Network configuration method, device, electronic device and storage medium for audio equipment

Abstract

The present application relates to the technical field of computers, and discloses a method for networking and configuring audio devices, including: establishing a connection between a main speaker and a source device through a Bluetooth protocol, starting a multi-synchronization mode on the main speaker to make the main speaker enter a waiting state to receive connection requests from slave speakers, and at least one slave speaker enters the multi-synchronization mode and sends a connection request to the main speaker; after the main speaker receives the connection request from the slave speaker, assign a unique device ID to each slave speaker and establish a corresponding audio data transmission channel; synchronously broadcast 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 for displaying the power status, charging status, Bluetooth connection status, and multi-synchronization status. The present application can achieve fast and stable connection of a large number of audio devices for synchronous playback.

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 networking and configuring audio devices. Background Art

[0002] With the development of wireless audio technology, users' demand for portable audio devices is increasing day by day. 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 fails to meet the requirements of wide coverage and multi-point synchronous playback.

[0003] The current solutions 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: in complex environments or long-distance transmissions, disconnections are likely to occur; complex settings: users need to manually perform complex setting steps to achieve multi-device synchronous playback; limited scalability: the number of devices supporting synchronous playback is limited, making it difficult to meet the needs of large-scale applications; insufficient user experience: lack of an intuitive feedback mechanism, making users feel confused during the operation process.

[0004] Therefore, it is particularly important to develop a method that can simplify the operation process and support networking configuration.

[0005] As can be seen from the above, how to achieve fast and stable connection of a large number of audio devices for synchronous playback remains to be solved. Summary of the Invention

[0006] In order to achieve fast and stable connection of a large number of audio devices for synchronous playback, the present application provides a method, device, electronic device, and storage medium for networking and configuring audio devices.

[0007] In a first aspect, the present application provides a method for networking and configuring audio devices, adopting the following technical solution:

[0008] A method for networking and configuring an audio device, comprising: establishing a connection between a main speaker and a source device through a Bluetooth protocol, starting a multi-synchronization mode on the main speaker to make the main speaker enter a waiting state to receive connection requests from slave speakers, and at least one slave speaker enters the multi-synchronization mode and sends a connection request to the main speaker; after the main speaker receives the connection request from the slave speaker, assigns 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 an adaptive audio stream control technology to dynamically adjust the size and transmission interval of audio data packets according to the current network condition; through the audio data transmission channel, synchronously broadcasts the audio signal received from the source device to all connected slave speakers, and both the main speaker and the slave speaker are equipped with LED indicators for displaying the power state, charging state, Bluetooth connection state, and multi-synchronization state.

[0009] By adopting the above technical solution, the access of speakers and audio synchronous playback are achieved. This method particularly designs an adaptive audio stream control technology to dynamically adjust the size and transmission interval of audio data packets according to the real-time network condition, ensuring stable data transmission even in complex or changing network environments. At the same time, by assigning a unique device ID to each slave speaker and recording its network parameters, fast and stable connection configuration of a large number of synchronous playback devices is realized. The LED indicators provide intuitive status feedback, helping users easily understand the working conditions of each audio device, thus simplifying the deployment and management process of a large-scale audio system. This method not only optimizes the connection efficiency and stability of a 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.

[0010] Optionally, the method further includes: obtaining key performance indicators of the audio data transmission channel, wherein the key performance indicators include signal strength, packet loss rate, latency, bandwidth usage, and error rate; retrieving pre-set performance indicator interval thresholds, matching the key performance indicators with the performance indicator interval thresholds to obtain corresponding matching results; determining corresponding adjustment priorities based on the matching results, and adjusting the key performance indicators based on the adjustment priorities.

[0011] By adopting the above technical solutions, by continuously monitoring the key performance indicators of the audio data transmission channel, such as signal strength, packet loss rate, latency, bandwidth usage, and error rate, and matching these indicators with the pre-set performance interval thresholds to determine the adjustment priority. This method ensures that the system can dynamically optimize network parameters according to the current transmission quality, thereby maintaining the best audio playback effect. Through the automated performance evaluation and adjustment mechanism, the system can actively respond to network fluctuations, ensure the stability and audio quality consistency when large-scale audio devices play synchronously, reduce the need for manual intervention at the same time, and improve the adaptive ability and user experience of the system.

[0012] Optionally, the method further includes: when monitoring that a new secondary speaker attempts to connect, obtaining the bearing capacity corresponding to the current network environment, where the bearing capacity includes network bandwidth, signal strength, and data transmission rate, and obtaining the number of secondary speakers currently connected to the main speaker; judging whether to accept the connection request from the at least one new secondary speaker based on the network bearing capacity and the number of secondary speakers; if accepting, 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 condition to the user.

[0013] By adopting the above technical solutions, when a new secondary speaker attempts to connect, by intelligently evaluating the bearing capacity of the current network environment and the number of connected devices, dynamically determining whether to accept a new connection request. This mechanism ensures that the system will not cause performance degradation or audio quality impairment due to exceeding the network load. 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 audio quality of the existing audio system, but also improves the scalability of the system, ensuring efficient and stable device connection management even in a complex and changeable network environment.

[0014] Optionally, the method further includes: obtaining the physical location data corresponding to at least two secondary speakers connected to the main speaker, where the physical location data includes relative distance, direction, and potential obstacles in the environment; based on the physical location data, applying an optimization algorithm to calculate the audio signal distribution order and audio signal distribution intensity for each secondary speaker; adjusting the audio signal sent to each secondary speaker based on the audio signal distribution order and audio signal distribution intensity; when detecting that the position of any secondary speaker changes, re-obtaining the physical location data corresponding to the secondary speaker and adjusting the corresponding audio signal.

[0015] By adopting the above technical solution, the physical position data of the slave speakers connected to the main speaker is automatically obtained and analyzed, including the relative distance, direction, and obstacles in the environment, and an optimization algorithm is used to accurately calculate the audio signal distribution order and intensity of each slave speaker. This process ensures that the audio signal can be dynamically adjusted according to the actual layout, providing a more natural and immersive surround sound experience. More importantly, the system can automatically re-evaluate and adjust the audio settings when any change in the position of the slave speaker is detected, maintaining the best sound effect, thus significantly improving the flexibility and user experience of the multi-speaker system.

[0016] Optionally, the method further 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; analyzing based on the Bluetooth signal strength and the interference situation to obtain a corresponding environmental analysis result; according to the environmental analysis result, determining whether the main speaker starts a 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 a suggestion to the user to improve the environmental conditions.

[0017] By adopting the above technical solution, by pre-evaluating the surrounding Bluetooth signal strength and interference situation before receiving the slave speaker connection request, it is ensured that the main speaker makes an intelligent decision on whether to start the multi-synchronization mode according to the actual environmental conditions. This preventive measure helps to avoid forcibly establishing a multi-synchronization connection in a poor network environment, thus preventing potential connection instability or audio quality degradation problems. By providing specific environmental improvement suggestions to the user, the system not only improves the success rate and stability of the initial connection, but also optimizes the performance of the overall audio system, ensuring that the user obtains the best multi-speaker synchronization experience.

[0018] Optionally, the method further includes: after connecting the slave speaker, triggering an initialization calibration process, and the main speaker sends a test audio signal to at least one slave speaker; obtaining the actual response time of the at least one slave speaker to the test audio signal; determining the playback delay compensation value set by the main speaker for each slave speaker based on the actual response time.

[0019] By adopting the above technical solution, by automatically triggering the initialization calibration process after connecting the slave speaker, the actual response time of each slave speaker is accurately measured using the test audio signal, and then the optimal playback delay compensation value is set for each slave speaker. This calibration process ensures that the audio synchronization accuracy among all speakers reaches the highest level, and a seamless and high-fidelity audio playback experience can be achieved even in a complex multi-speaker system. Through the automated delay compensation mechanism, the system can effectively eliminate the time deviation caused by the difference in the transmission path, providing a more consistent and immersive sound effect.

[0020] In a second aspect, the present application provides a networking configuration device for audio devices, adopting the following technical solution:

[0021] A networking configuration device for audio devices, comprising:

[0022] A synchronization mode startup module that establishes a connection between a main speaker and a source device via a Bluetooth protocol, and is used to start a multi-synchronization mode on the main speaker to make the main speaker enter a waiting state to receive connection requests from slave speakers, and at least one slave speaker enters the multi-synchronization mode and sends a connection request to the main speaker;

[0023] An audio data transmission channel establishment module, after the main speaker receives the connection request from the slave speaker, assigns a unique device ID to each slave speaker, and records the device ID and its corresponding network parameters, and is used to establish a corresponding audio data transmission channel between the main speaker and the slave speaker. Among them, the main speaker adopts an adaptive audio stream control technology to dynamically adjust the size and transmission interval of audio data packets according to the current network condition;

[0024] A synchronization module, through the audio data transmission channel, is used to synchronously broadcast 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, which are used to display the power status, charging status, Bluetooth connection status, and multi-synchronization status.

[0025] In a third aspect, the present application provides an electronic device, adopting the following technical solution:

[0026] An electronic device, comprising a processor, and a program of any one of the above-mentioned audio device networking configuration methods is running in the processor.

[0027] In a fourth aspect, the present application provides a storage medium, adopting the following technical solution:

[0028] A storage medium stores a program of any one of the above-mentioned audio device networking configuration methods.

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

[0030] 1. First, the main speaker performs an environmental assessment before receiving the connection request from the secondary speaker to ensure that the network conditions are suitable for starting the multi-sync mode and provides improvement suggestions to the user to optimize the connection quality. Then, after the connection is established, the system automatically triggers the initialization calibration process, precisely measures the response time of each secondary speaker by sending test audio signals, and dynamically adjusts the playback delay compensation value to ensure that the audio synchronization accuracy among 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.

[0031] 2. In addition, this method utilizes adaptive audio stream control technology and continuously monitors key performance indicators to dynamically adjust the audio packet size and transmission interval according to the real-time network conditions, ensuring optimal data transmission stability 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 synchronized playback devices, provide intuitive status feedback, further simplify the deployment and management of large-scale audio systems, and ensure that users obtain a seamless and high-fidelity audio experience. BRIEF DESCRIPTION OF THE DRAWINGS

[0032] Figure 1 is a flowchart of a method for configuring a network of audio devices according to an exemplary embodiment.

[0033] Figure 2 is a block diagram of a device for configuring a network of audio devices according to an exemplary embodiment. DETAILED DESCRIPTION OF THE EMBODIMENTS

[0034] The following details the embodiments of the present application, and the examples of the embodiments are shown in the drawings.

[0035] 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 connection with the embodiments or examples are included in at least one embodiment or example of the present application. In this specification, the schematic representations of the above terms do not necessarily refer to the same embodiments or examples. Moreover, the specific features, structures, materials, or characteristics described can be combined in any one or more embodiments or examples in a suitable manner.

[0036] The embodiments of the present application disclose a method for configuring a network of audio devices, referring to Figure 1 , including:

[0037] S100. Connect a main speaker to a source device via Bluetooth protocol. Start the multi - synchronization mode on the main speaker to put the main speaker into a waiting state to receive connection requests from slave speakers, and at least one slave speaker enters the multi - synchronization mode and sends a connection request to the main speaker.

[0038] First of all, the user connects a main speaker to an audio source device (such as a smartphone, a tablet or a computer) via 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 the audio stream from the source device.

[0039] After the connection is completed, the user can start the multi - synchronization mode on the main speaker. At this time, the main speaker enters a special state, ready to accept connection requests from other slave speakers. It should be noted here that the main speaker activates its multi - synchronization function module, opens specific Bluetooth services to allow slave speakers to connect. It may also broadcast its own presence information so that surrounding slave speakers can discover and attempt to connect.

[0040] Then, at least one slave speaker also needs to enter the multi - synchronization 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 via Bluetooth protocol, including its own basic information (such as model, current status, etc.).

[0041] S110. After the main speaker receives the connection request from the slave speaker, assign a unique device ID to each slave speaker, record the device ID and its corresponding network parameters, and establish a corresponding audio data transmission channel between the main speaker and the slave speaker.

[0042] 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, so as 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.

[0043] In addition, the main speaker generates a globally unique identifier or creates a unique ID based on existing standards. The main speaker stores the network parameters of each slave speaker, including but not limited to RSSI, channel information, connection quality, etc.

[0044] A dedicated audio data transmission channel is established between the main speaker and each secondary speaker. Using adaptive audio stream control technology, the main speaker dynamically adjusts the size and transmission interval of audio data packets according to the current network conditions to optimize bandwidth usage and reduce latency, achieving efficient and stable audio data transmission and maintaining good sound quality even when the network conditions change.

[0045] S120, through the audio data transmission channel, synchronously broadcasts the audio signal received from the source device to all connected secondary speakers. Both the main speaker and the secondary speakers are equipped with LED indicators for displaying the power status, charging status, Bluetooth connection status, and multi-sync status.

[0046] Among them, the main speaker synchronously broadcasts the audio signal received from the source device to all connected secondary speakers through the established audio data transmission channel, thereby ensuring that all secondary speakers play the same audio content simultaneously, providing a unified sound effect experience. It should be noted here that the main speaker processes the audio stream and ensures that all secondary speakers play synchronously according to a preset schedule or based on a real-time feedback mechanism.

[0047] In addition, both the main speaker and the secondary speakers are equipped with LED indicators for intuitively displaying the power status, charging status, Bluetooth connection status, and multi-sync status, and different colors or blinking patterns of the LED indicate different working states, helping users quickly understand the current situation of the device. The color and blinking frequency of the LED indicator are controlled by an internal state machine, reflecting the current state of the device.

[0048] Through the above implementation steps, the networking configuration method of this audio device ensures that a large number of audio devices for synchronous playback can be quickly and stably connected through a series of carefully designed technical steps. It not only simplifies the user's setup process, improves the flexibility and scalability of the audio system, but also guarantees 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, ensuring the efficient deployment and management of a large-scale audio system, thereby providing users with a consistent and immersive sound effect experience. And this method is particularly suitable for scenarios that require quickly setting up and adjusting the synchronous playback of a large number of audio devices, such as home theaters, commercial exhibition venues, outdoor activities, etc., significantly improving the user experience and the practicality of the system.

[0049] When starting the multi-sync mode on the main speaker to make the main speaker enter the waiting state, the situation of the current environment needs to be considered to determine whether it is suitable to connect to the secondary speakers. Therefore, the method further includes:

[0050] S101. Before receiving the connection request of at least one secondary speaker, obtain the Bluetooth signal strength and interference situation corresponding to the main speaker's execution environment.

[0051] Among them, before receiving the connection request of the secondary 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 the Bluetooth signal strength and interference situation in the surrounding environment.

[0052] Specifically, the main speaker starts the scanning function, detects and records the signal strength of all nearby Bluetooth devices, and analyzes the 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 the channel congestion to determine whether there are potential transmission conflicts.

[0053] S102. Analyze based on the Bluetooth signal strength and interference situation to obtain the corresponding environmental analysis result.

[0054] Among them, the collected Bluetooth signal strength and interference situation are sent to the analysis algorithm built in the main speaker for processing. And the algorithm calculates a comprehensive environmental score or evaluation result according to factors such as Bluetooth signal strength, the number and strength of interference sources, and channel congestion degree.

[0055] Specifically, use machine learning or a preset rule library to analyze the data to determine whether the current environment is suitable for starting the multi-synchronization 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 good. If there are strong interference sources or multiple devices are working on the same channel, which may lead to unstable data transmission, the environment may not be suitable for starting the multi-synchronization mode.

[0056] Finally, after the analysis is completed, the system generates a detailed environmental analysis report, which includes key information such as the specific values of Bluetooth signal strength, the number and strength of interference sources, and channel congestion degree. The environmental analysis result can be quantified as specific indicators, such as "good", "general", or "poor", or can be expressed as a percentage or a score.

[0057] S103. According to the environmental analysis result, judge whether the main speaker starts the multi-synchronization mode to manage the connection with at least one secondary speaker. If it is not suitable to start the multi-synchronization mode, the main speaker provides suggestions to the user to improve the environmental conditions.

[0058] Among them, according to the environmental analysis results, the main speaker decides whether to start the multi-synchronization mode. If the environmental score reaches the preset threshold, the multi-synchronization mode is allowed to be started; otherwise, the system will choose not to start. It should be noted here that the preset threshold can be flexibly adjusted according to different application scenarios. For example, there may be different standards for home environments and commercial display venues. The main speaker can make internal logical judgments or interact with the cloud server to obtain the latest configuration suggestions.

[0059] If the environmental analysis results show that it is not suitable to start the multi-synchronization mode, the main speaker will not accept the connection request from the secondary speaker. It can send a rejection message to the user, explain the reasons, and provide specific improvement suggestions. And the rejection message can be conveyed to the user through the display screen, mobile application or voice prompt, etc. The improvement suggestions can include specific measures such as moving the positions of some devices, turning off interference sources, and adjusting the settings of the Wi-Fi router.

[0060] Based on the execution steps of S101 to S103 above, by pre-evaluating the surrounding Bluetooth signal strength and interference before receiving the secondary speaker connection request, it is ensured that the main speaker makes an intelligent decision on whether to start the multi-synchronization mode according to the actual environmental conditions. This preventive measure helps to avoid forcibly establishing a multi-synchronization connection in a poor network environment, thus preventing potential connection instability or audio quality degradation problems. By providing specific environmental improvement suggestions to the user, the system not only improves the success rate and stability of the initial connection, but also optimizes the performance of the overall audio system, ensuring that users obtain the best multi-speaker synchronization experience. This method is particularly suitable for scenarios that require rapid deployment and adjustment of a large number of audio devices for synchronous playback, significantly improving the reliability of the system and user satisfaction.

[0061] After the connection is completed, it is necessary to monitor the audio data transmission channel. Therefore, the method also includes:

[0062] S1211, obtain the key performance indicators of the audio data transmission channel.

[0063] Among them, the system regularly or real-time monitors the audio data transmission channel between the main speaker and the secondary speaker, collects a series of key performance indicators, and the key performance indicators include but are not limited to signal strength, packet loss rate, latency, bandwidth usage, and error rate.

[0064] Specifically, signal strength: The quality of the Bluetooth connection is evaluated by measuring RSSI. A higher RSSI value indicates stronger signal quality; Packet loss rate: The proportion of packets that fail to be successfully transmitted within a certain period of time is counted. A lower packet loss rate means a more stable connection; Latency time: The time difference from sending data to receiving an acknowledgment is measured. Low latency is crucial for synchronous audio playback; Bandwidth usage: Monitor the amount of bandwidth currently in use to ensure there are sufficient resources to support high-quality audio streams; Error rate: Record the number of errors that occur during transmission, such as checksum failures, etc., to evaluate transmission reliability.

[0065] S1212, retrieve the pre-set performance metric interval thresholds, match the key performance metrics with the performance metric interval thresholds, and obtain the corresponding matching results.

[0066] Among them, the main speaker accesses its built-in database or the cloud server to obtain the pre-set performance metric interval thresholds; these thresholds define the optimal range and acceptable range for each performance metric.

[0067] It should be noted here that the pre-set performance metric interval thresholds can be adjusted according to different application scenarios. For example, the home environment and commercial display venues may have different standards. Each performance metric has an ideal range and a warning range to distinguish normal operation and potential problems. For example, the ideal range of signal strength may be -50dBm to -70dBm, while the warning range may be below -80dBm.

[0068] Compare the actually measured key performance metrics with the pre-set interval thresholds to determine the matching results for each metric. The matching results can be "Normal", "Warning", or "Error", depending on which range the actual value falls into.

[0069] Specifically, use an algorithm to automatically compare the actual value and the threshold to generate a detailed matching report. If a certain performance metric exceeds the warning range, it is marked as "Warning", and if it exceeds the error range, it is marked as "Error". For multiple performance metrics, the system will comprehensively consider all matching results to form an overall evaluation conclusion. By identifying which performance metrics deviate from the expected range, potential problem points can be located.

[0070] S1213, determine the corresponding adjustment priorities based on the matching results, and adjust the key performance metrics based on the adjustment priorities.

[0071] Among them, according to the matching results, the system assigns an adjustment priority to each performance metric. The adjustment priority reflects the degree of influence of the performance metric on the overall system performance.

[0072] Key performance indicators are divided into high priority, medium priority, and low priority. For example, if both the latency time and the packet loss rate are in the "warning" state, but the latency time 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.

[0073] In addition, the system takes corresponding measures to optimize the key performance indicators according to the determined adjusted priority, which may involve operations such as adjusting the audio packet size, changing the transmission interval, and switching channels.

[0074] Specific details include:

[0075] Adjusting the audio packet size: Reducing the packet size can reduce the risk of single transmission failure, especially in the case of weak signal strength; changing the transmission interval: Increasing the transmission interval can reduce bandwidth usage and congestion, which is suitable for high-bandwidth occupancy situations; switching channels: If the current channel is detected to be congested or there is a strong interference source, 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 the power output to enhance signal coverage.

[0076] 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, the intelligent evaluation and dynamic adjustment of the sound system network status are achieved. This mechanism ensures that the system can dynamically optimize network parameters according to the current transmission quality, thereby maintaining the best audio playback effect. Through the automated performance evaluation and adjustment mechanism, the system can actively respond to network fluctuations, ensure the stability and sound quality consistency when a large number of audio devices play synchronously, reduce the need for manual intervention at the same time, and improve the adaptive ability and user experience of the system. This method is particularly suitable for complex and changing network environments, significantly improving the reliability of the sound system and user satisfaction.

[0077] When monitoring the attempt of a new sub-speaker to connect, the method further includes:

[0078] S1221, obtaining the bearing capacity corresponding to the current network environment, where the bearing capacity includes network bandwidth, signal strength, and data transmission rate, and obtaining the number of sub-speakers currently connected to the main speaker.

[0079] Among them, the main speaker uses a Bluetooth module or other wireless communication interfaces (such as Wi-Fi) for regular scanning to identify the presence of new devices. Whenever a connection request sent by a sub-speaker is detected, the main speaker records the timestamp of the request and the basic information of the sub-speaker (such as MAC address, model, etc.).

[0080] When a new secondary speaker attempts to connect, the primary speaker immediately evaluates the bearing capacity of the current network environment, which includes but is not limited to network bandwidth, signal strength, and data transmission rate.

[0081] Specifically, for network bandwidth: by measuring the ratio of the currently used bandwidth to the available bandwidth, it is evaluated whether there is sufficient bandwidth to support the new connection; for signal strength: measure the RSSI of the existing connection to determine whether the signal quality is strong enough; for data transmission rate: check the current data transmission rate to ensure that it can support more devices without degrading performance.

[0082] Furthermore, the primary speaker queries its internal database or connection management module to obtain information about all connected secondary speakers. The statistical results include not only the quantity but also the working status of each secondary speaker (such as online, offline, playing, etc.).

[0083] S1222, based on the network bearing capacity and the number of secondary speakers, determine whether to accept the connection request from at least one new secondary speaker.

[0084] Among them, based on the data collected previously, the primary speaker makes a comprehensive evaluation to decide whether to accept the connection request from the new secondary speaker. If the current network bearing capacity is sufficient and the number of connected secondary speakers does not reach the preset upper limit, the new connection is accepted; otherwise, it is rejected.

[0085] The system can flexibly adjust the threshold according to the actual situation. For example, more connections may be allowed in a home environment, while it may be more strict in a commercial exhibition venue. If multiple new secondary speakers request connections simultaneously, the system can selectively accept the connections based on priority rules (such as user-specified priorities or automatically evaluated priorities).

[0086] 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 suggestions to the user to improve the network condition.

[0087] Among them, if the evaluation result shows that the connection request from the new secondary speaker can be accepted, the primary speaker will formally establish the connection. The primary speaker sends a confirmation message to the new secondary speaker to complete the pairing process, update the internal database, assign a unique device ID to the new secondary speaker, record the relevant network parameters, and establish a dedicated audio data transmission channel to ensure that the new secondary speaker can receive the audio stream synchronously.

[0088] If the evaluation result shows that it is not suitable to accept the connection request from the new secondary speaker, the primary speaker will send a rejection message to the new secondary speaker. At the same time, specific improvement suggestions will be provided to the user to help optimize the network conditions.

[0089] In addition, the rejection message can be conveyed to the user through means such as a display screen, a mobile application, or a voice prompt. The improvement suggestions can include specific measures such as moving the positions of certain devices, turning off interference sources, and adjusting the settings of the Wi-Fi router. The system can also provide a detailed report explaining why the current network environment is not suitable for adding more devices.

[0090] By intelligently evaluating the carrying capacity of the current network environment and the number of connected devices when a new secondary speaker attempts to connect, it dynamically decides whether to accept the new connection request. This mechanism ensures that the system will not experience performance degradation or impaired audio quality due to exceeding the network load. At the same time, by providing specific network improvement suggestions, it helps users optimize the 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 enhances the scalability of the system, ensuring efficient and stable device connection management even in a complex and changing network environment. This method is particularly suitable for scenarios that require rapid deployment and adjustment of a large number of audio devices for synchronous playback, significantly improving the reliability of the system and user satisfaction.

[0091] Furthermore, the method further includes:

[0092] S1231, obtaining physical location data corresponding to at least two secondary speakers connected to the main speaker.

[0093] Among them, the system automatically responds to the user's request and starts collecting the 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.

[0094] Specifically, relative distance: Use technologies such as Bluetooth ranging, Wi-Fi RTT, and ultrasonic sensors 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 detect potential obstacles in the environment with the help of external cameras, infrared sensors, etc.

[0095] S1232, based on the physical location data, applying an optimization algorithm to calculate the audio signal distribution order and audio signal distribution intensity for each secondary speaker.

[0096] Among them, based on the collected physical location data, the main speaker applies an optimization algorithm to calculate the audio signal distribution order and intensity for each secondary speaker. The optimization algorithm can be a rule-based static algorithm or a machine learning model. These algorithms consider multiple factors such as distance, direction, obstacles, and room acoustic characteristics.

[0097] Determine the time difference of the audio signal arrival according to the positional relationship of the secondary speakers to achieve the best surround sound effect, adjust the volume of each secondary speaker to ensure that the sounds output by all speakers can be harmoniously integrated in space, and avoid overly strong or weak sounds in certain areas.

[0098] S1233, adjust the audio signal sent to each secondary speaker based on the audio signal distribution order and audio signal distribution intensity.

[0099] Among them, according to the calculated audio signal distribution order and intensity, the main speaker adjusts the audio signal sent to each secondary speaker.

[0100] Specific details include the following situations. Time calibration: For each secondary speaker, set an appropriate playback delay to ensure that the audio signal can reach each position in a 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 operations 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 the expectations.

[0101] S1234, if it is detected that the position of any secondary speaker has changed, re-obtain the physical position data corresponding to the secondary speaker and adjust the corresponding audio signal.

[0102] Among them, the main speaker continuously monitors the position changes of the connected secondary speakers. Once it is detected that the position of any secondary speaker has changed, immediately re-obtain its physical position data and adjust the audio signal accordingly.

[0103] 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 it is detected that the position change exceeds the preset threshold, trigger the process of re-obtaining the position data; Apply the optimization algorithm again, re-calculate the audio signal distribution order and intensity, 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 an LED indicator, a mobile application or a voice prompt, and provide further guidance.

[0104] By automatically acquiring and analyzing the physical location data of the secondary speakers connected to the main speaker, and using an optimization algorithm to accurately calculate the audio signal distribution order and intensity of each secondary speaker, the 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, providing a more natural and immersive surround sound experience, but also can automatically re-evaluate and adjust the audio settings when any change in the position of the secondary speaker is detected, maintaining the best sound effect. This method significantly improves the flexibility and user experience of the multi-speaker system, and is particularly suitable for scenarios that require frequent adjustment of the audio equipment layout or where there are many changes in the environment, such as home theaters, commercial exhibition venues, or outdoor activities.

[0105] The method further includes:

[0106] S1241, after connecting the secondary speaker, trigger the initialization calibration process, and the main speaker sends a test audio signal to at least one secondary speaker.

[0107] Among them, after successfully connecting the secondary speaker, the main speaker automatically triggers an initialization calibration process, which aims to ensure that the audio synchronization accuracy between all secondary speakers reaches the highest level.

[0108] Specifically, the calibration process can be triggered when a new secondary speaker is added each time, after the system restarts, or when the user manually starts it. The main speaker pauses the current audio playback, enters the calibration mode, and notifies all connected secondary speakers to prepare to receive the test audio signal. The main speaker records the current timestamp, which serves as the basis for subsequent calculation of the response time.

[0109] The main speaker sends a series of carefully designed test audio signals to at least one secondary speaker. The test audio signal can be a tone of a specific frequency, a pulse sequence, or other forms of sound samples, which are used to accurately measure the characteristics on the transmission path.

[0110] It should be noted here that the test audio signal should have clear time characteristics, such as short pulses or sine waves of specific frequencies, so as to accurately measure the transmission delay. In order to comprehensively evaluate the response at different frequencies, test signals of multiple frequency bands can be sent.

[0111] If there are multiple secondary speakers, the test signals can be sent in groups or one by one to avoid signal overlap affecting the measurement results. Each test audio signal is attached with a unique synchronization mark to facilitate the secondary speaker to identify and return the confirmation signal.

[0112] S1242, obtain the actual response time of at least one secondary speaker to the test audio signal.

[0113] After each secondary speaker receives the test audio signal, it immediately sends a confirmation signal back to the primary speaker. The primary speaker records the time difference between sending the test audio signal and receiving the confirmation signal, which is the actual response time.

[0114] 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 taken and the average value can be used. At the same time, considering the influence of environmental factors such as temperature and humidity on the signal propagation speed, appropriate compensation is carried out, and the response time of each secondary speaker is recorded in detail to form a database for subsequent analysis and adjustment.

[0115] S1243, determine the playback delay compensation value set by the primary speaker for each secondary speaker based on the actual response time.

[0116] Among them, based on the measured actual response time, the primary speaker sets an appropriate playback delay compensation value for each secondary speaker. An internal algorithm is used to calculate the delay compensation value required for each secondary speaker according to the response time. For example, the secondary speaker with the longest response time does not require additional delay, while other secondary speakers need to increase the delay accordingly. And the compensation value can be dynamically adjusted according to real-time feedback to ensure the best synchronization effect in different environments. The calculated delay compensation value is stored in the configuration file of the primary speaker and immediately applied during the audio playback process. After the setting is completed, the test audio signal can be sent again for verification to ensure that the adjusted effect meets the expectations.

[0117] By automatically triggering the initialization calibration process after connecting the secondary speakers, using the test audio signal to accurately measure the actual response time of each secondary speaker, and setting the optimal playback delay compensation value for each secondary speaker accordingly. This calibration process ensures that the audio synchronization accuracy among all speakers reaches the highest level, and even in a complex multi-speaker system, a seamless and high-fidelity audio playback experience can be achieved. Through the automated delay compensation mechanism, the system can effectively eliminate the time deviation caused by transmission path differences, provide a more consistent and immersive sound effect, and significantly improve the flexibility and user experience of the multi-speaker system.

[0118] Scene description

[0119] There are three main spaces in the home, including the living room, dining room, and study. It is desired to install audio equipment in these three areas so that the same music can be played synchronously in different rooms or the sound sources in each room can be independently controlled as needed. To achieve this goal, a set of primary and secondary speaker systems that support the Bluetooth protocol is selected, and this system has advanced automatic environment assessment, intelligent connection management, and dynamic performance optimization functions.

[0120] Case implementation

[0121] First, place the main speaker in the living room and pair it with your smartphone via Bluetooth. After completing the pairing, the multi-sync mode on the main speaker is activated. At this time, the main speaker automatically starts scanning the surrounding wireless environment, evaluating the Bluetooth signal strength and possible interference situations. Through built-in algorithm analysis, the main speaker confirms that the current environment is suitable for activating the multi-sync mode and is ready to receive connection requests from other secondary speakers.

[0122] Subsequently, place two secondary speakers in the dining room and the study respectively and turn on their multi-sync modes. These two secondary speakers quickly detect the presence of the main speaker and send connection requests. After receiving these requests, the main speaker assigns a unique device ID to each secondary speaker and establishes a stable audio data transmission channel; the system also performs an initialization calibration process, sending test audio signals to measure the actual response time of each secondary speaker and setting appropriate playback delay compensation values accordingly to ensure the highest level of audio synchronization accuracy among all speakers.

[0123] When playing music, the main speaker synchronously broadcasts the audio stream to all connected secondary speakers through the established audio data transmission channel, filling the whole house with a harmonious music atmosphere. In addition, whenever a new secondary speaker tries to join the network, the main speaker will re-evaluate the current network carrying capacity and the number of connected devices and decide whether to accept the new device to join, so as to ensure the stability of the system and the audio quality.

[0124] To further enhance the user experience, the system can automatically adjust the audio signal distribution order and intensity according to the physical positions of the secondary speakers, achieving the best surround sound effect. Even if the user changes the position of a certain speaker, the system can detect the change in real time and adjust the audio settings accordingly to maintain the optimal sound effect; ultimately, the home audio system not only provides a high-quality, synchronous audio experience, but also its intelligent design makes the system maintenance simple and intuitive, and the status feedback of the LED indicator helps to understand the working conditions of the devices at any time, allowing you to easily enjoy the immersive music world.

[0125] This case demonstrates the flexibility and efficiency of the networking configuration method of audio equipment in practical application scenarios, and also emphasizes the importance of technology in improving the user experience.

[0126] The embodiment of the present application discloses a networking configuration device for audio equipment, referring to Figure 2 , the device includes but is not limited to:

[0127] The synchronous mode startup module 200 establishes a connection between a main speaker and a source device via the Bluetooth protocol, which is used to start a multi-synchronous mode on the main speaker to make the main speaker enter a waiting state to receive connection requests from slave speakers, and at least one slave speaker enters the multi-synchronous mode and sends a connection request to the main speaker;

[0128] The audio data transmission channel establishment module 210, after the main speaker receives the connection request from the slave speaker, assigns a unique device ID to each slave speaker, and records the device ID and its corresponding network parameters, which are used to establish corresponding audio data transmission channels between the main speaker and the slave speakers. Among them, the main speaker adopts adaptive audio stream control technology to dynamically adjust the size and transmission interval of audio data packets according to the current network conditions;

[0129] The synchronization module 220 is used to synchronously broadcast the audio signals received from the source device to all connected slave speakers through the audio data transmission channels. Both the main speaker and the slave speakers are equipped with LED indicators, which are used to display the power status, charging status, Bluetooth connection status, and multi-synchronous status.

[0130] Furthermore, the device includes but is not limited to:

[0131] The key performance indicator acquisition module is used to acquire the key performance indicators of the audio data transmission channel. Among them, the key performance indicators include signal strength, packet loss rate, latency, bandwidth usage, and error rate;

[0132] The matching module retrieves the pre-set performance indicator interval thresholds, which are used to match the key performance indicators with the performance indicator interval thresholds to obtain corresponding matching results;

[0133] The key performance indicator adjustment module determines the corresponding adjustment priority based on the matching results, and is used to adjust the key performance indicators based on the adjustment priority.

[0134] Furthermore, the device includes but is not limited to:

[0135] The bearing capacity acquisition module is used to acquire the bearing capacity corresponding to the current network environment when monitoring that a new slave speaker attempts to connect. Among them, the bearing capacity includes network bandwidth, signal strength, and data transmission rate, and acquires the number of slave speakers currently connected to the main speaker;

[0136] The judgment module is used to judge whether to accept the connection request from the at least one new slave speaker based on the network bearing capacity and the number of slave speakers;

[0137] If accepted, establish a connection with the at least one new secondary speaker; otherwise, send a rejection message to the at least one new secondary speaker and provide the user with suggestions for improving the network condition.

[0138] Further, the device includes but is not limited to:

[0139] A physical location data acquisition module, configured to acquire physical location data corresponding to at least two secondary speakers connected to the main speaker, where the physical location data includes relative distance, direction, and potential obstacles in the environment;

[0140] An audio signal calculation module, based on the physical location data, applies an optimization algorithm to calculate the audio signal distribution order and audio signal distribution intensity for each secondary speaker;

[0141] An audio signal adjustment module, based on the audio signal distribution order and audio signal distribution intensity, adjusts the audio signals sent to each secondary speaker;

[0142] If it is detected that the position of any secondary speaker changes, re-acquire the physical location data corresponding to the secondary speaker and adjust the corresponding audio signals.

[0143] Further, the device includes but is not limited to:

[0144] A Bluetooth signal strength acquisition module, before receiving a connection request from at least one secondary speaker, is configured to acquire the Bluetooth signal strength and interference situation corresponding to the execution environment of the main speaker;

[0145] An environment analysis result acquisition module, based on the Bluetooth signal strength and the interference situation, analyzes to acquire the corresponding environment analysis result;

[0146] A synchronous start judgment module, according to the environment analysis result, is configured to judge whether the main speaker starts a 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.

[0147] Further, the device includes but is not limited to:

[0148] A trigger module, after connecting the secondary speaker, is configured to trigger an initialization calibration process, and the main speaker sends a test audio signal to at least one secondary speaker;

[0149] An actual response time acquisition module, configured to acquire the actual response time of the at least one secondary speaker to the test audio signal;

[0150] A playback delay compensation value determination module, based on the actual response time, determines the playback delay compensation value set by the main speaker for each secondary speaker.

[0151] An embodiment of the present application also discloses an electronic device, including a processor, and a program of the audio device networking configuration method described in any one of the above is running in the processor.

[0152] An embodiment of the present application also discloses a storage medium storing a program of the audio device networking configuration method described in any one of the above.

[0153] Although the embodiments of the present application have been shown and described above, it can be understood that the above embodiments are exemplary and should not be construed as limiting the present application. Those of ordinary skill in the art can make changes, modifications, substitutions, and variations to the above embodiments within the scope of the present application.

Claims

1. A 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; 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; Analyze the Bluetooth signal strength and the interference situation to obtain a corresponding environmental analysis result; determine whether the main speaker starts a multi-synchronous mode to manage the connection with the at least one secondary speaker according to the environmental analysis result, and if it is not suitable to start the multi-synchronous mode, the main speaker provides a user with suggestions for improving environmental conditions; After the main speaker receives the connection request from the slave speaker, it 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; when monitoring a new slave speaker trying to connect, the carrying capacity corresponding to the current network environment is obtained, wherein the carrying capacity includes network bandwidth, signal strength and data transmission rate, and the number of slave speakers currently connected to the main speaker is obtained; based on the network carrying capacity and the number of slave speakers, it is determined whether to accept the connection request from the at least one new slave speaker; if accepted, a connection with the at least one new slave speaker is established, otherwise a rejection message is sent to the at least one new slave speaker, and suggestions for improving the network status are provided to the user; 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 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.

4. 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.

5. An audio equipment networking configuration device, used to execute the audio equipment networking configuration method according to any one of claims 1 to 4, characterized in that: The device includes: 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.

6. The audio equipment networking configuration device according to claim 5, 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.

7. 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 4 is run.

8. A storage medium, characterized in that: A program storing the audio equipment networking configuration method as described in any one of claims 1 to 4.

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