An audio control method and system

Abstract

The application relates to the technical field of audio control, in particular to a sound control method, which comprises the following steps: integrated multiple high-precision sensors are used to acquire comprehensive condition data of a sound use environment in real time; the comprehensive condition data is identified based on the comprehensive condition data; corresponding environment areas are determined; the comprehensive condition data comprises temperature, humidity, wind speed and crowd density; the environment areas comprise a sun-shading area, a direct sunlight area and a wind channel area; the working mode and audio output parameters of the sound are adjusted based on the environment areas; the working mode and the audio output parameters are applied to a sound use stage; in the sound use process, corresponding expected sound field distribution maps are generated based on current corresponding sound arrangement schemes; the expected sound field distribution maps are sent to users; corresponding user feedback data is acquired; and the corresponding sound arrangement schemes are adjusted based on the user feedback data; and the application can improve the sound experience of users outdoors.

Description

Technical Field

[0001] This application relates to the technical field of audio control, and in particular to a sound control method and system. Background Technology

[0002] Most audio systems on the market are primarily designed for indoor use, offering limited support for outdoor activities. As lifestyles change, more and more people desire a high-quality music experience during outdoor activities, such as camping, hiking, or playing background music while relaxing in a park. However, existing audio equipment often lacks effective solutions to 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, there are still problems to be solved in order to improve the user's audio experience outdoors. Summary of the Invention

[0004] To improve the user's audio experience outdoors, this application provides an audio control method and system.

[0005] Firstly, this application provides a sound control method, which adopts the following technical solution:

[0006] A sound system control method includes: acquiring comprehensive environmental condition data of the sound system's operating environment in real time by integrating multiple high-precision sensors, the comprehensive condition data including temperature, humidity, wind speed, and crowd density; determining corresponding environmental areas based on the comprehensive condition data, wherein the environmental areas include shaded areas, direct sunlight areas, and wind duct areas; identifying a shaded area when the temperature is below a first preset threshold, the wind speed is below a second preset threshold, and the crowd density is below a third preset threshold; identifying a direct sunlight area when the temperature is above a fourth preset threshold, the wind speed is below a fifth preset threshold, and the crowd density is above a sixth preset threshold; identifying a wind duct area when the wind speed is above a seventh preset threshold and the temperature difference in the surrounding area exceeds an eighth preset threshold; adjusting the sound system's operating mode and audio output parameters according to the environmental areas, and applying the operating mode and audio output parameters to the sound system's operating phase; generating a corresponding expected sound field distribution map based on the current corresponding sound system layout scheme during sound system use, sending the expected sound field distribution map to the user, obtaining corresponding user feedback data, and adjusting the corresponding sound system layout scheme based on the user feedback data; and monitoring changes in the comprehensive condition data in real time, and dynamically updating the environmental area division and sound system parameter settings based on real-time data.

[0007] By adopting the above technical solution, high-precision sensors are integrated to collect real-time data on temperature, humidity, wind speed, and crowd density. Outdoor environmental areas (such as shaded areas, direct sunlight areas, and wind tunnel areas) are dynamically divided based on preset thresholds. The volume, sound effects, and windproof mode of the speakers are automatically adjusted according to the characteristics of each area. At the same time, the parameter settings are dynamically updated by monitoring environmental changes in real time, and the speaker layout is further optimized based on user feedback on the expected sound field distribution. This allows for precise adaptation of sound effects (such as reducing wind noise, enhancing bass, and adjusting volume distribution) in different outdoor scenarios, ensuring that sound clarity and coverage always match the needs of the environment and the people. This significantly improves auditory comfort, anti-interference ability, and sound field uniformity during outdoor activities, achieving an immersive sound experience.

[0008] Optionally, the method further includes: real-time monitoring of comprehensive condition data of the usage environment; evaluating the situational characteristics corresponding to the current usage environment based on the comprehensive condition data; determining whether an emergency is occurring based on the situational characteristics; if so, determining the severity of the situation and determining the corresponding outdoor conditions based on the situational characteristics; and adjusting the urgency level and propagation range of the alarm based on the severity of the situation and the outdoor conditions.

[0009] By employing the aforementioned technical solution and monitoring comprehensive environmental data in real time, the system can assess the characteristics of the current situation and intelligently determine whether an emergency has occurred. Upon detecting an emergency, the system determines the severity and corresponding outdoor conditions based on the specific circumstances, automatically adjusting the urgency level and dissemination range of the alarm. This method ensures that information can be quickly and accurately delivered to those in need during emergencies, enhancing user safety while minimizing unnecessary panic and disruption. This intelligent situational awareness and response mechanism significantly improves the safety and organizational efficiency of outdoor activities.

[0010] Optionally, when multiple speaker nodes play audio simultaneously, the method further includes: continuously acquiring the playback status of each speaker node based on a built-in real-time audio stream monitoring module, wherein 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 speaker node based on the playback status using a delay compensation algorithm; and adjusting the playback start time and audio frame delay of each speaker node based on the time difference and the phase difference.

[0011] By employing the above technical solution, the built-in real-time audio stream monitoring module continuously tracks the playback status of each speaker node and uses a latency compensation algorithm to accurately calculate the time and phase differences between nodes. Based on these differences, the system dynamically adjusts the playback start time and audio frame latency of each speaker node, ensuring perfect synchronization when multiple speaker nodes play simultaneously. This method effectively eliminates audio desynchronization problems caused by network transmission or hardware processing, providing a seamless and consistent sound experience. It is particularly suitable for large outdoor events or multi-area coverage scenarios, significantly improving the overall stability of the audio system and the user experience.

[0012] Optionally, the method further includes: real-time monitoring of the speaker status of each speaker node, wherein the speaker status includes online status, location information, and the number of speaker nodes; when a change in speaker status is detected, obtaining the corresponding latest speaker node layout; starting a sound field modeling algorithm based on the latest speaker node layout, recalculating the sound field distribution based on the latest speaker node layout, and adjusting the audio output parameters of each speaker node based on the sound field distribution.

[0013] By employing the above technical solution, the system automatically detects changes in the speaker status by monitoring the online status, location information, and number of each speaker node in real time. Once a change is detected, the system immediately obtains the latest speaker node layout and initiates a 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 speaker node to ensure consistently uniform and clear sound coverage. This method enables the speaker system to quickly adapt to changes in nodes, maintaining a consistent auditory experience, enhancing the flexibility and intelligence of the sound system during outdoor activities, and ensuring users enjoy the best sound experience regardless of any layout adjustments.

[0014] Optionally, the method further includes: setting a corresponding sound focus position in the actual environment; calculating the corresponding audio output parameters based on the sound focus position using an acoustic modeling algorithm, wherein 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 speaker network and generating a new sound field distribution map to be displayed to the user in real time.

[0015] By employing the above technical solution, the sound focus positions are set in the actual environment, and the optimal audio output parameters (such as volume, equalizer settings, and phase difference) are calculated using acoustic modeling algorithms to ensure that the sound is concentrated and evenly covered to each focus. When the user adjusts or adds a new sound focus, the system instantly recalculates and optimizes the audio output parameters of the speaker network, generating a new sound field distribution map that is displayed to the user in real time. This method not only provides highly personalized speaker configurations but also enhances the user's control and sense of participation, ensuring that every adjustment takes effect immediately. It delivers a precise and consistent sound experience to listeners in a specific area, greatly improving the sound quality and user satisfaction for outdoor activities.

[0016] Optionally, the method further includes: obtaining the load status of each speaker node and retrieving the corresponding comprehensive condition data, wherein the load status includes CPU utilization, memory usage, and network bandwidth; determining the priority of each speaker node based on the load status and the comprehensive condition data; and adjusting the audio stream parameters of the corresponding speaker node based on the priority when a change in the load status or comprehensive condition data of a speaker node is detected.

[0017] By employing the above technical solution, the priority of each speaker node is dynamically determined by monitoring its load (such as CPU utilization, memory usage, and network bandwidth) and other comprehensive condition data. When a change in the load or conditions of any speaker node is detected, the system automatically adjusts the audio stream parameters of the corresponding node according to its priority to optimize resource allocation and performance. This approach ensures the stability and efficiency of the speaker system, maintaining high-quality audio output even in complex or variable environments, while preventing service quality degradation due to single-point overload.

[0018] Optionally, the method further includes: real-time monitoring of the user's environment and determining whether the current environment has changed; if it has changed, determining the current environment and adjusting the corresponding working mode and audio output parameters based on the current environment.

[0019] By employing the aforementioned technical solution, the system automatically detects and responds to changes in the user's environment in real time. Upon detecting a change, the system immediately identifies the new environmental area and adjusts the speaker's operating mode and audio output parameters accordingly to ensure optimal sound quality at all times. This approach allows the speaker system to dynamically adapt to environmental changes, providing a consistent and optimized listening experience. Whether the user is in a shaded area, a sunlit area, or a windy area, they can enjoy high-quality sound best suited to the current environment, greatly enhancing the flexibility of the audio system and the user experience during outdoor activities.

[0020] Secondly, this application provides an audio control system, which adopts the following technical solution:

[0021] An audio control system, characterized in that it comprises:

[0022] The environmental area determination module integrates multiple high-precision sensors to acquire comprehensive environmental condition data in real time, including temperature, humidity, wind speed, and crowd density. Based on this comprehensive condition data, it determines the corresponding environmental area, which includes a shaded area, a direct sunlight area, and a wind tunnel area. When the temperature is below a first preset threshold, the wind speed is below a second preset threshold, and the crowd density is below a third preset threshold, it is determined to be a shaded area. When the temperature is above a fourth preset threshold, the wind speed is below a fifth preset threshold, and the crowd density is above a sixth preset threshold, it is determined to be a direct sunlight area. When the wind speed is above a seventh preset threshold and the temperature difference between the surrounding area and the surrounding area exceeds an eighth preset threshold, it is determined to be a wind tunnel area.

[0023] The application module adjusts the working mode and audio output parameters of the speaker according to the environmental area, and applies the working mode and audio output parameters to the speaker during use.

[0024] The audio setup adjustment module generates a corresponding expected sound field distribution map based on the current audio setup during audio use, sends the expected sound field distribution map to the user, obtains corresponding user feedback data, and adjusts the corresponding audio setup based on the user feedback data; it also monitors changes in the comprehensive condition data in real time and dynamically updates the environmental area division and audio parameter settings based on the real-time data.

[0025] Thirdly, this application provides a sound control method, which adopts the following technical solution:

[0026] An audio control method includes a processor, wherein the processor runs a program of any one of the above-described audio control methods.

[0027] Fourthly, this application provides a storage medium, which adopts the following technical solution:

[0028] A storage medium storing a program for the audio control method described in any one of the above.

[0029] In summary, this application includes at least one of the following beneficial technical effects:

[0030] 1. This technology uses high-precision sensors to collect real-time environmental data such as temperature, humidity, wind speed, and crowd density. Combined with preset thresholds, it dynamically divides outdoor environments into zones (e.g., shaded areas, direct sunlight areas, wind tunnel areas). Based on the characteristics of each zone, it automatically adjusts speaker parameters such as volume, sound effects, and windproof mode, while simultaneously monitoring environmental changes to dynamically update settings. This technical solution ensures that sound clarity, coverage, and sound quality always match the needs of the current scenario through precise environmental perception and parameter adaptation. For example, it activates windproof noise reduction mode in wind tunnel areas to reduce wind noise interference, and optimizes volume distribution in densely populated areas to avoid localized overload, thereby significantly improving auditory comfort and sound field uniformity in outdoor scenarios and achieving an immersive audio experience.

[0031] 2. Furthermore, multi-node synchronization technology (such as delay compensation algorithms) eliminates audio desynchronization issues, ensuring seamless integration when multiple speakers play together. Combined with dynamic sound field modeling and user feedback, the system adjusts speaker layout and output parameters in real time, ensuring sound coverage accurately matches crowd distribution and focal areas. Simultaneously, the system optimizes resource allocation through load monitoring and priority adjustment, guaranteeing stability in complex environments and intelligently triggering tiered alarms in emergencies to enhance safety response capabilities. These technologies work synergistically to not only solve the problem of unstable sound quality in outdoor scenarios caused by environmental changes, node additions or removals, or load fluctuations, but also empower users with active control over the sound field, achieving deep adaptation of sound effects to the scene and significantly enhancing the sense of participation, safety, and refinement of the sound experience in outdoor activities. Attached Figure Description

[0032] Figure 1 This is a flowchart illustrating an audio control method according to an exemplary embodiment.

[0033] Figure 2 This is a structural block diagram of an audio control method system according to an exemplary embodiment. Detailed Implementation

[0034] The embodiments of this application are described in detail below, and examples of the embodiments are shown in the accompanying drawings.

[0035] In the description of this specification, the references to "certain embodiments," "one embodiment," "some embodiments," "illustrative embodiment," "example," "specific example," or "some examples" refer to specific features, structures, materials, or characteristics described in connection with the described embodiment or example, which are included in at least one embodiment or example of this application. In this specification, the illustrative expressions of the above terms do not necessarily refer to the same embodiment or example. Furthermore, the specific features, structures, materials, or characteristics described may be combined in any suitable manner in one or more embodiments or examples.

[0036] This application discloses an audio control method, referring to... Figure 1 ,include:

[0037] The S100 integrates multiple high-precision sensors to acquire comprehensive environmental data in real time, including temperature, humidity, wind speed, and crowd density; and determines the corresponding environmental area based on the comprehensive environmental data.

[0038] The environmental area includes a shaded area, a direct sunlight area, and a wind tunnel area. A shaded area is defined when the temperature is below the first preset threshold, the wind speed is below the second preset threshold, and the population density is below the third preset threshold. A direct sunlight area is defined when the temperature is above the fourth preset threshold, the wind speed is below the fifth preset threshold, and the population density is above the sixth preset threshold. A wind tunnel area is defined when the wind speed is above the seventh preset threshold and the temperature difference between the surrounding areas exceeds the eighth preset threshold.

[0039] First, high-precision sensors (such as thermometers, hygrometers, anemometers, and crowd density detectors) are deployed in and around the audio equipment. These sensors can be embedded in the audio equipment or deployed independently within the audio coverage area. The sensors continuously collect environmental data, including parameters such as temperature, humidity, wind speed, and crowd density, and transmit this data in real time to the central control system via wireless / wired networks.

[0040] The central control system preprocesses the raw data, including noise reduction, calibration (such as zero-point calibration of thermometers), timestamp annotation, and stores it in the database; it extracts key feature parameters, such as: temperature gradient, temperature difference in different areas; wind speed and direction, prevailing wind direction and local wind channel effect; population distribution patterns, density hotspots and flow trends.

[0041] In this embodiment, the following conditions are met when sensor data in the shaded area are satisfied: temperature ≤ first preset threshold (e.g., 25℃); wind speed ≤ second preset threshold (e.g., 2m / s); crowd density ≤ third preset threshold (e.g., 0.5 people / m²). In the direct sunlight area, the following conditions are met when sensor data in the shaded area are satisfied: temperature ≥ fourth preset threshold (e.g., 30℃); wind speed ≤ fifth preset threshold (e.g., 3m / s); crowd density ≥ sixth preset threshold (e.g., 1.5 people / m²). In the wind tunnel area, the following conditions are met when sensor data in the wind tunnel are satisfied: wind speed ≥ seventh preset threshold (e.g., 5m / s); temperature difference from the surrounding area ≥ eighth preset threshold (e.g., 5℃).

[0042] The system monitors environmental data changes in real time. When key parameters (such as temperature and wind speed) exceed threshold ranges, it automatically re-delineates area boundaries and updates feature information. For example, if wind speed suddenly increases and temperature difference widens, the system will trigger a wind corridor zone determination. The dynamic update mechanism ensures that area division always reflects the real-time environmental status. For example, when the afternoon sun intensifies, some areas may switch from shaded areas to areas under direct sunlight.

[0043] S110 adjusts the operating mode and audio output parameters of the speaker according to the environmental area, and applies the operating mode and audio output parameters to the speaker during use.

[0044] The system selects the most suitable audio operating mode for the 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 ambient 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 synchronized playback; compression ratio and sampling rate: optimize the transmission efficiency of the audio stream without affecting the sound quality.

[0045] In addition, the collaborative work between multiple speaker nodes needs to be considered to ensure the best performance of the overall sound system. This includes adjusting the distance, angle and relative position between speaker nodes to achieve uniform sound coverage and a consistent listening experience.

[0046] S120 generates a corresponding expected sound field distribution map based on the current corresponding sound system layout during the use of the audio system, sends the expected sound field distribution map to the user, obtains corresponding user feedback data, adjusts the corresponding sound system layout based on the user feedback data, monitors changes in comprehensive condition data in real time, and dynamically updates the division of environmental areas and audio parameter settings according to real-time data.

[0047] Based on the current sound system layout, the system uses advanced acoustic modeling technology to predict the sound distribution around each sound node, thereby generating a detailed expected sound field distribution map. The generated sound field distribution map not only shows the coverage of the sound system, but also includes the spatial variation of key parameters (such as volume level). The system will convert this map into an easy-to-understand graphical interface to highlight the sound characteristics of each area.

[0048] In addition, the generated expected sound field distribution map will be sent to users for viewing via mobile applications, web platforms, or other visual interfaces, allowing users to provide feedback or confirm the final layout plan through these channels.

[0049] In addition, the system encourages users to provide feedback, whether it is positive evaluation or suggestions for improvement. This 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 system layout accordingly. For example, if most users report that the volume in a certain area is insufficient, the system will suggest increasing the number of speakers in that area or adjusting the position of the existing speakers.

[0050] If necessary, the system will recalculate the optimal audio output parameters and may even suggest changing the position or number of speakers. For major adjustments, the system will also regenerate a new sound field distribution map for user review. Once the adjustment plan is confirmed by the user, the system will automatically implement the new configuration and continue to monitor environmental conditions and user feedback to ensure that the adjusted sound effect meets expectations.

[0051] By executing steps S100 to S120 as described above, listeners in different locations can enjoy a consistent and high-quality auditory experience even during the same outdoor event. This intelligent adjustment also helps save energy and extend the lifespan of the sound equipment. The anticipated sound field distribution map helps users intuitively understand the effect of the current sound setup and allows them to make adjustments based on actual needs. This is especially important for large events, as it enables organizers to optimize the sound layout before the event, thereby improving the overall quality of the event and participant satisfaction. Furthermore, by continuously incorporating user feedback and making corresponding adjustments, the sound system can gradually self-optimize, better meeting user expectations.

[0052] In this embodiment of the application, the method further includes:

[0053] S1211, monitors comprehensive condition data of the usage environment in real time, and evaluates the contextual characteristics corresponding to the current usage environment based on the comprehensive condition data.

[0054] The system collects comprehensive environmental data in real time and transmits it to the central control system via wireless or wired networks. The data includes, but is not limited to, temperature, humidity, wind speed, population density, and air quality. The central control system preprocesses the received data, such as by noise reduction, calibration, and timestamp, and stores the processed data in a database. The system also continuously monitors environmental conditions to ensure that all data is up-to-date and can respond quickly to any emergencies.

[0055] Different environmental conditions may indicate different situational characteristics. For example, a sudden increase in crowd density may indicate that an event is reaching its climax; abnormal temperature or air quality changes may indicate potential safety hazards; and changes in wind speed and direction may affect the propagation of sound.

[0056] The central control system utilizes machine learning algorithms or rule engines to conduct in-depth analysis of the collected comprehensive conditional data, extracting key features and identifying current situational patterns. The system maintains a predefined situational 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 system determines the most matching situational features. Based on the above analysis results, the system generates a detailed situational feature assessment report, including the main characteristics of the current environment, potential risk factors, and recommended measures.

[0057] S1212, determine whether it is an emergency based on the context features. If so, determine the severity of the situation and determine the corresponding outdoor situation based on the context features.

[0058] Based on historical data and expert experience, the system has set emergency thresholds for various scenarios. For example, an emergency alarm is triggered when the temperature exceeds a certain limit, the smoke concentration reaches a specific level, or the population density surges abnormally. The central control system compares the current environmental conditions with the preset emergency thresholds in real time. Once any condition is detected to exceed the safe range, the system immediately triggers an emergency alarm. To reduce false alarms, the system incorporates multiple verification mechanisms, such as cross-validating data from different sensors and confirming the information with external sources (such as weather forecasts and official announcements), ensuring the accuracy of the alarm.

[0059] Furthermore, once an emergency is confirmed, the system assesses its severity based on specific environmental conditions and situational characteristics. A tiered system is typically used, such as Level 1 (minor), Level 2 (moderate), and Level 3 (severe), each corresponding to a different emergency response level. The system further analyzes the current outdoor conditions, considering factors such as terrain, building distribution, and traffic conditions to determine the most suitable emergency response strategy. For example, more extensive evacuation route planning may be needed in open areas, while in densely built-up areas, the unobstructed access of fire lanes must be prioritized. Combining the severity assessment and outdoor condition analysis results, the system generates a detailed comprehensive evaluation report to guide subsequent emergency response efforts.

[0060] S1213, adjust the urgency and reach of the alarm based on the severity of the situation and outdoor conditions.

[0061] Based on the comprehensive evaluation report, the system automatically adjusts the urgency level of the alarm (such as volume, frequency, and content) to ensure that the alarm can attract sufficient attention without causing excessive fright. For example, a lower volume and slower frequency can be selected during quiet nighttime hours, while the volume needs to be increased and the range of propagation expanded in crowded places.

[0062] In addition to sound alarms, the system can also combine visual cues (such as flashing lights) and tactile cues (such as vibration alerts) to help users understand emergency information more quickly, which is especially suitable for noisy environments or people with hearing impairments. As the emergency 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 is not effectively transmitted, the system will immediately adjust the relevant parameters to improve the coverage.

[0063] By executing steps S1211 to S1213 above, and through in-depth analysis of environmental data, the system can proactively detect potential risks. Quickly and accurately determining whether an emergency has occurred is crucial for ensuring user safety. Timely alarm triggering allows organizers and participants to take swift action, preventing further escalation. Furthermore, clearly defining the severity of the situation and specific outdoor conditions helps in developing more precise emergency response plans. This refined management maximizes personnel safety while minimizing unnecessary panic and chaos. Finally, by intelligently adjusting the urgency and reach of alarms, the system can rapidly and accurately convey vital information in emergencies, ensuring all relevant personnel are informed and can take appropriate action, thereby maximizing public safety.

[0064] When multiple speaker nodes play audio simultaneously, the method also includes:

[0065] S1221 continuously acquires the playback status of each speaker node based on the built-in real-time audio stream monitoring module.

[0066] Each speaker node integrates a high-precision real-time audio stream monitoring module, which can capture and record the playback status information of the speaker 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.

[0067] Playback status parameters are defined as follows: transmission delay, the time difference between the audio source and 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; and playback progress, the current position of the audio being played by the audio node, usually represented by a timestamp or the number of samples.

[0068] Meanwhile, the monitoring module collects the aforementioned playback status parameters at an extremely high frequency (e.g., multiple times per second) and sends this data to the central control system in real time. The system updates this data periodically (e.g., every few milliseconds) to ensure that all information is up-to-date.

[0069] S1222, based on the playback status, the time difference and phase difference corresponding to each audio node are determined by a delay compensation algorithm.

[0070] First, the system selects a reference audio node as a synchronization reference point, and the time difference and phase difference of all other audio nodes will be calculated relative to this reference point.

[0071] For time difference calculation: compare the transmission delay between each audio node and the reference node to calculate the time difference 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 difference.

[0072] For phase difference calculation: Based on the playback progress of each audio node, its phase difference relative to the reference node is calculated. This involves precise positioning of audio frames to ensure that the correct phase relationship is maintained even with small time differences. In addition, combined with the above calculation results, the system applies a specially designed delay compensation algorithm to comprehensively consider the effects of transmission delay, processing time, and playback progress, and accurately determine the time difference and phase difference of each audio node.

[0073] S1223 adjusts the playback start time and audio frame delay of each audio node based on the time difference and phase difference.

[0074] Among them, playback start time adjustment: For audio nodes with long transmission delays or processing times, the system will calculate an appropriate advance amount so that these nodes can start playing at the correct time, thereby maintaining synchronization with the baseline node; the system will dynamically adjust the playback start time of each audio node based on real-time monitoring data to ensure that synchronized playback can be maintained even when network conditions change or node load fluctuates.

[0075] Audio frame delay adjustment: For 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 with other nodes in time; as environmental conditions change (such as wind speed affecting the speed of sound propagation), the system will continuously re-evaluate and adjust the audio frame delay to maintain the best synchronization effect.

[0076] After each adjustment, the system will perform a synchronization verification to ensure that all audio nodes have achieved the expected synchronization effect. If any desynchronization is still found, the system will continue to fine-tune the relevant parameters until the optimal synchronization state is achieved.

[0077] By executing steps S1221 to S1223 as described above, the built-in real-time audio stream monitoring module continuously acquires the playback status of each speaker node. A latency compensation algorithm is used to accurately calculate the time and phase differences between nodes, and the playback start time and audio frame latency of each speaker node are dynamically adjusted accordingly. This method not only eliminates audio desynchronization issues caused by network transmission or hardware processing but also provides a seamless and consistent sound experience. In this way, whether in large outdoor events or much more complex coverage scenarios, users can enjoy perfectly synchronized, high-quality sound effects, greatly improving the overall stability of the audio system and the user experience.

[0078] In addition, the method in this embodiment further includes:

[0079] S1231 monitors the audio status of each audio node in real time.

[0080] Each speaker node integrates a high-precision status monitoring module, which can capture and record key status information of the speaker 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.

[0081] Status parameter definitions: Online status, whether each speaker node is online, including whether it is working properly, whether there is a fault, etc.; Location information, the precise location of each speaker node in physical space, which can be obtained through GPS or other positioning technologies; Number of speaker nodes, the total number of active speaker nodes in the current speaker system, including newly added or removed nodes.

[0082] Meanwhile, the status monitoring module collects the above status parameters at an extremely high frequency (e.g., multiple times per second) and sends these data to the central control system in real time. The central control system updates these data periodically (e.g., every few seconds) to ensure that all information is up-to-date and can respond promptly to any status changes.

[0083] S1232, when a change in the audio status is detected, obtain the latest audio node layout.

[0084] The central control system analyzes the status data received from each speaker node in real time. Once a change in the online status, location information, or number of speaker nodes is detected, the status change processing procedure is immediately triggered. The system automatically collects the latest speaker node layout information, including: the new location of each speaker node (if it has been moved); newly added or removed speaker nodes; and the relative positional relationships between the speaker nodes.

[0085] At the same time, the system will verify the newly acquired speaker node layout to ensure that all information is accurate. For example, it will check for duplicate location information or unreasonable node distribution; if any problems are found, the system will prompt the user to manually confirm or correct them.

[0086] S1233, the sound field modeling algorithm is started based on the latest speaker node layout, the sound field distribution is recalculated based on the latest speaker node layout, and the audio output parameters of each speaker node are adjusted based on the sound field distribution.

[0087] Based on the latest speaker node layout, the system selects the most suitable sound field modeling algorithm. Different scenarios may require different modeling methods; for example, a wide-area sound field model is used in open spaces, while a reflection and diffraction model is used in enclosed spaces. The latest speaker node layout information (including location, quantity, and relative relationships) and comprehensive environmental condition data (such as temperature, humidity, and wind speed) are passed as input parameters to the sound field modeling algorithm.

[0088] The sound field modeling algorithm calculates the sound distribution around each speaker node, such as sound wave propagation paths, reflection, diffraction, absorption, and other phenomena; and outputs a detailed sound field distribution map, showing the coverage area of ​​the speaker system, the spatial variation of key parameters (such as volume level), and other information.

[0089] Based on the information provided by the sound field distribution map, the system automatically adjusts the audio output parameters of each speaker node, taking into account the collaborative work between multiple speaker nodes to ensure the best performance of the overall sound system. This includes adjusting the distance, angle and relative position between speaker nodes to achieve uniform sound coverage and a consistent listening experience.

[0090] The system dynamically adjusts the audio output parameters of the speaker nodes based on the latest sound field distribution map, and continuously optimizes the adjustment results through real-time monitoring and user feedback. For example, if the volume in some areas is still insufficient, the system will further adjust the relevant parameters until the ideal effect is achieved.

[0091] By executing steps S1231 to S1233 as described above, the system automatically detects changes in the speaker nodes by monitoring their status in real time and obtains the latest speaker node layout upon detection. Based on this updated layout, the system initiates a sound field modeling algorithm to recalculate the sound field distribution and adjusts the audio output parameters of each speaker node accordingly. This method allows the sound system to quickly adapt to node changes, maintaining a consistent auditory experience and greatly enhancing the flexibility and intelligence of sound systems during outdoor activities. Users can enjoy the best sound experience after any layout adjustment.

[0092] In addition, the method also includes:

[0093] S1241 sets the corresponding sound focus position in the actual environment.

[0094] The system can provide a user-friendly mobile application or web platform that allows users to set the sound focus position through an intuitive interface. Users can select the specific focus position by clicking, dragging, or entering coordinates. Furthermore, the application incorporates augmented reality technology to display the sound nodes in the actual environment and the set sound focus position in real time on the user's mobile device screen. Users can visually adjust the focus position in the real environment to ensure the accuracy and rationality of each focus position.

[0095] After the user sets the position of each sound focus, the system will verify it to ensure that these positions are physically feasible and effective. Once the verification is successful, the system will store the focus position information in the central control system and prepare it for subsequent calculations.

[0096] S1242 calculates the corresponding audio output parameters based on the sound focus position using an acoustic modeling algorithm.

[0097] As mentioned earlier, different scenarios may require different modeling methods. For example, a wide-area sound field model may be used in open spaces, while a reflection and diffraction model may be used in enclosed spaces. The acoustic modeling algorithm is fed with user-defined sound focus locations, speaker node layouts, and comprehensive environmental conditions as input parameters. This includes the location, number, relative relationships, and environmental characteristics of each speaker node.

[0098] The acoustic modeling algorithm uses input parameters to perform complex physical simulations and calculates the optimal audio output parameters for each speaker node to ensure that the sound is concentrated and evenly covers each focal position. The algorithm outputs detailed audio output parameters, including volume, equalizer settings, and phase difference.

[0099] S1243: When the user adjusts or adds a new sound focus position, the audio output parameters of the speaker network are recalculated and optimized, and a new sound field distribution map is generated and displayed to the user in real time.

[0100] The system monitors in real time whether the user has adjusted the existing sound focus position or added a new focus position. Once a change is detected, a recalculation process is immediately triggered. The system verifies the newly added or adjusted focus position to ensure its rationality and effectiveness, for example, by checking for duplicate positions or unreasonable selections.

[0101] Based on the updated focus position information, the system restarts the acoustic modeling algorithm to recalculate the audio output parameters of the speaker network. This process is similar to the initial calculation, but takes into account the latest focus position and environmental changes. The system will dynamically adjust the audio output parameters of the speaker network according to the recalculation results and continuously optimize the adjustment results through real-time monitoring and user feedback. For example, if the volume in some areas is still insufficient, the system will further adjust the relevant parameters until the ideal effect is achieved.

[0102] Finally, based on the recalculated audio output parameters, the system generates a new sound field distribution map, showing the coverage area of ​​the sound system and the spatial variations of key parameters (such as volume level). This map not only helps users intuitively understand the effect of the current sound system setup but also provides a basis for subsequent adjustments.

[0103] By executing steps S1241 to S1243 as described above, users can set the sound focus positions in the actual environment. Based on these focus positions, the system calculates the optimal audio output parameters using acoustic modeling algorithms, ensuring that the sound is concentrated and evenly covers each focus point. When the user adjusts or adds a new sound focus position, the system recalculates and optimizes the audio output parameters of the speaker network and generates a new sound field distribution map in real time, displaying it to the user. This method not only provides highly personalized speaker configurations but also enhances the user's sense of control and participation, ensuring that every adjustment takes effect immediately, bringing a precise and consistent sound experience to listeners in a specific area. The system significantly improves the sound effects and user satisfaction of outdoor activities, achieving a high degree of unity between intelligence and flexibility.

[0104] In this embodiment of the application, the method further includes:

[0105] S12513: Obtain the load status of each audio node and retrieve the corresponding comprehensive condition data.

[0106] Each audio node is equipped with a lightweight monitoring agent program. This program is responsible for collecting the node's load information (such as CPU utilization, memory usage, and network bandwidth) as well as comprehensive environmental condition data. The monitoring agent program collects the above data at preset time intervals (such as every second or every minute) and transmits the data to the central control system through a secure and efficient communication protocol.

[0107] S1252 determines the priority of each audio node based on load conditions and comprehensive condition data.

[0108] Based on the characteristics of the audio system and the needs of actual application scenarios, a scientific and reasonable priority evaluation model is developed. The model considers factors including, but not limited to, the current load of each speaker node, its importance in the overall audio layout, and the physical characteristics of its location. Combining the latest load data and comprehensive condition data, the priority of each speaker node is dynamically calculated. High-priority speaker nodes may be those located in critical areas or undertaking more important audio output tasks.

[0109] Furthermore, the system incorporates a flexible priority adjustment mechanism that allows for automatic adjustment of speaker node priorities based on changes in actual conditions (such as newly added sound focus positions or changes in environmental factors). This mechanism helps ensure the efficient operation of the audio system and optimal resource utilization.

[0110] S1253, when a change in the load status or overall condition data of an audio node is detected, the audio stream parameters of the corresponding audio node are adjusted based on priority.

[0111] The system continuously monitors the load and overall condition data of the speaker nodes. Upon detecting any significant changes (such as a sudden increase in CPU usage of a speaker node or a sharp rise in the temperature of its area), it immediately triggers a parameter adjustment process. Based on the speaker node's priority, the system formulates corresponding audio stream parameter adjustment strategies. This may involve adjustments to multiple aspects, including volume, equalizer settings, and phase difference, with the aim of ensuring that the speaker nodes can continue to provide high-quality audio output under new conditions.

[0112] Once the adjustment strategy is confirmed, the system will automatically apply it to the relevant speaker nodes without manual intervention. Simultaneously, the system will record the adjustment process and results for subsequent analysis. Furthermore, the system will verify the effect of the adjusted speaker nodes to ensure that the adjustment achieves the intended purpose. If the desired effect is not achieved, the system will further fine-tune the relevant parameters until satisfactory results are obtained.

[0113] By executing steps S1241 to S1243 as described above, and by rationally allocating the priority of the audio nodes, the system can make optimal decisions under limited resources, ensuring that key audio nodes receive sufficient support, thereby maintaining the stability and reliability of the entire audio system. It can also enable the audio system to respond quickly to changes in internal and external conditions and maintain its best working state. By adjusting the audio stream parameters based on priority, the system not only improves its adaptability and flexibility, but also ensures that it can provide users with a consistent and high-quality audio experience even in complex and ever-changing environments.

[0114] In this embodiment of the application, the method involves determining the corresponding environmental region, so it further includes:

[0115] S1251 monitors the user's environment in real time and determines whether the current environment has changed.

[0116] The system utilizes technologies such as GPS, Bluetooth beacons, and Wi-Fi positioning to track the user's precise location in real time. This location information is continuously transmitted to the central control system via a mobile application or wearable device. The central control system then uses a predefined environmental area model, combined with sensor data and user location information, to intelligently identify the user's current location.

[0117] In addition, the system continuously monitors changes in the user's location and surrounding environmental conditions, setting reasonable thresholds to determine whether significant changes have occurred in the environment. For example, when temperature, humidity, or wind speed exceeds a certain range, a status change detection is triggered. The system compares and analyzes the latest sensor data with previously recorded data to determine if there are significant differences. If certain key parameters (such as increased temperature or humidity) exceed preset thresholds, the system considers the environment to have changed.

[0118] S1252, if a change occurs, determine the corresponding current environment region, and adjust the corresponding working mode and audio output parameters based on the current environment region.

[0119] When the system detects a change in the environmental area, it immediately initiates a new regional feature extraction process. This includes reassessing key parameters such as temperature, humidity, wind speed, and light intensity to determine the characteristics of the user's current new environmental area. The system then matches the extracted new regional features with a predefined environmental area model to find the environmental area category that best matches the current conditions. For example, it might change from a "shaded area" to a "direct sunlight area," or from a "wind tunnel area" to an "open area."

[0120] Once the new environmental area is determined, the system will generate a detailed area confirmation report and notify the user of the current environmental area and its characteristics through a mobile application or other interface. The user can understand the environmental changes through this notification and prepare for possible sound system adjustments.

[0121] Based on the newly identified environmental area, the system automatically selects the most suitable audio operating mode for that area. For example, in a windy area, a low-frequency enhancement mode might be selected to combat wind noise; in a shaded area, an energy-saving mode could be chosen to reduce energy consumption. The system dynamically adjusts audio output parameters based on the new environmental characteristics, including adjusting the distance, angle, and relative position between speaker nodes to achieve uniform sound coverage and a consistent listening experience.

[0122] By adjusting the speaker's operating mode and audio output parameters based on the new environmental zones, the system ensures optimal sound coverage and clarity even under changing environmental conditions. This approach not only enhances the user experience but also strengthens the speaker system's stability and flexibility, ensuring immediate response to every environmental change and providing users with the most suitable sound experience.

[0123] This application discloses an audio control system, referring to... Figure 2 The system includes, but is not limited to:

[0124] The environmental area determination module 200 integrates multiple high-precision sensors to acquire comprehensive environmental condition data in real time, including temperature, humidity, wind speed, and crowd density. Based on this comprehensive data, it determines the corresponding environmental area, which includes a shaded area, a direct sunlight area, and a wind tunnel area. When the temperature is below a first preset threshold, the wind speed is below a second preset threshold, and the crowd density is below a third preset threshold, it is determined to be a shaded area. When the temperature is above a fourth preset threshold, the wind speed is below a fifth preset threshold, and the crowd density is above a sixth preset threshold, it is determined to be a direct sunlight area. When the wind speed is above a seventh preset threshold and the temperature difference between the surrounding area and the surrounding area exceeds an eighth preset threshold, it is determined to be a wind tunnel area.

[0125] Application module 210 adjusts the working mode and audio output parameters of the speaker according to the environmental area, and is used to apply the working mode and audio output parameters to the speaker during use.

[0126] The audio setup adjustment module 220 generates a corresponding expected sound field distribution map based on the current audio setup plan during audio use, sends the expected sound field distribution map to the user, obtains corresponding user feedback data, and adjusts the corresponding audio setup plan based on the user feedback data; it also monitors changes in comprehensive condition data in real time and dynamically updates the division of environmental areas and audio parameter settings based on real-time data.

[0127] Furthermore, the system includes, but is not limited to:

[0128] The comprehensive condition data monitoring module is used to monitor the comprehensive condition data of the usage environment in real time and to evaluate the contextual characteristics of the current usage environment based on the comprehensive condition data.

[0129] The situation assessment module uses contextual features to determine whether an emergency is in progress. If so, it determines the severity of the situation and the corresponding outdoor conditions based on the contextual features.

[0130] The urgency and spread adjustment module adjusts the urgency and spread of the alarm based on the severity of the situation and outdoor conditions.

[0131] Furthermore, the system includes, but is not limited to:

[0132] The playback status acquisition module, based on the built-in real-time audio stream monitoring module, is used to continuously acquire the playback status of each speaker node. The playback status includes the transmission delay, processing time, and playback progress of the audio stream.

[0133] The time difference and phase difference determination module uses a delay compensation algorithm based on the playback status to determine the time difference and phase difference corresponding to each audio node.

[0134] The playback start time and audio frame delay adjustment module adjusts the playback start time and audio frame delay of each speaker node based on time difference and phase difference.

[0135] Furthermore, the system includes, but is not limited to:

[0136] The speaker status monitoring module is used to monitor the speaker status of each speaker node in real time. The speaker status includes online status, location information, and the number of speaker nodes.

[0137] The latest speaker node layout acquisition module is used to acquire the corresponding latest speaker node layout when a change in speaker status is detected.

[0138] The audio output parameter adjustment module starts the sound field modeling algorithm based on the latest speaker node layout, recalculates the sound field distribution based on the latest speaker node layout, and adjusts the audio output parameters of each speaker node based on the sound field distribution.

[0139] Furthermore, the system includes, but is not limited to:

[0140] The sound focus position setting module is used to set the corresponding sound focus position in the actual environment;

[0141] The audio output parameter determination module calculates the corresponding audio output parameters based on the sound focus position using an acoustic modeling algorithm. These audio output parameters include volume, equalizer settings, and phase difference.

[0142] The optimization module recalculates and optimizes the audio output parameters of the speaker network and generates a new sound field distribution map that is displayed to the user in real time when the user adjusts or adds a new sound focus position.

[0143] Furthermore, the system includes, but is not limited to:

[0144] The load acquisition module is used to acquire the load status of each audio node and retrieve the corresponding comprehensive condition data, including CPU utilization, memory usage, and network bandwidth.

[0145] The priority determination module uses load conditions and comprehensive condition data to determine the priority of each audio node.

[0146] The audio stream parameter adjustment module adjusts the audio stream parameters of the corresponding speaker node based on priority when it detects changes in the load or overall condition data of the speaker node.

[0147] Furthermore, the system includes, but is not limited to:

[0148] The environmental area monitoring module is used to monitor the user's environmental area in real time and determine whether the corresponding environmental area has changed.

[0149] The current environment region determination module is used to determine the corresponding current environment region if it changes, and adjust the corresponding working mode and audio output parameters based on the current environment region.

[0150] This application also discloses an audio control method, including a processor, wherein the processor runs a program of any one of the audio control methods described above.

[0151] This application also discloses a storage medium storing a program for the audio control method described in any one of the above embodiments.

[0152] Although embodiments of this application have been shown and described above, it is understood that the above embodiments are exemplary and should not be construed as limiting this application. Those skilled in the art can make changes, modifications, substitutions and variations to the above embodiments within the scope of this application.

Claims

1. A sound control method, characterized in that, include: By integrating multiple high-precision sensors, comprehensive environmental data for the use of the audio equipment is acquired in real time. This comprehensive environmental data includes temperature, humidity, wind speed, and crowd density. Based on the comprehensive condition data, the corresponding environmental areas are determined, including shaded areas, direct sunlight areas, and wind tunnel areas. When the temperature is below a first preset threshold, the wind speed is below a second preset threshold, and the population density is below a third preset threshold, it is determined to be a shaded area. When the temperature is above a fourth preset threshold, the wind speed is below a fifth preset threshold, and the population density is above a sixth preset threshold, it is determined to be a direct sunlight area. When the wind speed is above a seventh preset threshold and the temperature difference in the surrounding area exceeds an eighth preset threshold, it is determined to be a wind tunnel area. Adjust the working mode and audio output parameters of the speaker according to the environmental area, and apply the working mode and audio output parameters to the speaker during use. During the use of the audio system, a corresponding expected sound field distribution map is generated based on the current audio system layout scheme. The expected sound field distribution map is sent to the user to obtain corresponding user feedback data. The corresponding audio system layout scheme is adjusted based on the user feedback data. The changes in 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. The system monitors changes in the comprehensive condition data in real time and dynamically updates the environmental area division and audio parameter settings based on the real-time data. When the user adjusts or adds a new sound focus position, the system calculates the corresponding audio output parameters based on the sound focus position using an acoustic modeling algorithm. The audio output parameters include volume, equalizer settings, and phase difference. At the same time, the system recalculates and optimizes the audio output parameters of the speaker network and generates a new sound field distribution map, which is displayed to the user in real time.

2. The audio control method according to claim 1, characterized in that, The method also includes: Real-time monitoring of comprehensive environmental conditions data, and assessment of the contextual characteristics corresponding to the current usage environment based on the comprehensive environmental conditions data; Based on the aforementioned situational characteristics, it is determined whether an emergency situation exists. If so, the severity of the situation is determined, and the corresponding outdoor situation is determined based on the aforementioned situational characteristics. The urgency level and range of the alert are adjusted based on the severity of the situation and the outdoor conditions.

3. The audio control method according to claim 1, characterized in that, When multiple speaker nodes play audio simultaneously, the method also includes: The built-in real-time audio stream monitoring module continuously acquires the playback status of each speaker node, including the audio stream transmission delay, processing time, and playback progress. Based on the playback status, the time difference and phase difference corresponding to each audio node are determined by a delay compensation algorithm. The playback start time and audio frame delay of each audio node are adjusted based on the time difference and the phase difference.

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

5. The audio 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, including CPU utilization, memory usage, and network bandwidth. The priority of each audio node is determined based on the load conditions and the comprehensive condition data. When a change in the load status or overall condition data of an audio node is detected, the audio stream parameters of the corresponding audio node are adjusted based on the priority.

6. The audio control method according to claim 1, characterized in that, The method also includes: Real-time monitoring of the user's environment and determination of whether the current environment has changed; If a change occurs, the corresponding current environment region is determined, and the corresponding working mode and audio output parameters are adjusted based on the current environment region.

7. An audio control system for performing the audio control method as described in any one of claims 1-6, characterized in that, include: The environmental area determination module integrates multiple high-precision sensors to acquire comprehensive environmental condition data in real time, including temperature, humidity, wind speed, and crowd density. Based on the comprehensive condition data, the corresponding environmental areas are determined, including shaded areas, direct sunlight areas, and wind tunnel areas. When the temperature is below a first preset threshold, the wind speed is below a second preset threshold, and the population density is below a third preset threshold, it is determined as a shaded area. When the temperature is above a fourth preset threshold, the wind speed is below a fifth preset threshold, and the population density is above a sixth preset threshold, it is determined as a direct sunlight area. When the wind speed is above a seventh preset threshold and the temperature difference in the surrounding area exceeds an eighth preset threshold, it is determined as a wind tunnel area. The application module adjusts the working mode and audio output parameters of the speaker according to the environmental area, and applies the working mode and audio output parameters to the speaker during use. The audio setup adjustment module generates a corresponding expected sound field distribution map based on the current audio setup during audio use, sends the expected sound field distribution map to the user, obtains corresponding user feedback data, and adjusts the corresponding audio setup based on the user feedback data; it also monitors changes in the comprehensive condition data in real time and dynamically updates the environmental area division and audio parameter settings based on the real-time data.

8. A storage medium, characterized in that, The program stores the audio control method as described in any one of claims 1-6.

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