in advance; and quickly and accurately determining whether it is an emergency situation is crucial for ensuring user safety. Triggering the alarm in a timely manner allows organizers and participants to take action quickly and avoid the situation from deteriorating further. At the same time, clarifying the severity of the situation and the specific outdoor situation helps to formulate a more precise emergency response plan. This refined management can protect personnel safety to the greatest extent while reducing unnecessary panic and chaos. Finally, by intelligently adjusting the urgency and spread range of the alarm, the system can quickly and accurately convey important information in an emergency, ensuring that all relevant personnel can understand the situation in a timely manner and take appropriate actions, thus maximizing public safety.

[0054] When multiple audio nodes play audio simultaneously, the method further includes: S1221, continuously obtaining the playback status of each audio node based on the built-in real-time audio stream monitoring module.

[0055] Among them, a high-precision real-time audio stream monitoring module is integrated in each audio node. This module can capture and record the playback status information of the audio node. These modules are connected to the central control system through a high-speed network to ensure low latency and high reliability of data transmission.

[0056] Definition of playback status parameters: Transmission delay, the time difference from when the audio source is sent to when it is received by the audio node, including network transmission time and buffering time; Processing time, the time required for the audio node to process audio frames internally, covering operations such as decoding and mixing; Playback progress, the current audio position being played by the audio node, usually represented by a timestamp or the number of samples.

[0057] At the same time, the monitoring module collects the above playback status parameters at a very high frequency (e.g., multiple times per second) and sends this data to the central control system in real time. The system will update this data regularly (e.g., every few milliseconds) to ensure that all information is up-to-date.

[0058] S1222, determining the time difference and phase difference corresponding to each audio node based on the playback status through a delay compensation algorithm.

[0059] Among them, first, the system selects a reference audio node as the synchronization reference point, and the time differences and phase differences of all other audio nodes will be calculated relative to this reference point.

[0060] For the time difference calculation: Compare the transmission delays between each audio node and the reference node to calculate the time differences between them. In addition, considering that different audio nodes may have different hardware configurations or load conditions, the system will also analyze the processing time differences of each node to further correct the time differences.

[0061] For the phase difference calculation: Calculate the phase differences of each audio node relative to the reference node according to the playback progress of each audio node, which involves the precise positioning of audio frames to ensure the correct phase relationship even under small time differences; in addition, combining the above calculation results, the system applies a specially designed delay compensation algorithm to comprehensively consider the effects of transmission delays, processing times, and playback progress to accurately determine the time differences and phase differences of each audio node.

[0062] S1223, adjust the playback start time and audio frame delay of each audio node based on the time difference and phase difference.

[0063] Among them, for the adjustment of the playback start time: For those audio nodes with longer transmission delays or processing times, the system will calculate an appropriate advance amount so that these nodes can start playing at the correct time point to maintain synchronization with the reference node; the system will dynamically adjust the playback start time of each audio node according to the real-time monitored data to ensure that even under changing network conditions or fluctuating node loads, the synchronized playback effect can be maintained.

[0064] For the adjustment of the audio frame delay: For the audio nodes with phase differences, the system will apply corresponding delay compensation to the audio frames to ensure that the audio frames played by each node are aligned in time with those of other nodes; as the environmental conditions change (such as the wind speed affecting the sound propagation speed), the system will continuously re-evaluate and adjust the audio frame delay to maintain the best synchronization effect.

[0065] After each adjustment, the system will perform synchronization verification to ensure that all audio nodes have indeed achieved the expected synchronization effect. If any asynchronous phenomenon is still found, the system will continue to finely adjust the relevant parameters until the optimal synchronization state is reached.

[0066] By performing the steps from S1221 to S1223 above, the playback status of each audio node is continuously obtained through the built-in real-time audio stream monitoring module. The time difference and phase difference between each node are accurately calculated using the delay compensation algorithm, and based on this, the playback start time and audio frame delay of each audio node are dynamically adjusted. This method not only eliminates the audio out-of-sync problem caused by network transmission or hardware processing but also provides a seamless and consistent sound experience. In this way, whether in large-scale outdoor activities or complex multi-area coverage scenarios, users can enjoy perfectly synchronized high-quality audio effects, greatly improving the stability of the overall audio system and the user experience.

[0067] Meanwhile, in the embodiments of the present application, the method further includes: S1231, monitoring the audio status of each audio node in real time.

[0068] Among them, a high-precision status monitoring module is integrated in each audio node. This module can capture and record the key status information of the audio node. The status monitoring module is connected to the central control system through a high-speed network to ensure low latency and high reliability of data transmission.

[0069] Definition of status parameters: Online status, whether each audio node is in an online state, including whether it is working properly, whether there are faults, etc.; Location information, the exact location of each audio node in the physical space, which can be obtained through GPS or other positioning technologies; Number of audio nodes, the total number of active audio nodes in the current audio system, including newly added or removed nodes.

[0070] Meanwhile, the status monitoring module collects the above status parameters at a very high frequency (such as multiple times per second) and sends this data to the central control system in real time. The central control system updates this data regularly (such as every few seconds) to ensure that all information is up-to-date and can respond in a timely manner to any status changes.

[0071] S1232, when a change in the audio status is detected, obtain the corresponding latest audio node layout.

[0072] Among them, the central control system analyzes the status data received from each audio node in real time. Once a change in the online status, location information, or number of audio nodes of a certain audio node is detected, the status change processing flow is immediately triggered. The system will automatically collect the latest audio node layout information, which includes: the new location of each audio node (if it has moved); newly added or removed audio nodes; the relative position relationship between each audio node.

[0073] Meanwhile, the system will verify the newly obtained layout of the audio nodes to ensure that all information is accurate. For example, it checks for duplicate location information or unreasonable node distributions; if any problems are found, the system will prompt the user for manual confirmation or correction.

[0074] S1233, start the sound field modeling algorithm based on the latest audio node layout, recalculate the sound field distribution based on the latest audio node layout, and adjust the audio output parameters of each audio node based on the sound field distribution.

[0075] Among them, according to the latest audio node layout, the system selects the most suitable sound field modeling algorithm. Different scenarios may require different modeling methods. For example, the wide-area sound field model is used in open areas, and the reflection and diffraction models are used in enclosed spaces. The latest audio node layout information (including position, quantity, relative relationship) and environmental comprehensive condition data (such as temperature, humidity, wind speed, etc.) are passed to the sound field modeling algorithm as input parameters.

[0076] The sound field modeling algorithm calculates the sound distribution around each audio node, such as phenomena like sound wave propagation paths, reflection, diffraction, absorption, etc.; and outputs a detailed sound field distribution map, showing information such as the coverage area of the audio system and the spatial variation of key parameters (such as volume level).

[0077] Based on the information provided by the sound field distribution map, the system automatically adjusts the audio output parameters of each audio node, considering the collaborative work among multiple audio nodes to ensure the best performance of the overall audio system. This includes adjusting the distance, angle, and relative position between the audio nodes to achieve uniform sound coverage and a consistent auditory experience.

[0078] The system will dynamically adjust the audio output parameters of the audio nodes according to the latest sound field distribution map, and continuously optimize the adjustment results through real-time monitoring and user feedback. For example, if the volume in certain areas is still insufficient, the system will further adjust the relevant parameters until the ideal effect is achieved.

[0079] By performing the steps of S1231 to S1233 above, by real-time monitoring the audio status of each audio node, automatically sensing changes in the audio nodes, and obtaining the latest audio node layout when a change is detected. Based on the latest layout, the system starts the sound field modeling algorithm to recalculate the sound field distribution and adjusts the audio output parameters of each audio node accordingly. This method enables the audio system to quickly adapt to node changes, maintain a consistent auditory effect, and greatly improves the flexibility and intelligence level of the audio system in outdoor activities. Users can enjoy the best sound experience after any layout adjustment.

[0080] In addition, the method further includes: S1241, set the corresponding sound focus position in the actual environment.

[0081] Among them, the system can provide a user-friendly mobile application or a web-based platform, allowing users to set the sound focus position through an intuitive interface. Users can select specific focus positions by clicking, dragging, or entering coordinates, etc. And the application combines augmented reality technology to display the sound nodes and the set sound focus positions in the actual environment on the screen of the user's mobile device in real time. Users can visually adjust the focus positions in the real environment to ensure the accuracy and rationality of each focus position.

[0082] After the user sets each sound focus position, the system will verify to ensure that these positions are physically feasible and effective. After passing the verification, the system stores the focus position information in the central control system and prepares for subsequent calculations.

[0083] S1242, based on the sound focus position, calculate through an acoustic modeling algorithm to determine the corresponding audio output parameters.

[0084] Among them, as mentioned before, different scenarios may require different modeling methods. For example, a wide-area sound field model is used in an open area, and a reflection and diffraction model is used in a closed space. The sound focus positions set by the user, the layout of the sound nodes, and the comprehensive environmental condition data are passed to the acoustic modeling algorithm as input parameters, which include the positions, quantities, relative relationships, and environmental characteristics of each sound node.

[0085] The acoustic modeling algorithm uses the input parameters to perform complex physical simulations, calculates the optimal audio output parameters for each sound node to ensure that the sound covers each focus position intensively and evenly. The algorithm outputs detailed audio output parameters, and the audio output parameters include volume, equalizer settings, and phase difference.

[0086] S1243, when the user adjusts or adds a new sound focus position, recalculate and optimize the audio output parameters of the sound network and generate a new sound field distribution map to be displayed to the user in real time.

[0087] Among them, the system monitors in real time whether the user has adjusted the existing sound focus positions or added new focus positions. Once a change is detected, the recalculation process is immediately triggered. The system will verify the newly added or adjusted focus positions to ensure their rationality and effectiveness. For example, check whether there are duplicate positions or unreasonable selections.

[0088] Based on the updated focus position information, the system restarts the acoustic modeling algorithm and recalculates the audio output parameters of the sound network. This process is similar to the initial calculation but takes into account the latest focus position and environmental changes. The system dynamically adjusts the audio output parameters of the sound network according to the recalculated results and continuously optimizes the adjustment results through real-time monitoring and user feedback. For example, if the volume in certain areas is still insufficient, the system will further adjust the relevant parameters until the ideal effect is achieved.

[0089] Finally, based on the recalculated audio output parameters, the system generates a new sound field distribution map, showing information such as the coverage area of the sound system and the spatial variation of key parameters (such as volume levels). This map not only helps users intuitively understand the effect of the current sound arrangement but also provides a basis for subsequent adjustments.

[0090] By performing the steps of S1241 to S1243 above, the user sets the sound focus positions in the actual environment, and based on these focus positions, the optimal audio output parameters are calculated through the acoustic modeling algorithm to ensure that the sound is concentrated and evenly covers each focus. When the user adjusts or adds new sound focus positions, the system recalculates and optimizes the audio output parameters of the sound network and generates a new sound field distribution map in real time for the user to view. This method not only provides a highly personalized sound configuration but also enhances the user's sense of control and participation, ensuring that each adjustment takes effect immediately and bringing a precise and consistent sound experience to the listeners in a specific area. The system significantly improves the sound effect and user satisfaction of outdoor activities, achieving a high degree of unity of intelligence and flexibility.

[0091] In the embodiment of the present application, the method further includes: S12513, obtaining the load situation of each sound node and retrieving the corresponding comprehensive condition data.

[0092] Among them, a lightweight monitoring agent program is deployed on each sound node. This program is responsible for collecting the load information of the node (such as CPU usage, memory occupancy, network bandwidth) and the comprehensive environmental condition data. The monitoring agent program collects the above data at a preset time interval (such as every second or every minute) and transmits the data to the central control system through a secure and efficient communication protocol.

[0093] S1252, determining the corresponding priority for each sound node based on the load situation and the comprehensive condition data.

[0094] Among them, according to the characteristics of the audio system and the requirements of the actual application scenario, a scientific and reasonable priority evaluation model is developed. The factors considered in the priority evaluation model include, but are not limited to, the current load situation of the audio node, its importance in the overall audio layout, the physical environment characteristics of its location, etc. Combining the latest load situation and comprehensive condition data, the priority of each audio node is dynamically calculated. High-priority audio nodes may be because they are located in key areas or undertake more important audio output tasks.

[0095] And the system is built-in with a flexible priority adjustment mechanism, allowing the automatic adjustment of the priority of audio nodes according to actual situation changes (such as newly added sound focus positions, environmental factor changes). This mechanism helps to ensure the efficient operation of the audio system and the optimal utilization of resources.

[0096] S1253, when it is detected that there is a change in the load situation or comprehensive condition data of the audio node, the audio stream parameters of the corresponding audio node are adjusted based on the priority.

[0097] Among them, the system continuously monitors the load situation and comprehensive condition data of the audio node. Once any significant changes are found (such as a sudden increase in the CPU usage rate of a certain audio node or a sharp rise in the temperature of the area where it is located), the parameter adjustment process is immediately triggered. According to the priority of the audio node, the system formulates corresponding audio stream parameter adjustment strategies. This may involve adjustments in multiple aspects such as volume size, equalizer settings, phase difference, etc., aiming to ensure that the audio node can continue to provide high-quality audio output under the new conditions.

[0098] Once the adjustment strategy is confirmed, the system will automatically apply it to the relevant audio nodes without manual intervention. At the same time, the system will also record the process and results of this adjustment for subsequent analysis. In addition, the system will verify the effect of the adjusted audio node to ensure that the adjustment has achieved the expected purpose. If it is found that the ideal effect is not achieved, the system will further fine-tune the relevant parameters until satisfactory.

[0099] By performing the steps of S1241 to S1243 above, through the reasonable allocation of the priority of audio nodes, the system can make optimal decisions under limited resources, ensure that key audio nodes obtain sufficient support, thereby maintaining the stability and reliability of the entire audio system; it can also enable the audio system to quickly respond to changes in internal and external conditions and maintain the best working state. Through the priority-based adjustment of audio stream parameters, the system not only improves its own adaptability and flexibility but also ensures that it can provide users with a consistent and high-quality audio experience even in a complex and changing environment.

[0100] In the embodiment of this application, since it involves determining the corresponding environmental area, the method further includes: S1251, Monitor the environmental area where the user is located in real time, and determine whether the current corresponding environmental area has changed.

[0101] Among them, the system uses technologies such as GPS, Bluetooth beacons, and Wi-Fi positioning to track the user's precise location in real time. Through a mobile application or wearable device, the user's location information is continuously sent to the central control system. The central control system uses a predefined environmental area model, combines sensor data and user location information, and intelligently identifies the environmental area where the user is currently located.

[0102] In addition, the system continuously monitors changes in the user's location and the surrounding environmental conditions, sets reasonable thresholds to determine whether the environmental area has changed significantly. For example, when the temperature, humidity, or wind speed exceeds a certain range, it triggers a status change detection. The system compares and analyzes the latest sensor data with the previously recorded data to determine whether there are significant differences. If it is found that some key parameters (such as rising temperature, increasing humidity) exceed the preset threshold, it is considered that the environmental area has changed.

[0103] S1252, If a change occurs, determine the corresponding current environmental area, and adjust the corresponding working mode and audio output parameters based on the current environmental area.

[0104] Among them, when the system detects a change in the environmental area, it immediately starts a new area feature extraction process, which includes re-evaluating key parameters such as temperature, humidity, wind speed, and light intensity to determine the characteristics of the new environmental area where the user is currently located. The system matches the extracted new area features with the predefined environmental area model to find the environmental area category that best suits the current conditions. For example, changing from a "shaded area" to a "sunny area", or from a "wind duct area" to an "open area".

[0105] After determining the new environmental area, the system generates a detailed area confirmation report and notifies the user of the current environmental area and its characteristics through a mobile application or other interface. The user can understand the environmental change through this notification and be prepared for possible audio adjustments.

[0106] According to the newly determined environmental area, the system automatically selects the most suitable audio working mode for this area. For example, in the wind duct area, an enhanced low-frequency mode may be selected to combat wind noise; in the shaded area, an energy-saving mode can be selected to reduce energy consumption. The system dynamically adjusts the audio output parameters based on the characteristics of the new environmental area, and this includes adjusting the distance, angle, and relative position between audio nodes to achieve uniform sound coverage and a consistent auditory experience.

[0107] Based on the new environmental area to adjust the working mode and audio output parameters of the sound system, the system can ensure that even under changing environmental conditions, optimal sound coverage and clarity can be provided. This method not only improves the user experience, but also enhances the stability and flexibility of the sound system, ensuring that every environmental change takes effect immediately and providing the most suitable sound experience for users.

[0108] An embodiment of this application discloses an audio control system. Referring to Figure 2 , the system includes but is not limited to: An environmental area determination module 200, which obtains comprehensive condition data of the sound system usage environment in real time by integrating multiple high-precision sensors. The comprehensive condition data includes temperature, humidity, wind speed, and crowd density; based on the comprehensive condition data, the corresponding environmental area is determined. Among them, the environmental area includes a shaded area, a direct sunlight area, and a wind duct area; when the temperature is lower than the first preset threshold, the wind speed is lower than the second preset threshold, and the crowd density is lower than the third preset threshold, it is determined as the shaded area; when the temperature is higher than the fourth preset threshold, the wind speed is lower than the fifth preset threshold, and the crowd density is higher than the sixth preset threshold, it is determined as the direct sunlight area; when the wind speed is higher than the seventh preset threshold and the temperature difference in the surrounding area exceeds the eighth preset threshold, it is determined as the wind duct area; An application module 210, which adjusts the working mode and audio output parameters of the sound system according to the environmental area, and is used to apply the working mode and audio output parameters during the sound system usage stage; A sound arrangement plan adjustment module 220, during the sound system usage process, generates a corresponding expected sound field distribution map based on the current corresponding sound arrangement plan, sends the expected sound field distribution map to the user, obtains the corresponding user feedback data, and adjusts the corresponding sound arrangement plan based on the user feedback data; monitors the change of the comprehensive condition data in real time, and dynamically updates the division of the environmental area and the sound parameter settings according to the real-time data.

[0109] Furthermore, the system includes but is not limited to: A comprehensive condition data monitoring module, which is used to monitor the comprehensive condition data of the usage environment in real time, and evaluate the corresponding situation characteristics of the current usage environment based on the comprehensive condition data; A situation judgment module, which is used to judge whether it is in an emergency based on the situation characteristics. If so, determine the corresponding severity of the situation, and determine the corresponding outdoor situation based on the situation characteristics; An emergency severity and propagation range adjustment module, which adjusts the emergency severity and propagation range of the alarm based on the severity of the situation and the outdoor situation.

[0110] Furthermore, the system includes but is not limited to: A playback status acquisition module, based on a built-in real-time audio stream monitoring module, is used to continuously acquire the playback status of each audio node. Among them, the playback status includes the transmission delay, processing time, and playback progress of the audio stream; A time difference and phase difference determination module, based on the playback status, uses a delay compensation algorithm to determine the time difference and phase difference corresponding to each audio node; A playback start time and audio frame delay adjustment module, based on the time difference and phase difference, is used to adjust the playback start time and audio frame delay of each audio node.

[0111] Furthermore, the system includes but is not limited to: An audio device status monitoring module, used to monitor the audio device status of each audio node in real time. Among them, the audio device status includes the online status, location information, and the number of audio nodes; A latest audio node layout acquisition module, when detecting a change in the audio device status, is used to acquire the corresponding latest audio node layout; An audio output parameter adjustment module, based on the latest audio node layout, starts a sound field modeling algorithm, recalculates the sound field distribution based on the latest audio node layout, and is used to adjust the audio output parameters of each audio node based on the sound field distribution.

[0112] Furthermore, the system includes but is not limited to: A sound focus position setting module, used to set the corresponding sound focus position in the actual environment; An audio output parameter determination module, based on the sound focus position, calculates through an acoustic modeling algorithm, and is used to determine the corresponding audio output parameters. Among them, the audio output parameters include volume, equalizer settings, and phase difference; An optimization module, when the user adjusts or adds a new sound focus position, recalculates and is used to optimize the audio output parameters of the audio network and generate a new sound field distribution map to be displayed to the user in real time.

[0113] Furthermore, the system includes but is not limited to: A load condition acquisition module, used to acquire the load condition of each audio node, and retrieve the corresponding comprehensive condition data. Among them, the load condition includes CPU usage rate, memory occupancy, and network bandwidth; A priority determination module, based on the load condition and comprehensive condition data, is used to determine the priority corresponding to each audio node; An audio stream parameter adjustment module, when detecting a change in the load condition or comprehensive condition data of an audio node, adjusts the audio stream parameters of the corresponding audio node based on the priority.

[0114] Furthermore, the system includes but is not limited to: An environmental area monitoring module is used to monitor the environmental area where the user is located in real time and determine whether the corresponding current environmental area has changed; A current environmental area determination module, if a change occurs, is used to determine the corresponding current environmental area and adjust the corresponding working mode and audio output parameters based on the current environmental area.

[0115] An embodiment of the present application also discloses an audio control method, including a processor, and a program of the audio control method described in any one of the above is run in the processor.

[0116] An embodiment of the present application also discloses a storage medium storing a program of the audio control method described in any one of the above.

[0117] Although the embodiments of the present application have been shown and described above, it can be understood that the above embodiments are exemplary and should not be construed as limiting the present application. Those of ordinary skill in the art can make changes, modifications, substitutions, and variations to the above embodiments within the scope of the present application.

Claims

1. A sound control method, characterized in that: include: By integrating multiple high-precision sensors, comprehensive condition data of the audio environment can be obtained in real time, including temperature, humidity, wind speed and crowd density; Determine the corresponding environmental area based on the comprehensive condition data, wherein the environmental area includes a sunshade area, a direct sunlight area, and a wind duct area; when the temperature is lower than a first preset threshold, the wind speed is lower than a second preset threshold, and the crowd density is lower than a third preset threshold, it is determined to be a sunshade area; when the temperature is higher than a fourth preset threshold, the wind speed is lower than a fifth preset threshold, and the crowd density is higher than a sixth preset threshold, it is determined to be a direct sunlight area; when the wind speed is higher than a seventh preset threshold and the temperature difference of the surrounding area exceeds an eighth preset threshold, it is determined to be a wind duct area; Adjusting the working mode and audio output parameters of the audio system according to the environmental area, and applying the working mode and audio output parameters to the audio system use stage; During the use of the audio system, a corresponding expected sound field distribution map is generated based on the current corresponding audio layout plan, the expected sound field distribution map is sent to the user, the corresponding user feedback data is obtained, and the corresponding audio layout plan is adjusted based on the user feedback data; the changes of the comprehensive condition data are monitored in real time, and the division of the environmental area and the audio parameter settings are dynamically updated according to the real-time data.

2. The sound control method according to claim 1, characterized in that: The method also includes: Monitor the comprehensive condition data of the usage environment in real time, and evaluate the situational characteristics corresponding to the current usage environment based on the comprehensive condition data; Determine whether it is in an emergency situation based on the situational characteristics, and if so, determine the severity of the situation, and determine the corresponding outdoor situation based on the situational characteristics; The urgency and propagation range of the alarm are adjusted based on the severity of the situation and the outdoor conditions.

3. The sound control method according to claim 1, characterized in that: When multiple audio nodes play audio simultaneously, the method further includes: Based on the built-in real-time audio stream monitoring module, the playback status of each audio node is continuously obtained, where the playback status includes the transmission delay, processing time and playback progress of the audio stream; Determine the time difference and phase difference corresponding to each audio node through a delay compensation algorithm based on the playback state; The play start time and the audio frame delay of each audio node are adjusted based on the time difference and the phase difference.

4. The sound control method according to claim 3, characterized in that: The method also includes: Monitor the audio status of each audio node in real time, where the audio status includes online status, location information, and the number of audio nodes; When a change in the audio state is detected, the corresponding latest audio node layout is obtained; A sound field modeling algorithm is started based on the latest sound node layout, the sound field distribution is recalculated based on the latest sound node layout, and the audio output parameters of each sound node are adjusted based on the sound field distribution.

5. The sound control method according to claim 1, characterized in that: The method also includes: Set the corresponding sound focus position in the actual environment; Based on the sound focus position, a corresponding audio output parameter is determined by calculating through an acoustic modeling algorithm, wherein the audio output parameter includes volume, equalizer setting, and phase difference; When the user adjusts or adds a new sound focus position, the audio output parameters of the audio network are recalculated and optimized, and a new sound field distribution map is generated and displayed to the user in real time.

6. The sound control method according to claim 1, characterized in that: The method also includes: Obtain the load status of each audio node and retrieve the corresponding comprehensive condition data, where the load status includes CPU usage, memory usage, and network bandwidth; Determine the priority corresponding to each audio node based on the load condition and the comprehensive condition data; When it is detected that there is a load condition of the audio node or a change in the comprehensive condition data, the audio stream parameters of the corresponding audio node are adjusted based on the priority.

7. The sound control method according to claim 1, characterized in that: The method also includes: Monitor the user's environment in real time to determine whether the corresponding environment has changed; If a change occurs, the corresponding current environment area is determined, and the corresponding working mode and audio output parameters are adjusted based on the current environment area.

8. A sound control system, characterized in that: include: The environment area determination module integrates a variety of high-precision sensors to obtain comprehensive condition data of the audio environment in real time, including temperature, humidity, wind speed and crowd density; Determine the corresponding environmental area based on the comprehensive condition data, wherein the environmental area includes a sunshade area, a direct sunlight area, and a wind duct area; when the temperature is lower than a first preset threshold, the wind speed is lower than a second preset threshold, and the crowd density is lower than a third preset threshold, it is used to determine the sunshade area; when the temperature is higher than a fourth preset threshold, the wind speed is lower than a fifth preset threshold, and the crowd density is higher than a sixth preset threshold, it is determined to be a direct sunlight area; when the wind speed is higher than a seventh preset threshold and the temperature difference of the surrounding area exceeds an eighth preset threshold, it is used to determine the wind duct area; An application module, which adjusts the working mode and audio output parameters of the audio system according to the environmental area, and is used to apply the working mode and audio output parameters to the audio system use stage; The audio layout adjustment module generates a corresponding expected sound field distribution map based on the current corresponding audio layout plan during the use of the audio, sends the expected sound field distribution map to the user, obtains corresponding user feedback data, and adjusts the corresponding audio layout plan based on the user feedback data; monitors the changes of the comprehensive condition data in real time, and dynamically updates the division of the environmental area and the audio parameter settings according to the real-time data.

9. A sound control method, characterized in that: The device comprises a processor, wherein a program of the sound control method according to any one of claims 1 to 7 is executed in the processor.

10. A storage medium, characterized in that: A program for the sound control method according to any one of claims 1 to 7 is stored.

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