Sound equipment networking configuration method and device, electronic equipment and storage medium

Abstract

The invention relates to the technical field of computers, and discloses a sound equipment networking configuration method, which comprises the following steps of: establishing connection between a main loudspeaker and source equipment through a Bluetooth protocol, starting a multi-synchronization mode on the main loudspeaker to enable the main loudspeaker to enter a waiting state to receive a connection request from an auxiliary loudspeaker, at least one auxiliary loudspeaker enters a multi-synchronization mode and sends a connection request to the main loudspeaker; after the main loudspeaker receives the connection requests of the auxiliary loudspeakers, a unique equipment ID is distributed to each auxiliary loudspeaker, and a corresponding audio data transmission channel is established; audio signals received from the source device are synchronously broadcasted to all the connected auxiliary loudspeakers, and the main loudspeaker and the auxiliary loudspeakers are each provided with an LED indicator lamp used for displaying the power state, the charging state, the Bluetooth connection state and the multi-synchronization state. According to the invention, rapid and stable connection of a large number of synchronous playing sound equipment can be realized.

Description

Technical Field

[0001] The present application relates to the technical field of computers, and in particular to a method, device, electronic device and storage medium for configuring a network of audio equipment. Background Art

[0002] With the development of wireless audio technology, users' demand for portable audio equipment is growing. Although existing Bluetooth speakers can provide a good personal listening experience, in scenarios such as multi-person gatherings or outdoor activities, a single speaker often cannot meet the needs of large-scale coverage and multi-point simultaneous playback.

[0003] The solutions currently on the market usually have the following problems: slow connection speed: the connection process between multiple audio devices takes a long time, affecting the user experience; poor stability: disconnection is prone to occur in complex environments or long-distance transmission; complex settings: users need to manually perform complex settings to achieve synchronous playback of multiple devices; limited scalability: the number of devices that support synchronous playback is limited, which makes it difficult to meet large-scale application needs; insufficient user experience: the lack of an intuitive feedback mechanism makes users confused during operation.

[0004] Therefore, it is particularly important to develop a method that can simplify the operation process and support networking configuration. From the above, we can see that how to achieve fast and stable connection of a large number of audio devices playing simultaneously remains to be solved. Summary of the invention

[0005] In order to achieve fast and stable connection of a large number of audio devices playing simultaneously, the present application provides an audio device networking configuration method, device, electronic device and storage medium.

[0006] In a first aspect, the present application provides a method for configuring a network of audio equipment, which adopts the following technical solution: A method for configuring a network of audio equipment comprises: establishing a connection between a main speaker and a source device via a Bluetooth protocol, starting a multi-synchronous mode on the main speaker so that the main speaker enters a waiting state to receive a connection request from a secondary speaker, and at least one secondary speaker enters a multi-synchronous mode and sends a connection request to the main speaker; after the main speaker receives the connection request from the secondary speaker, assigns a unique device ID to each secondary speaker, records the device ID and its corresponding network parameters, and establishes a corresponding audio data transmission channel between the main speaker and the secondary speakers, wherein the main speaker adopts an adaptive audio stream control technology to dynamically adjust the size and transmission interval of the audio data packet according to the current network status; through the audio data transmission channel, synchronously broadcasting the audio signal received from the source device to all connected secondary speakers, and the main speaker and the secondary speakers are both equipped with LED indicator lights for displaying power status, charging status, Bluetooth connection status and multi-synchronous status.

[0007] By adopting the above technical solution, the speaker access and audio playback are synchronized. This method specially designs adaptive audio stream control technology to dynamically adjust the audio data packet size and transmission interval according to the real-time network conditions, ensuring stable data transmission even in complex or changing network environments. At the same time, by assigning a unique device ID to each secondary speaker and recording its network parameters, fast and stable connection configuration for a large number of synchronous playback devices is achieved. The LED indicator provides intuitive status feedback, helping users to easily understand the working status of each audio device, thereby simplifying the deployment and management process of large-scale audio systems. This method not only optimizes the connection efficiency and stability of the multi-speaker system, but also greatly improves the user experience, especially in scenarios where a large number of audio devices need to be quickly set up and adjusted for synchronous playback.

[0008] Optionally, the method also includes: obtaining key performance indicators of the audio data transmission channel, wherein the key performance indicators include signal strength, packet loss rate, delay time, bandwidth usage and error rate; retrieving a pre-set performance indicator interval threshold, matching the key performance indicator with the performance indicator interval threshold, and obtaining a corresponding matching result; determining a corresponding adjustment priority based on the matching result, and adjusting the key performance indicator based on the adjustment priority.

[0009] By adopting the above technical solutions, the key performance indicators of the audio data transmission channel, such as signal strength, packet loss rate, latency, bandwidth usage, and error rate, are continuously monitored, and these indicators are matched with the pre-set performance interval thresholds to determine the adjustment priority. This method ensures that the system can dynamically optimize network parameters based on the current transmission quality to maintain the best audio playback effect. Through the automated performance evaluation and adjustment mechanism, the system can proactively respond to network fluctuations, ensure stability and consistency of sound quality during synchronous playback of large-scale audio equipment, while reducing the need for manual intervention and improving the system's adaptability and user experience.

[0010] Optionally, the method also includes: when monitoring a new secondary speaker attempting to connect, obtaining the carrying capacity corresponding to the current network environment, wherein the carrying capacity includes network bandwidth, signal strength and data transmission rate, and obtaining the number of secondary speakers currently connected to the main speaker; based on the network carrying capacity and the number of secondary speakers, determining whether to accept the connection request from the at least one new secondary speaker; if accepted, establishing a connection with the at least one new secondary speaker, otherwise sending a rejection message to the at least one new secondary speaker, and providing suggestions for improving the network conditions to the user.

[0011] By adopting the above technical solution, when a new secondary speaker tries to connect, the system intelligently evaluates the carrying capacity of the current network environment and the number of connected devices, and dynamically decides whether to accept the new connection request. This mechanism ensures that the system will not suffer performance degradation or audio quality loss due to exceeding the network load, and at the same time, by providing specific network improvement suggestions, it helps users optimize network conditions to support the addition of more devices. This method not only maintains the stability and sound quality of the existing audio system, but also improves the scalability of the system, ensuring efficient and stable device connection management even in complex and changing network environments.

[0012] Optionally, the method also includes: obtaining physical position data corresponding to at least two secondary speakers connected to the main speaker, wherein the physical position data includes relative distance, direction, and potential obstacles in the environment; based on the physical position data, applying an optimization algorithm to calculate the audio signal distribution order and audio signal distribution strength of each secondary speaker; based on the audio signal distribution order and audio signal distribution strength, adjusting the audio signal sent to each secondary speaker; if it is detected that the position of any secondary speaker has changed, re-obtaining the physical position data corresponding to the secondary speaker to adjust the corresponding audio signal.

[0013] By adopting the above technical solution, the physical location data of the secondary speakers connected to the main speakers is automatically obtained and analyzed, including the relative distance, direction and obstacles in the environment, and the optimization algorithm is used to accurately calculate the order and strength of the audio signal distribution of each secondary speaker. This process ensures that the audio signal can be dynamically adjusted according to the actual layout to provide a more natural and immersive surround sound experience. More importantly, the system can automatically re-evaluate and adjust the audio settings when any changes in the position of the secondary speakers are detected to maintain the best sound effect, thereby significantly improving the flexibility and user experience of the multi-speaker system.

[0014] Optionally, the method also includes: before receiving a connection request from at least one slave speaker, obtaining the Bluetooth signal strength and interference situation corresponding to the execution environment of the main speaker; performing analysis based on the Bluetooth signal strength and the interference situation to obtain corresponding environmental analysis results; based on the environmental analysis results, determining whether the main speaker starts the multi-synchronization mode to manage the connection with the at least one slave speaker, and if it is not suitable to start the multi-synchronization mode, the main speaker provides the user with suggestions for improving environmental conditions.

[0015] By adopting the above technical solution, the surrounding Bluetooth signal strength and interference are pre-evaluated before receiving the connection request of the secondary speaker, ensuring that the main speaker makes an intelligent decision whether to start the multi-synchronous mode based on the actual environmental conditions. This preventive measure helps to avoid forcibly establishing a multi-synchronous connection when the network environment is poor, thereby preventing potential connection instability or audio quality degradation. By providing users with specific environmental improvement suggestions, the system not only improves the success rate and stability of the initial connection, but also optimizes the performance of the overall audio system to ensure that users get the best multi-speaker synchronization experience.

[0016] Optionally, the method also includes: after connecting the secondary speakers, triggering an initialization calibration process, and the main speaker sending a test audio signal to at least one secondary speaker; obtaining an actual response time of the at least one secondary speaker to the test audio signal; and determining a playback delay compensation value set by the main speaker for each secondary speaker based on the actual response time.

[0017] By adopting the above technical solution, the initialization calibration process is automatically triggered after the sub-speaker is connected, and the actual response time of each sub-speaker is accurately measured using the test audio signal, thereby setting the optimal playback delay compensation value for each sub-speaker. This calibration process ensures the highest level of audio synchronization accuracy between all speakers, and achieves a seamless, high-fidelity audio playback experience even in complex multi-speaker systems. Through the automated delay compensation mechanism, the system can effectively eliminate the time deviation caused by differences in the transmission path, providing a more consistent and immersive sound effect.

[0018] In a second aspect, the present application provides a device for configuring a network of audio equipment, which adopts the following technical solution: A device for configuring a network of audio equipment, comprising: A synchronous mode starting module, which establishes a connection between a main speaker and a source device through a Bluetooth protocol, and is used to start a multi-synchronous mode on the main speaker so that the main speaker enters a waiting state to receive a connection request from a secondary speaker, and at least one secondary speaker enters the multi-synchronous mode and sends a connection request to the main speaker; an audio data transmission channel establishment module, wherein after the main speaker receives the connection request from the secondary speaker, a unique device ID is assigned to each secondary speaker, and the device ID and its corresponding network parameters are recorded, so as to establish a corresponding audio data transmission channel between the main speaker and the secondary speaker, wherein the main speaker adopts adaptive audio stream control technology to dynamically adjust the size and transmission interval of the audio data packet according to the current network status; The synchronization module is used to synchronously broadcast the audio signal received from the source device to all connected secondary speakers through the audio data transmission channel. The main speaker and the secondary speakers are equipped with LED indicators for displaying the power status, charging status, Bluetooth connection status and multi-synchronization status.

[0019] In a third aspect, the present application provides an electronic device, which adopts the following technical solution: An electronic device comprises a processor, wherein a program of any one of the above-mentioned audio equipment networking configuration methods is run in the processor.

[0020] In a fourth aspect, the present application provides a storage medium, which adopts the following technical solution: A storage medium stores a program of any one of the above-mentioned audio equipment networking configuration methods.

[0021] In summary, the present application includes at least one of the following beneficial technical effects: 1. First, the main speaker performs an environmental assessment before receiving a connection request from the slave speaker to ensure that the network conditions are suitable for starting the multi-sync mode, and provides users with improvement suggestions to optimize the connection quality. Then, after the connection is established, the system automatically triggers the initialization calibration process, accurately measures the response time of each slave speaker by sending a test audio signal, and dynamically adjusts the playback delay compensation value to ensure that the audio synchronization accuracy between all speakers reaches the highest level. This intelligent decision-making and automated calibration mechanism not only simplifies the user's setup process, but also effectively improves the system's stability and sound quality consistency.

[0022] 2. In addition, the method uses adaptive audio flow control technology and continuous monitoring of key performance indicators to dynamically adjust the audio data packet size and transmission interval according to real-time network conditions, ensuring that the best data transmission stability can be maintained even in complex or changing network environments. At the same time, by assigning a unique device ID to each secondary speaker and recording its network parameters, the system can efficiently manage a large number of synchronous playback devices, provide intuitive status feedback, further simplify the deployment and management of large-scale audio systems, and ensure that users get a seamless, high-fidelity audio experience. BRIEF DESCRIPTION OF THE DRAWINGS

[0023] Figure 1 The present invention is a flowchart of a method for configuring a network of audio devices according to an exemplary embodiment.

[0024] Figure 2 The figure is a structural block diagram of a device for configuring audio equipment networking according to an exemplary embodiment. DETAILED DESCRIPTION

[0025] Embodiments of the present application are described in detail below, examples of which are shown in the accompanying drawings.

[0026] In the description of this specification, the description with reference to the 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 conjunction 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 embodiment or example. Moreover, the specific features, structures, materials, or characteristics described may be combined in any one or more embodiments or examples in a suitable manner.

[0027] The present application embodiment discloses a method for configuring a network of audio equipment, referring to Figure 1 ,include: S100, establish a connection between a main speaker and a source device via a Bluetooth protocol, start a multi-synchronous mode on the main speaker to put the main speaker into a waiting state to receive a connection request from a secondary speaker, and at least one secondary speaker enters the multi-synchronous mode and sends a connection request to the main speaker.

[0028] First, the user connects a main speaker to an audio source device (such as a smartphone, tablet, or computer) through the Bluetooth protocol. This process usually involves the user selecting the main speaker as the output device on the source device and completing the pairing. The Bluetooth module of the main speaker searches for available Bluetooth source devices and establishes a secure connection with them. Once the connection is successful, the main speaker can start receiving audio streams from the source device.

[0029] After the connection is completed, the user can start the multi-sync mode on the main speaker. At this time, the main speaker enters a special state and is ready to accept connection requests from other slave speakers. It should be noted that the main speaker activates its multi-sync function module and opens specific Bluetooth services to allow slave speakers to connect. It may also broadcast its presence information so that surrounding slave speakers can discover and try to connect.

[0030] Then, at least one slave speaker also needs to enter the multi-synchronous mode and send a connection request to the main speaker; after the slave speaker detects the presence of the main speaker, it initiates a connection request through the Bluetooth protocol, including its own basic information (such as model, current status, etc.).

[0031] S110, after receiving the connection request from the slave speaker, the main speaker allocates a unique device ID to each slave speaker, records the device ID and its corresponding network parameters, and establishes a corresponding audio data transmission channel between the main speaker and the slave speaker.

[0032] Among them, after the main speaker receives the connection request from the slave speaker, it assigns a unique device ID to each slave speaker and records the relevant network parameters to ensure that each slave speaker has a clear identity in the network; the main speaker saves all necessary information about each slave speaker, such as MAC address, Bluetooth version, signal strength, etc.

[0033] In addition, the master speaker generates a globally unique identifier or creates a unique ID based on existing standards; the master speaker stores network parameters of each slave speaker, including but not limited to RSSI, channel information, connection quality, etc.

[0034] A dedicated audio data transmission channel is established between the main speaker and each slave speaker. Using adaptive audio stream control technology, the main speaker dynamically adjusts the size and transmission interval of the audio data packet according to the current network conditions to optimize bandwidth usage and reduce latency. This achieves efficient and stable audio data transmission, maintaining good sound quality even when network conditions change.

[0035] S120, through the audio data transmission channel, synchronously broadcasts the audio signal received from the source device to all connected slave speakers. Both the main speaker and the slave speakers are equipped with LED indicators to display the power status, charging status, Bluetooth connection status and multi-sync status.

[0036] The main speaker broadcasts the audio signal received from the source device to all connected slave speakers synchronously through the established audio data transmission channel, thereby ensuring that all slave speakers play the same audio content at the same time and provide a unified sound experience. It should be noted here that the main speaker processes the audio stream and ensures that all slave speakers play synchronously according to a preset schedule or based on a real-time feedback mechanism.

[0037] In addition, both the main speaker and the sub-speaker are equipped with LED indicators to intuitively display the power status, charging status, Bluetooth connection status, and multi-sync status. LEDs of different colors or flashing modes indicate different working states, helping users quickly understand the current status of the device. The color and flashing frequency of the LED indicator are controlled by an internal state machine to reflect the current status of the device.

[0038] Through the above-mentioned execution steps, the audio equipment networking configuration method ensures that a large number of audio devices playing synchronously can be connected quickly and stably through a series of carefully designed technical steps. It not only simplifies the user's setup process and improves the flexibility and scalability of the audio system, but also ensures the best data transmission stability and audio quality even in complex or changing network environments through intelligent decision-making, automatic calibration and dynamic optimization mechanisms. In addition, the intuitive status feedback provided by the LED indicator further simplifies user operations and ensures efficient deployment and management of large-scale audio systems, thereby providing users with a consistent and immersive sound experience. And this method is particularly suitable for scenarios where a large number of audio devices need to be quickly set up and adjusted for synchronous playback, such as home theaters, commercial display venues, outdoor activities and other occasions, which significantly improves the user experience and the practicality of the system.

[0039] When the multi-synchronous mode is started on the main speaker to put the main speaker into a waiting state, it is necessary to consider the current environment to determine whether it is suitable to connect to the secondary speaker, so the method also includes: S101, before receiving a connection request of at least one slave speaker, obtaining Bluetooth signal strength and interference conditions corresponding to an execution environment of a master speaker.

[0040] Among them, before receiving the connection request from the slave speaker, the main speaker actively performs a comprehensive environmental assessment. The main speaker uses its Bluetooth module and other wireless communication interfaces (such as Wi-Fi, RFID, etc.) to collect data on Bluetooth signal strength and interference in the surrounding environment.

[0041] Specifically, the main speaker starts the scanning function, detects and records the signal strength of all nearby Bluetooth devices, and analyzes possible interference sources in the environment, including but not limited to other Bluetooth devices, Wi-Fi routers, microwave ovens, etc. These devices may interfere with the Bluetooth signal. The main speaker can also evaluate the usage of the current frequency band by measuring channel congestion to determine whether there is a potential transmission conflict.

[0042] S102, performing analysis based on Bluetooth signal strength and interference conditions to obtain corresponding environmental analysis results.

[0043] The collected Bluetooth signal strength and interference conditions are sent to the analysis algorithm built into the main speaker for processing. The algorithm calculates a comprehensive environmental score or evaluation result based on factors such as Bluetooth signal strength, the number and strength of interference sources, and channel congestion.

[0044] Specifically, machine learning or a preset rule base is used to analyze the data to determine whether the current environment is suitable for starting the multi-synchronous mode. For example, if the RSSI values ​​of most Bluetooth devices are within a reasonable range and there are no obvious interference sources, the environment is considered to be good. If there is a strong interference source or multiple devices work on the same channel, which may cause unstable data transmission, the environment may not be suitable for starting the multi-synchronous mode.

[0045] Finally, after the analysis is completed, the system generates a detailed environmental analysis report, which contains key information such as the specific value of Bluetooth signal strength, the number and strength of interference sources, channel congestion, etc. The environmental analysis results can be quantified into specific indicators, such as "good", "average" or "poor", or expressed as a percentage or score.

[0046] S103, judging whether the main speaker starts the multi-synchronous mode to manage the connection with at least one secondary speaker according to the result of the environmental analysis, and if it is not suitable to start the multi-synchronous mode, the main speaker provides the user with suggestions for improving the environmental conditions.

[0047] Among them, according to the results of environmental analysis, the main speaker decides whether to start the multi-synchronous mode. If the environmental score reaches the preset threshold, the multi-synchronous mode is allowed to start; otherwise, the system will choose not to start. It should be pointed out here that the preset threshold can be flexibly adjusted according to different application scenarios. For example, home environments and commercial exhibition venues may have different standards. The main speaker can make internal logical judgments or interact with the cloud server to obtain the latest configuration suggestions.

[0048] If the environmental analysis results show that it is not suitable to start the multi-sync mode, the main speaker will not accept the connection request from the secondary speaker, and a rejection message can be sent to the user to explain the reason and provide specific improvement suggestions. The rejection message can be conveyed to the user through a display, mobile application or voice prompt, and the improvement suggestions can include specific measures such as moving the location of certain devices, turning off interference sources, and adjusting the settings of Wi-Fi routers.

[0049] Based on the execution steps S101 to S103 above, by pre-evaluating the surrounding Bluetooth signal strength and interference before receiving the connection request of the secondary speaker, the main speaker can make an intelligent decision whether to start the multi-synchronous mode based on the actual environmental conditions. This preventive measure helps to avoid forcibly establishing multi-synchronous connections when the network environment is not good, thereby preventing potential connection instability or audio quality degradation. By providing users with specific environmental improvement suggestions, the system not only improves the success rate and stability of the initial connection, but also optimizes the performance of the overall audio system to ensure that users get the best multi-speaker synchronization experience. This method is particularly suitable for scenarios where a large number of audio devices need to be quickly deployed and adjusted for synchronous playback, which significantly improves the reliability of the system and user satisfaction.

[0050] After the connection is completed, the audio data transmission channel needs to be monitored, so the method also includes: S1211, obtaining key performance indicators of the audio data transmission channel.

[0051] The system periodically or in real time monitors the audio data transmission channel between the main speaker and the auxiliary speaker, and collects a series of key performance indicators, including but not limited to signal strength, packet loss rate, delay time, bandwidth usage and error rate.

[0052] Specifically, signal strength: evaluates the quality of the Bluetooth connection by measuring RSSI, with higher RSSI values ​​indicating stronger signal quality; packet loss rate: counts the proportion of packets that failed to be successfully transmitted within a certain period of time. A lower packet loss rate means a more stable connection; latency: measures the time difference from sending data to receiving confirmation. Low latency is critical for synchronous audio playback; bandwidth usage: monitors the amount of bandwidth currently used to ensure that there are enough resources to support high-quality audio streaming; error rate: records the number of errors that occur during transmission, such as verification failures, to evaluate transmission reliability.

[0053] S1212, retrieve the preset performance indicator interval threshold, match the key performance indicator with the performance indicator interval threshold, and obtain the corresponding matching result.

[0054] The main speaker accesses its built-in database or cloud server to obtain pre-set performance indicator range thresholds; these thresholds define the optimal range and acceptable range of each performance indicator.

[0055] It should be noted here that the preset performance indicator interval thresholds can be adjusted according to different application scenarios. For example, home environments and commercial display venues may have different standards. Each performance indicator has an ideal range and a warning range to distinguish normal operation from potential problems. For example, the ideal range of signal strength may be -50dBm to -70dBm, while the warning range may be below -80dBm.

[0056] The actual measured key performance indicators are compared with the preset interval thresholds to determine the matching results of each indicator. The matching results can be "normal", "warning" or "error", depending on which range the actual value falls into.

[0057] Specifically, an algorithm is used to automatically compare actual values ​​with thresholds and generate a detailed matching report. If a performance indicator exceeds the warning range, it is marked as "warning"; if it exceeds the error range, it is marked as "error". For multiple performance indicators, the system will comprehensively consider all matching results to form an overall evaluation conclusion. By identifying which performance indicators deviate from the expected range, potential problem points can be located.

[0058] S1213: Determine a corresponding adjustment priority based on the matching result, and adjust the key performance indicator based on the adjustment priority.

[0059] Among them, according to the matching result, the system assigns an adjustment priority to each performance indicator, and the adjustment priority reflects the impact of the performance indicator on the overall system performance.

[0060] Key performance indicators are divided into high priority, medium priority and low priority. For example, if both latency and packet loss rate are in "warning" state, but latency has a greater impact, it will be given a higher priority. The system can also dynamically adjust the priority based on historical data analysis to adapt to different usage scenarios.

[0061] In addition, the system takes corresponding measures to optimize key performance indicators based on the determined adjustment priorities, which may involve operations such as adjusting the audio data packet size, changing the transmission interval, and switching channels.

[0062] Specific details include: Adjust the audio packet size: Reducing the packet size can reduce the risk of a single transmission failure, especially when the signal strength is weak; Change the transmission interval: Increasing the transmission interval can reduce bandwidth usage and reduce congestion, which is suitable for situations with high bandwidth occupancy; Switch channels: If the current channel is detected to be crowded or has a strong source of interference, the system can choose to switch to a less used channel; Other optimization measures: such as enabling QoS settings to prioritize audio traffic; or adjusting power output to enhance signal coverage.

[0063] Based on the execution steps from S1211 to S1213 above, by continuously monitoring the key performance indicators of the audio data transmission channel and matching them with the preset performance indicator interval thresholds, intelligent evaluation and dynamic adjustment of the audio system network status are achieved. This mechanism ensures that the system can dynamically optimize network parameters based on the current transmission quality to maintain the best audio playback effect. Through the automated performance evaluation and adjustment mechanism, the system can proactively respond to network fluctuations to ensure stability and consistency of sound quality during synchronous playback of large-scale audio equipment, while reducing the need for manual intervention and improving the system's adaptive capabilities and user experience. This method is particularly suitable for complex and changeable network environments, significantly improving the reliability of the audio system and user satisfaction.

[0064] When monitoring for a new secondary speaker attempting to connect, the method further includes: S1221, obtaining the carrying capacity corresponding to the current network environment, wherein the carrying capacity includes network bandwidth, signal strength and data transmission rate, and obtaining the number of secondary speakers currently connected to the primary speaker.

[0065] The main speaker uses a Bluetooth module or other wireless communication interface (such as Wi-Fi) to perform periodic scanning to identify the presence of new devices. Whenever a slave speaker is detected sending a connection request, the main speaker records the timestamp of the request and the basic information of the slave speaker (such as MAC address, model, etc.).

[0066] When a new secondary speaker is detected trying to connect, the primary speaker immediately evaluates the carrying capacity of the current network environment, including but not limited to network bandwidth, signal strength, and data transmission rate.

[0067] Specifically, network bandwidth: by measuring the ratio of the currently used bandwidth to the available bandwidth, evaluate whether there is enough bandwidth to support new connections; signal strength: measure the RSSI of existing connections to determine whether the signal quality is strong enough; data transmission rate: check the current data transmission rate to ensure that it can support more devices without reducing performance.

[0068] Furthermore, the main speaker queries its internal database or connection management module to obtain information about all connected slave speakers. The statistical results include not only the number but also the working status of each slave speaker (such as online, offline, playing, etc.).

[0069] S1222: Determine whether to accept a connection request from at least one new secondary speaker based on the network carrying capacity and the number of secondary speakers.

[0070] Among them, based on the previously collected data, the main speaker makes a comprehensive evaluation and decides whether to accept the connection request from the new secondary speaker. If the current network carrying capacity is sufficient and the number of connected secondary speakers has not reached the preset upper limit, the new connection is accepted; otherwise, it is rejected.

[0071] The system can flexibly adjust the threshold according to actual conditions. For example, more connections may be allowed in a home environment, while they may be more stringent in a commercial display venue. If multiple new secondary speakers request connection at the same time, the system can selectively accept the connection based on priority rules (such as user-specified priority or automatically evaluated priority).

[0072] S1223: If accepted, establish a connection with at least one new secondary speaker; otherwise, send a rejection message to at least one new secondary speaker and provide the user with suggestions for improving the network condition.

[0073] If the evaluation result shows that the connection request of the new slave speaker can be accepted, the main speaker will formally establish the connection. The main speaker sends a confirmation message to the new slave speaker to complete the pairing process, update the internal database, assign a unique device ID to the new slave speaker, record relevant network parameters, and establish a dedicated audio data transmission channel to ensure that the new slave speaker can synchronously receive the audio stream.

[0074] If the evaluation results show that it is not suitable to accept the connection request of the new slave speaker, the main speaker will send a rejection message to the new slave speaker and provide the user with specific improvement suggestions to help optimize network conditions.

[0075] In addition, rejection messages can be conveyed to users through display screens, mobile applications or voice prompts. Improvement suggestions can include specific measures such as moving the location of certain devices, turning off interference sources, and adjusting the settings of Wi-Fi routers. The system can also provide detailed reports explaining why the current network environment is not suitable for adding more devices.

[0076] By intelligently evaluating the current network environment's carrying capacity and the number of connected devices when a new secondary speaker attempts to connect, it dynamically decides whether to accept a new connection request. This mechanism ensures that the system does not suffer from performance degradation or audio quality loss due to exceeding the network load, while helping users optimize network conditions to support the addition of more devices by providing specific network improvement suggestions. This approach not only maintains the stability and sound quality of the existing audio system, but also improves the scalability of the system, ensuring efficient and stable device connection management even in complex and changing network environments. This method is particularly suitable for scenarios where a large number of audio devices need to be quickly deployed and adjusted for synchronous playback, significantly improving system reliability and user satisfaction.

[0077] Furthermore, the method also includes: S1231, obtaining physical position data corresponding to at least two secondary speakers connected to the main speaker.

[0078] The system automatically responds to the user's request and starts collecting physical location data of at least two secondary speakers connected to the main speaker. The physical location data includes but is not limited to relative distance, direction, and potential obstacles in the environment.

[0079] Specifically, relative distance: use Bluetooth ranging, Wi-Fi RTT, ultrasonic sensors and other technologies to measure the distance between each secondary speaker and the main speaker; direction: determine the direction of the secondary speaker relative to the main speaker through a multi-antenna array (such as MIMO technology) or triangulation; potential obstacles: use the built-in microphone array for sound reflection analysis, or use external cameras, infrared sensors, etc. to detect possible obstacles in the environment.

[0080] S1232: Based on the physical location data, an optimization algorithm is applied to calculate the audio signal distribution order and audio signal distribution strength of each secondary speaker.

[0081] Among them, based on the collected physical location data, the main speaker applies an optimization algorithm to calculate the audio signal distribution order and strength of each secondary speaker. The optimization algorithm can be a rule-based static algorithm or a machine learning model. These algorithms take into account multiple factors, such as distance, direction, obstacles, room acoustic characteristics, etc.

[0082] According to the position relationship of the sub-speakers, the time difference of the arrival of the audio signal is determined to achieve the best surround sound effect, and the volume of each sub-speaker is adjusted to ensure that the sound output by all speakers can be harmoniously integrated in the space to avoid excessive or weak sound in certain areas.

[0083] S1233: Adjust the audio signal sent to each secondary speaker based on the audio signal distribution order and the audio signal distribution strength.

[0084] According to the calculated audio signal distribution order and strength, the main speaker adjusts the audio signal sent to each secondary speaker.

[0085] Specific details include the following: Time calibration: for each secondary speaker, set an appropriate playback delay to ensure that the audio signal can reach each location in the predetermined order; Volume adjustment: adjust the volume of each secondary speaker according to the calculation results to ensure the consistency and balance of the overall sound effect; Dynamic adjustment: the system can dynamically adjust the audio signal parameters according to real-time feedback (such as user operation or environmental changes) to maintain the best sound effect. After the adjustment is completed, the system can send a test audio signal for verification to ensure that the adjusted effect meets expectations.

[0086] S1234: If it is detected that the position of any secondary speaker has changed, re-acquire the physical position data corresponding to the secondary speaker to adjust the corresponding audio signal.

[0087] Among them, the main speaker continuously monitors the position changes of the connected secondary speakers. Once the position change of any secondary speaker is detected, its physical position data is immediately re-acquired and the audio signal is adjusted accordingly.

[0088] Continuous Bluetooth ranging, Wi-Fi RTT or other sensor data can be used to track the position of the secondary speaker in real time. When the position change is detected to exceed the preset threshold, the process of re-acquiring the position data is triggered; the optimization algorithm is applied again to recalculate the audio signal distribution order and strength, and adjust the audio signal sent to each secondary speaker accordingly. If the position change significantly affects the sound effect, the system can inform the user through LED indicators, mobile applications or voice prompts, and provide further guidance.

[0089] By automatically acquiring and analyzing the physical location data of the sub-speakers connected to the main speakers, and using an optimization algorithm to accurately calculate the order and strength of the audio signal distribution of each sub-speaker, dynamic adjustment of the audio signal is achieved. This process not only ensures that the audio signal can be dynamically adjusted according to the actual layout to provide a more natural and immersive surround sound experience, but also automatically re-evaluates and adjusts the audio settings when any changes in the position of the sub-speakers are detected to maintain the best sound effect. This method significantly improves the flexibility and user experience of multi-speaker systems, and is particularly suitable for scenarios where the layout of audio equipment needs to be adjusted frequently or there are many changes in the environment, such as home theaters, commercial display venues, or outdoor activities.

[0090] The method also includes: S1241, after the secondary speaker is connected, triggering an initialization calibration process, the main speaker sends a test audio signal to at least one secondary speaker.

[0091] Among them, after successfully connecting the slave speaker, the main speaker automatically triggers an initialization calibration process. This process is designed to ensure the highest level of audio synchronization accuracy between all slave speakers.

[0092] Specifically, the calibration process can be triggered each time a new slave speaker is added, after the system is restarted, or when manually started by the user. The main speaker pauses the current audio playback, enters the calibration mode, and notifies all connected slave speakers to prepare to receive the test audio signal. The main speaker records the current timestamp as the basis for subsequent calculation of the response time.

[0093] The main speaker sends a series of carefully designed test audio signals to at least one secondary speaker. The test audio signals can be tones of specific frequencies, pulse trains, or other forms of sound samples for accurately measuring the characteristics on the transmission path.

[0094] It should be pointed out here that the test audio signal should have clear time characteristics, such as short pulses or sine waves of a specific frequency, in order to accurately measure the transmission delay. In order to comprehensively evaluate the response at different frequencies, test signals in multiple frequency bands can be sent.

[0095] If there are multiple slave speakers, test signals can be sent in groups or one by one to avoid signal overlap affecting the measurement results. Each test audio signal is accompanied by a unique synchronization mark to facilitate the slave speaker to identify and send back a confirmation signal.

[0096] S1242: Obtain an actual response time of at least one secondary speaker to a test audio signal.

[0097] After receiving the test audio signal, each slave speaker immediately sends a confirmation signal back to the main speaker, and the main speaker records the time difference between sending the test audio signal and receiving the confirmation signal, that is, the actual response time.

[0098] Among them, a high-precision timer (such as nanosecond level) is used to record the time difference to ensure the accuracy of the measurement; to reduce accidental errors, multiple measurements can be made and the average value can be taken. At the same time, the impact of environmental factors such as temperature and humidity on the signal propagation speed is considered, and appropriate compensation is made. The response time of each sub-speaker is recorded in detail to form a database for subsequent analysis and adjustment.

[0099] S1243: Determine a playback delay compensation value set by the main speaker for each secondary speaker based on the actual response time.

[0100] Among them, based on the measured actual response time, the main speaker sets an appropriate playback delay compensation value for each slave speaker. A built-in algorithm is used to calculate the delay compensation value required for each slave speaker based on the response time. For example, the slave speaker with the longest response time does not require additional delay, while other slave speakers need to increase the delay accordingly; and the compensation value can be dynamically adjusted based on real-time feedback to ensure that the best synchronization effect can be maintained in different environments. The calculated delay compensation value is stored in the configuration file of the main speaker and immediately applied to the audio playback process. After completing the settings, you can send a test audio signal again for verification to ensure that the adjusted effect meets expectations.

[0101] By automatically triggering the initialization calibration process after connecting the sub-speaker, the actual response time of each sub-speaker is accurately measured using the test audio signal, and the optimal playback delay compensation value is set for each sub-speaker accordingly. This calibration process ensures the highest level of audio synchronization accuracy between all speakers, and a seamless, high-fidelity audio playback experience can be achieved even in complex multi-speaker systems. Through the automated delay compensation mechanism, the system can effectively eliminate the time deviation caused by differences in transmission paths, provide more consistent and immersive sound effects, and significantly improve the flexibility and user experience of multi-speaker systems.

[0102] Scenario Description The home consists of three main spaces: the living room, dining room, and study. We want to install audio equipment in these three areas so that we can play the same music in different rooms simultaneously or control the sound source of each room independently as needed. To achieve this goal, we chose a main and sub-speaker system that supports the Bluetooth protocol and has advanced automatic environmental assessment, intelligent connection management, and dynamic performance optimization functions.

[0103] Case Implementation First, place the main speaker in the living room and pair it with your smartphone via Bluetooth. After pairing, activate the multi-sync mode on the main speaker. At this time, the main speaker automatically starts scanning the surrounding wireless environment, evaluating the Bluetooth signal strength and possible interference. Through the built-in algorithm analysis, the main speaker confirms that the current environment is suitable for starting the multi-sync mode and is ready to receive connection requests from other slave speakers.

[0104] Subsequently, two slave speakers were placed in the dining room and the study, and their multi-sync mode was turned on. The two slave speakers quickly detected the presence of the master speakers and sent connection requests. After receiving these requests, the master speakers assigned a unique device ID to each slave speaker and established a stable audio data transmission channel; the system also performed an initial calibration process, sending test audio signals to measure the actual response time of each slave speaker, and set appropriate playback delay compensation values ​​accordingly to ensure the highest level of audio synchronization accuracy between all speakers.

[0105] When playing music, the main speaker broadcasts the audio stream to all connected slave speakers through the established audio data transmission channel, filling the whole home with a harmonious and consistent music atmosphere. In addition, every time a new slave speaker tries to join the network, the main speaker will re-evaluate the current network carrying capacity and the number of connected devices to decide whether to accept the new device, in order to ensure system stability and audio quality.

[0106] To further enhance the user experience, the system can automatically adjust the order and strength of audio signal distribution according to the physical position of the secondary speaker to achieve the best surround sound effect. Even if the user changes the position of a speaker, the system can monitor the change in real time and adjust the audio settings accordingly to maintain the best sound effect; in the end, the home audio system not only provides a high-quality, synchronized audio experience, but its intelligent design makes system maintenance simple and intuitive, and the status feedback of the LED indicator helps to understand the working status of the device at any time, so that you can easily enjoy the immersive world of music.

[0107] This case demonstrates the flexibility and efficiency of audio equipment networking configuration methods in actual application scenarios, and also emphasizes the importance of technology in improving user experience.

[0108] The present application embodiment discloses a device for configuring a network of audio equipment, referring to Figure 2 , the devices include but are not limited to: The synchronous mode starting module 200 establishes a connection between a main speaker and a source device through a Bluetooth protocol, and is used to start a multi-synchronous mode on the main speaker so that the main speaker enters a waiting state to receive a connection request from a secondary speaker, and at least one secondary speaker enters the multi-synchronous mode and sends a connection request to the main speaker; The audio data transmission channel establishing module 210, after the main speaker receives the connection request of the secondary speaker, assigns a unique device ID to each secondary speaker, and records the device ID and its corresponding network parameters, so as to establish a corresponding audio data transmission channel between the main speaker and the secondary speaker, wherein the main speaker adopts adaptive audio stream control technology to dynamically adjust the size and transmission interval of the audio data packet according to the current network status; The synchronization module 220 is used to synchronously broadcast the audio signal received from the source device to all connected secondary speakers through the audio data transmission channel. The main speaker and the secondary speakers are equipped with LED indicator lights for displaying power status, charging status, Bluetooth connection status and multi-synchronization status.

[0109] Furthermore, the device includes but is not limited to: A key performance indicator acquisition module, used to acquire key performance indicators of the audio data transmission channel, wherein the key performance indicators include signal strength, packet loss rate, delay time, bandwidth usage and error rate; A matching module, which retrieves a preset performance indicator interval threshold value, and is used to match the key performance indicator with the performance indicator interval threshold value to obtain a corresponding matching result; The key performance indicator adjustment module determines a corresponding adjustment priority based on the matching result, and adjusts the key performance indicator based on the adjustment priority.

[0110] Furthermore, the device includes but is not limited to: A bearing capacity acquisition module, when monitoring a new secondary speaker trying to connect, is used to obtain the bearing capacity corresponding to the current network environment, wherein the bearing capacity includes network bandwidth, signal strength and data transmission rate, and obtain the number of secondary speakers currently connected to the main speaker; a determination module, configured to determine whether to accept a connection request from the at least one new secondary speaker based on the network carrying capacity and the number of secondary speakers; If accepted, a connection with the at least one new secondary speaker is established; otherwise, a rejection message is sent to the at least one new secondary speaker, and suggestions for improving the network condition are provided to the user.

[0111] Furthermore, the device includes but is not limited to: A physical location data acquisition module, used to acquire physical location data corresponding to at least two secondary speakers connected to the main speaker, wherein the physical location data includes relative distance, direction, and potential obstacles in the environment; An audio signal calculation module, based on the physical position data, applies an optimization algorithm to calculate the audio signal distribution order and audio signal distribution strength of each secondary speaker; an audio signal adjustment module, used for adjusting the audio signal sent to each secondary speaker based on the audio signal distribution order and the audio signal distribution strength; If it is detected that the position of any sub-speaker changes, the physical position data corresponding to the sub-speaker is re-acquired to adjust the corresponding audio signal.

[0112] Furthermore, the device includes but is not limited to: A Bluetooth signal strength acquisition module, used for acquiring Bluetooth signal strength and interference conditions corresponding to the execution environment of the main speaker before receiving a connection request of at least one secondary speaker; An environment analysis result acquisition module is used to acquire corresponding environment analysis results based on the analysis of the Bluetooth signal strength and the interference situation; The synchronous start judgment module is used to judge whether the main speaker starts the multi-synchronous mode to manage the connection with the at least one secondary speaker according to the environmental analysis result. If it is not suitable to start the multi-synchronous mode, the main speaker provides suggestions for improving the environmental conditions to the user.

[0113] Furthermore, the device includes but is not limited to: A trigger module, after connecting to the secondary speaker, is used to trigger the initialization calibration process, and the main speaker sends a test audio signal to at least one secondary speaker; An actual response time acquisition module, used to acquire an actual response time of the at least one secondary speaker to the test audio signal; The playback delay compensation value determination module is used to determine the playback delay compensation value set by the main speaker for each secondary speaker based on the actual response time.

[0114] An embodiment of the present application further discloses an electronic device, including a processor, wherein a program of any one of the above-mentioned audio device networking configuration methods is running on the processor.

[0115] The embodiment of the present application further discloses a storage medium storing a program of any one of the above-mentioned audio equipment networking configuration methods.

[0116] Although the embodiments of the present application have been shown and described above, it can be understood that the above embodiments are exemplary and cannot be understood as limitations on the present application. Ordinary technicians in this field can change, modify, replace and modify the above embodiments within the scope of the present application.

Claims

1. A method for configuring a network of audio equipment, characterized in that: include: A main speaker is connected to a source device via a Bluetooth protocol, a multi-synchronous mode is started on the main speaker so that the main speaker enters a waiting state to receive a connection request from a secondary speaker, and at least one secondary speaker enters the multi-synchronous mode and sends a connection request to the main speaker; After receiving the connection request from the slave speaker, the main speaker allocates a unique device ID to each slave speaker, records the device ID and its corresponding network parameters, and establishes a corresponding audio data transmission channel between the main speaker and the slave speaker, wherein the main speaker adopts adaptive audio stream control technology to dynamically adjust the size and transmission interval of the audio data packet according to the current network status; The audio signal received from the source device is synchronously broadcast to all connected slave speakers through the audio data transmission channel. The main speaker and slave speakers are equipped with LED indicators for displaying power status, charging status, Bluetooth connection status and multi-synchronization status.

2. The audio equipment networking configuration method according to claim 1, characterized in that: The method also includes: Acquire key performance indicators of the audio data transmission channel, wherein the key performance indicators include signal strength, packet loss rate, delay time, bandwidth usage, and error rate; Retrieving a preset performance indicator interval threshold, matching the key performance indicator with the performance indicator interval threshold, and obtaining a corresponding matching result; A corresponding adjustment priority is determined based on the matching result, and the key performance indicator is adjusted based on the adjustment priority.

3. The audio equipment networking configuration method according to claim 2, characterized in that: The method also includes: When monitoring a new secondary speaker trying to connect, obtaining the carrying capacity corresponding to the current network environment, wherein the carrying capacity includes network bandwidth, signal strength and data transmission rate, and obtaining the number of secondary speakers currently connected to the primary speaker; Determining whether to accept a connection request from the at least one new secondary speaker based on the network carrying capacity and the number of secondary speakers; If accepted, a connection with the at least one new secondary speaker is established; otherwise, a rejection message is sent to the at least one new secondary speaker, and suggestions for improving the network condition are provided to the user.

4. The audio equipment networking configuration method according to claim 1, characterized in that: The method also includes: Acquire physical location data corresponding to at least two secondary speakers connected to the primary speaker, wherein the physical location data includes relative distance, direction, and potential obstacles in the environment; Based on the physical location data, an optimization algorithm is applied to calculate the order and strength of audio signal distribution of each secondary speaker; adjusting the audio signal sent to each secondary speaker based on the audio signal distribution order and the audio signal distribution strength; If it is detected that the position of any sub-speaker changes, the physical position data corresponding to the sub-speaker is re-acquired to adjust the corresponding audio signal.

5. The audio equipment networking configuration method according to claim 1, characterized in that: The method also includes: Before receiving a connection request from at least one slave speaker, obtaining a Bluetooth signal strength and interference condition corresponding to the execution environment of the master speaker; Perform analysis based on the Bluetooth signal strength and the interference situation to obtain corresponding environment analysis results; According to the environmental analysis result, it is determined whether the main speaker starts the multi-synchronous mode to manage the connection with the at least one secondary speaker. If it is not suitable to start the multi-synchronous mode, the main speaker provides the user with suggestions for improving the environmental conditions.

6. The audio equipment networking configuration method according to claim 1, characterized in that: The method also includes: After the secondary speaker is connected, an initialization calibration process is triggered, and the primary speaker sends a test audio signal to at least one secondary speaker; Acquire an actual response time of the at least one secondary speaker to the test audio signal; A playback delay compensation value set by the main speaker for each sub-speaker is determined based on the actual response time.

7. A device for configuring a network of audio equipment, characterized in that: include: A synchronous mode starting module, which establishes a connection between a main speaker and a source device through a Bluetooth protocol, and is used to start a multi-synchronous mode on the main speaker so that the main speaker enters a waiting state to receive a connection request from a secondary speaker, and at least one secondary speaker enters the multi-synchronous mode and sends a connection request to the main speaker; an audio data transmission channel establishment module, wherein after the main speaker receives the connection request from the secondary speaker, a unique device ID is assigned to each secondary speaker, and the device ID and its corresponding network parameters are recorded, so as to establish a corresponding audio data transmission channel between the main speaker and the secondary speaker, wherein the main speaker adopts adaptive audio stream control technology to dynamically adjust the size and transmission interval of the audio data packet according to the current network status; The synchronization module is used to synchronously broadcast the audio signal received from the source device to all connected secondary speakers through the audio data transmission channel. The main speaker and the secondary speakers are equipped with LED indicators for displaying the power status, charging status, Bluetooth connection status and multi-synchronization status.

8. The audio equipment networking configuration device according to claim 7, characterized in that: include: A key performance indicator acquisition module, used to acquire key performance indicators of the audio data transmission channel, wherein the key performance indicators include signal strength, packet loss rate, delay time, bandwidth usage and error rate; A matching module, which retrieves a preset performance indicator interval threshold value, and is used to match the key performance indicator with the performance indicator interval threshold value to obtain a corresponding matching result; The key performance indicator adjustment module determines a corresponding adjustment priority based on the matching result, and adjusts the key performance indicator based on the adjustment priority.

9. An electronic device, characterized in that: It comprises a processor, in which a program of the audio equipment networking configuration method as described in any one of claims 1 to 6 is run.

10. A storage medium, characterized in that: A program storing the audio equipment networking configuration method as described in any one of claims 1 to 6.

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