Multi-user audio sharing method and system based on le-audio Bluetooth headset

Abstract

The invention relates to the technical field of audio sharing, in particular to a multi-user audio sharing method and system based on a le-audio Bluetooth headset. The method comprises the following steps of: acquiring a connection state and a topological structure of LE-Audio equipment in real time through a network discovery function of a Bluetooth protocol stack so as to generate a Bluetooth dynamic network topological graph; dividing the available bandwidth into a plurality of small-bandwidth channels based on the dynamic network topological graph, and allocating independent small-bandwidth channels for each user for audio transmission to form a plurality of parallel audio stream channels; modeling a multi-user audio sharing scene according to the audio stream channel to obtain a channel response matrix describing multi-user audio transmission characteristics; and performing adaptive dynamic adjustment on the audio transmitted in each small-bandwidth channel according to the channel response matrix to generate adaptive audio stream data. According to the invention, through small bandwidth division and multi-channel transmission, efficient transmission of multi-user bidirectional audio streams is realized.

Description

Technical Field

[0001] The present invention relates to the technical field of audio sharing, and in particular to a multi-user audio sharing method and system based on a LE-audio Bluetooth headset. Background Art

[0002] With the rapid development of wireless audio technology, Bluetooth headsets have gradually become an indispensable audio device in people's daily lives. However, traditional Bluetooth audio technology has many limitations in multi-user scenarios, such as channel interference, audio delay, and difficulty in synchronizing multiple devices. These problems greatly affect the user's audio experience. In order to solve these problems, LE-Audio technology came into being.

[0003] LE-Audio (Low Energy Audio) is an emerging Bluetooth audio standard that aims to improve the quality and efficiency of wireless audio transmission by introducing a number of advanced technologies such as low-latency transmission, bandwidth optimization, and multi-device synchronization. The multi-user audio sharing method based on LE-Audio technology aims to overcome the limitations of traditional Bluetooth audio technology and provide users with higher quality, lower latency, and more stable audio experience.

[0004] Traditional Bluetooth audio transmission methods are susceptible to channel interference in multi-user environments, resulting in unstable audio transmission and degraded sound quality; in multi-user audio sharing scenarios, audio delay differences between different devices can cause synchronization issues, affecting user experience; traditional Bluetooth technology only supports one-way audio stream transmission, while multi-user audio sharing needs to support two-way audio stream transmission between multiple devices, which puts forward new requirements for audio encoding and decoding, clock synchronization, etc. Summary of the invention

[0005] Based on this, it is necessary for the present invention to provide a multi-user audio sharing method and system based on le-audio Bluetooth headset to solve at least one of the above technical problems.

[0006] To achieve the above purpose, a multi-user audio sharing method based on le-audio bluetooth headset includes the following steps:

[0007] Step S1: The connection status and topology of the LE-Audio device are obtained in real time through the network discovery function of the Bluetooth protocol stack, thereby generating a dynamic network topology map of Bluetooth; based on the dynamic network topology map, the available bandwidth is divided into multiple small bandwidth channels, and an independent small bandwidth channel is allocated to each user for audio transmission, forming multiple parallel audio stream channels;

[0008] Step S2: Modeling a multi-user audio sharing scenario according to the audio stream channel to obtain a channel response matrix describing the multi-user audio transmission characteristics; adaptively and dynamically adjusting the audio transmitted in each small bandwidth channel according to the channel response matrix to generate adaptive audio stream data;

[0009] Step S3: using the IEEE 1588PTP protocol to synchronize the clocks between the devices participating in the sharing according to the dynamic network topology, measuring the transmission delay from each device to the audio source, and performing delay compensation on the adaptive audio stream data, thereby obtaining synchronous audio stream data;

[0010] Step S4: construct an audio data transmission network according to the dynamic network topology diagram, and perform consistency confirmation transmission on the synchronous audio stream data through a consensus mechanism to obtain consistent audio stream data; perform information verification based on hash fingerprint on the consistent audio stream data to obtain hash verification result data, and perform selective retransmission triggering or forward error correction on the consistent audio stream data according to the hash verification result data to achieve multi-user sharing of audio.

[0011] The present invention utilizes the network discovery function of the Bluetooth protocol stack to obtain the connection status and topology of the LE-Audio device in real time, ensuring the dynamics and accuracy of the network. The available bandwidth is divided into multiple small bandwidth channels, and an independent channel is allocated to each user, which can avoid channel interference, optimize bandwidth utilization, and ensure the stability and integrity of audio transmission. By forming multiple parallel audio stream channels, multiple users are supported to perform audio transmission at the same time, meet the audio sharing needs of multiple users, and improve the scalability and flexibility of the system. The multi-user audio sharing scenario is modeled to obtain a channel response matrix, which can accurately describe the multi-user audio transmission characteristics and improve the prediction and optimization capabilities of audio transmission. According to the channel response matrix, the audio is adaptively and dynamically adjusted to ensure high quality and low latency of audio transmission, adapt to transmission requirements in different environments, and enhance user experience. Through the adaptive adjustment mechanism, the transmission path and transmission parameters of audio stream data are optimized, the transmission efficiency and stability are improved, and the transmission delay and error are reduced. The IEEE 1588PTP protocol can achieve accurate clock synchronization between devices participating in sharing, ensure that each device works under the same time reference, and avoid audio asynchrony problems caused by clock deviation. By measuring the transmission delay from each device to the audio source and performing delay compensation, the delay difference between devices can be eliminated, the synchronous transmission of audio streams can be achieved, and the synchronous audio playback between multiple devices can be ensured. Synchronous audio stream data ensures the time consistency of audio playback on each device, avoids audio delay and distortion caused by asynchronous audio playback on multiple devices, and improves the user experience in multi-user sharing scenarios. An audio data transmission network is constructed based on a dynamic network topology diagram to ensure the optimal path of audio data transmission and the stability of the network structure, thereby improving the reliability and efficiency of data transmission. The synchronous audio stream data is transmitted with consistency confirmation through a consensus mechanism to ensure that the consistent audio stream data received by each device is the same, thereby avoiding data inconsistency during data transmission. Information verification based on hash fingerprints can effectively detect errors and tampering in audio data transmission to ensure data integrity and security. Selective retransmission triggering or forward error correction mechanism can promptly repair transmission errors and improve the reliability and quality of audio data transmission. Through the combination of the above steps and mechanisms, multi-user sharing of audio is achieved, ensuring the consistency and high quality of audio experience of multiple users in sharing scenarios.

[0012] Preferably, step S1 comprises the following steps:

[0013] Step S11: Scan the surrounding LE-Audio devices using the scanning function of the Bluetooth protocol stack, and collect the scanned device information to obtain the original device scanning data;

[0014] Step S12: authenticating the scanned device according to the original device scan data and establishing a secure pairing connection, thereby obtaining a list of verified devices;

[0015] Step S13: obtaining the connection relationship between devices in the verified device list through the network discovery function of the Bluetooth protocol stack, determining the master and slave device roles, identifying the communication path between devices, and obtaining initial topology structure data;

[0016] Step S14: establishing a graphical representation of device nodes and connection edges according to the initial topology data, and adding device attribute information, thereby obtaining a dynamic network topology diagram of Bluetooth;

[0017] Step S15: performing link quality analysis on the dynamic network topology diagram and estimating the total available bandwidth, thereby obtaining bandwidth resource evaluation data;

[0018] Step S16: performing small bandwidth channel division according to the bandwidth resource evaluation data and the number of user equipments, and assigning a unique identifier to each channel, thereby obtaining channel division data;

[0019] Step S17: allocating an independent small-bandwidth channel for audio transmission to each user according to the channel division data, thereby forming multiple parallel audio stream channels.

[0020] The present invention can discover the surrounding LE-Audio devices in real time through the scanning function of the Bluetooth protocol stack, ensuring that all devices can participate in the audio sharing network. The scanned device information, including device type, signal strength, etc., is collected to provide basic data for subsequent device identity authentication and network topology construction. The wide scanning range ensures that no possible participating devices are missed, and the coverage and user experience of the audio sharing network are improved. By authenticating the scanned devices, it is ensured that only authorized devices can join the audio sharing network, preventing unauthorized devices from accessing, and improving network security. A secure pairing connection is established to ensure the confidentiality and integrity of data transmission, avoid data leakage and tampering; through the verified device list, unnecessary interference and conflict can be avoided, and the stability and quality of audio transmission can be improved. The connection relationship between verified devices is obtained, and the communication path of each device is clarified, laying the foundation for building a stable network topology. The master and slave device roles are determined, network resources are reasonably allocated, and network efficiency and data transmission reliability are improved. Identifying the communication path between devices helps to optimize the data transmission path, reduce delays and transmission errors, and improve the overall transmission performance. By means of graphical representation, the device nodes and connection edges in the network are intuitively displayed, which is convenient for monitoring and managing the network topology. Adding device attribute information, such as device type, signal strength, etc., helps to further analyze and optimize network performance. The dynamic network topology map can be updated in real time to reflect changes in the network and the status of the device, improving the adaptability and flexibility of the network. Through link quality analysis, possible bottlenecks and weaknesses in the network can be identified, providing a basis for subsequent optimization. Estimation of the total available bandwidth, accurate assessment of the network's bandwidth resources, ensure the rationality and effectiveness of bandwidth allocation, and avoid resource waste and conflicts. Through accurate bandwidth assessment, the quality and stability of audio data transmission are guaranteed, and the user's auditory experience is improved. Small bandwidth channels are divided according to bandwidth resource assessment data and the number of user devices to ensure the optimal bandwidth utilization of each channel; unique identifiers are assigned to each channel to avoid interference between channels and improve the stability and quality of audio transmission; the division method of small bandwidth channels can flexibly adapt to different numbers of user devices, with good scalability and adaptability. An independent small-bandwidth channel is allocated to each user to ensure the independence and integrity of audio data transmission and avoid mutual interference; multiple parallel audio stream channels can support the audio transmission needs of multiple users at the same time, realize multi-user audio sharing, and improve user experience; through parallel transmission, the overall audio transmission efficiency and quality are improved, and transmission delay and packet loss rate are reduced.

[0021] Preferably, step S2 comprises the following steps:

[0022] Step S21: Analyze the bandwidth, signal-to-noise ratio, and bit error rate of each audio stream channel to obtain channel characteristic parameter data;

[0023] Step S22: constructing a multi-user interference model based on user space distribution according to the channel characteristic parameter data, thereby obtaining scenario model parameter data;

[0024] Step S23: Calculate the interference coefficient of each user to other users according to the scenario model parameter data, thereby generating a time-varying channel response matrix;

[0025] Step S24: using a singular value decomposition algorithm to improve the sparsity and computational efficiency of the time-varying channel response matrix, thereby obtaining a channel response matrix;

[0026] Step S25: designing an adaptive equalizer for compensating channel distortion and suppressing interference according to the channel response matrix, thereby obtaining equalizer coefficient data;

[0027] Step S26: Adaptively and dynamically adjust the audio transmitted in each small bandwidth channel according to the channel response matrix and the equalizer coefficient data to generate adaptive audio stream data.

[0028] The present invention comprehensively evaluates the performance and quality of the channel by analyzing the bandwidth, signal-to-noise ratio and bit error rate of each channel, and ensures the reliability of audio transmission; discovers potential problems in the channel in advance, such as insufficient bandwidth, low signal-to-noise ratio or high bit error rate, to provide a basis for subsequent optimization and adjustment; accurately understands the bandwidth and quality of the channel, which helps to optimize the allocation and utilization of bandwidth and improve overall transmission efficiency. By constructing a multi-user interference model, the interference between different users can be identified and quantified, providing data support for subsequent interference suppression and channel equalization; modeling based on user spatial distribution can more accurately reflect the interference situation in the actual use scenario, improve the accuracy and practicality of the model; the dynamic adjustment ability of the model parameter data can adapt to different use scenarios and environmental changes, and improve the stability and quality of audio transmission. By calculating the interference coefficient, the interference effect of each user on other users is quantified, providing a clear interference situation, and providing a basis for interference suppression and channel optimization; generating a time-varying channel response matrix, it can track and reflect the changes in the channel state in real time, and improve the dynamic adaptability and accuracy of the channel model; by accurately calculating and tracking the interference coefficient, it can more effectively suppress interference and improve the quality and stability of audio transmission. The singular value decomposition algorithm is used to perform sparse processing on the matrix, reduce the computational complexity, and improve the computational efficiency and real-time performance of the channel response matrix; sparse processing can simplify the channel model, reduce the amount of data and computing resources occupied, and improve the overall performance and response speed of the system; singular value decomposition can retain the key information in the channel response matrix, while removing redundant and noise data, and improving the accuracy and reliability of the channel model. Designing an adaptive equalizer based on the channel response matrix can effectively compensate for the distortion in channel transmission and improve the quality of audio signals; the adaptive equalizer can dynamically adjust parameters, effectively suppress interference between different users, and ensure the clarity and stability of audio transmission; through precise equalizer design and adjustment, the auditory effect of the audio signal is optimized and the user's audio experience is improved. Adaptive dynamic adjustment is performed according to the real-time changing channel response matrix and equalizer coefficient data to ensure the stability and continuity of audio transmission; dynamic adjustment can optimize the audio signal in real time according to the channel conditions, reduce distortion and interference, and improve the clarity and quality of audio; the adaptive dynamic adjustment mechanism can adapt to the complex transmission environment in multi-user audio sharing scenarios, ensuring that each user can obtain a high-quality audio experience.

[0029] Preferably, step S26 comprises the following steps:

[0030] Step S261: dynamically adjusting the transmission power of each user according to the channel response matrix and the equalizer coefficient data, thereby obtaining power allocation scheme data;

[0031] Step S262: Selecting an audio coding scheme for each user according to the channel quality in the channel response matrix and the available power in the power allocation scheme data, thereby obtaining a user coding scheme mapping table;

[0032] Step S263: performing noise reduction compression on the audio transmitted in each small bandwidth channel according to the user coding scheme mapping table, thereby obtaining noise-reduced audio data;

[0033] Step S264: Adaptively adjust the noise reduction audio data according to the channel response matrix, the equalizer coefficient data and the power allocation scheme data, so as to obtain adaptive audio stream data, wherein the adaptive adjustment includes power control, spectrum shaping and time domain equalization.

[0034] The present invention dynamically adjusts the transmission power according to the channel response and the equalizer data, can reasonably allocate power resources, avoid too high or too low power settings, and improve power utilization efficiency; by dynamically adjusting the power, the interference between different users is reduced, and the signal quality and transmission stability of the overall system are improved; the effective power allocation strategy can reduce unnecessary power consumption, extend the battery life of the device, and improve the user experience. According to the channel quality and power allocation scheme, the optimal audio coding scheme is selected to ensure that high-quality audio signals can be transmitted under different channel conditions; the appropriate coding scheme is selected to optimize the transmission efficiency and reduce the bandwidth occupancy under the premise of ensuring the sound quality; the adaptive coding selection mechanism can improve the system's adaptability to different channel conditions and enhance the robustness of audio transmission. Through the noise reduction compression technology, the noise in the transmission process is effectively removed, and the clarity and quality of the audio signal are improved; the noise reduction compression can reduce the redundant part of the audio data, reduce the bandwidth occupancy, and improve the transmission efficiency; in the scenario shared by multiple users, the noise reduction compression can ensure that each user can obtain clear audio and avoid the degradation of sound quality due to insufficient bandwidth. Through comprehensive adaptive adjustment of power control, spectrum shaping and time domain equalization, the audio signal can be optimized in multiple dimensions and the sound quality can be improved; the adaptive adjustment mechanism can respond to channel changes in real time, ensure the continuity and stability of audio transmission, and avoid sound quality degradation caused by channel quality fluctuations; comprehensive optimization measures not only improve the quality of audio signals, but also adapt to the needs of different users and environments, providing a consistent and high-quality listening experience.

[0035] Preferably, step S3 comprises the following steps:

[0036] Step S31: Initializing the IEEE 1588PTP protocol stack on the device in the dynamic network topology diagram and configuring PTP parameters to obtain PTP initialization configuration data;

[0037] Step S32: Execute the best master clock algorithm according to the PTP initialization configuration data and the dynamic network topology diagram to select the best master clock source in the network, thereby obtaining the master clock device identification data;

[0038] Step S33: constructing a master-slave clock hierarchical structure based on the network topology according to the master clock device identification data and the dynamic network topology diagram, thereby obtaining a clock synchronization hierarchical diagram;

[0039] Step S34: exchanging PTP messages between the master and slave devices according to the clock synchronization hierarchy diagram, thereby obtaining original timestamp data;

[0040] Step S35: Calculate the network delay from each device to the master clock source according to the original timestamp data, thereby obtaining network delay estimation data;

[0041] Step S36: Delay compensation is performed on the adaptive audio stream data according to the network delay estimation data, thereby obtaining synchronous audio stream data.

[0042] The present invention initializes the IEEE 1588PTP protocol stack and configures PTP parameters, ensuring that all devices can use the same time source for synchronization, which helps to achieve accurate time synchronization; through precise PTP parameter configuration, the accuracy of time synchronization between devices can be significantly improved, and clock deviation can be reduced; PTP initialization is performed in a dynamic network topology, so that the system can adapt to changes in network topology and ensure the stability and reliability of time synchronization. By executing the best master clock algorithm, the best master clock source is selected to ensure the accuracy and stability of time synchronization in the network; selecting the best master clock source can improve the robustness of the system and reduce the time synchronization error caused by the instability of the clock source; selecting the master clock source according to the dynamic network topology enables the system to dynamically adapt to network changes and maintain the continuity of time synchronization. Construct a master-slave clock hierarchical structure, clarify the clock hierarchical relationship of each device in the network, and ensure the orderliness of time synchronization; by constructing a clock hierarchy diagram, optimize the time synchronization path, improve the synchronization efficiency, and reduce the delay of time synchronization; support a multi-level clock synchronization structure, adapt to complex network topologies, and improve the coverage and accuracy of time synchronization. By exchanging PTP messages, the original timestamp data of each device can be obtained in real time, providing basic data for subsequent delay calculation and time synchronization; through standardized PTP message exchange, the accuracy and consistency of timestamp data can be ensured, and synchronization errors can be reduced; frequent PTP message exchange can support high-frequency time synchronization requirements and improve the accuracy of time synchronization. According to the original timestamp data, the network delay from each device to the master clock source is accurately calculated to provide accurate delay data for subsequent time synchronization; through accurate delay calculation, the time synchronization error caused by network delay is reduced and the synchronization accuracy is improved; the delay estimation data can be dynamically adjusted to adapt to changes in network status and maintain the accuracy of time synchronization. Delay compensation is performed based on the network delay estimation data to ensure that the audio playback time on each device is consistent and synchronized playback is achieved; through accurate delay compensation, the delay difference in audio playback is eliminated to improve the user experience of audio sharing between multiple devices; the delay compensation is dynamically adjusted to adapt to changes in network delay to ensure the synchronization and stability of audio stream data.

[0043] Preferably, step S36 includes the following steps:

[0044] Step S361: Calculate the clock offset between each device and the master clock source according to the network delay estimation data and the original timestamp data, thereby obtaining clock offset value data;

[0045] Step S362: adjusting the local clock of each device to align with the master clock source according to the clock offset value data, thereby obtaining device clock calibration data;

[0046] Step S363: measuring the end-to-end transmission delay from the audio source to each receiving device according to the device clock calibration data and the dynamic network topology diagram, thereby obtaining transmission delay matrix data;

[0047] Step S364: Calculate the audio playback delay compensation value required by each device according to the transmission delay matrix data, thereby obtaining delay compensation value data;

[0048] Step S365: resample and adjust the buffer of the adaptive audio stream data according to the delay compensation value data to obtain synchronous audio stream data.

[0049] The present invention can accurately calibrate the device clock and reduce the time synchronization error by calculating the clock offset between each device and the main clock source; obtain accurate clock offset value, ensure the synchronization accuracy of the device clock, and reduce the delay difference during playback; calculate the clock offset in real time, adapt to changes in network conditions, and maintain continuous synchronization accuracy. Adjust the local clock of each device to eliminate the clock error between the device and the main clock source to ensure time synchronization; calibrate the device clock to ensure the time consistency of all devices in the system, improve the stability of the overall system, ensure the clock synchronization between multiple devices, and achieve high-precision collaborative work between multiple devices. Use the calibrated device clock to accurately measure the end-to-end transmission delay from the audio source to each receiving device and obtain accurate delay data; provide accurate delay estimation through detailed transmission delay measurement, reduce delay compensation error, provide a detailed transmission delay matrix, provide a basis for subsequent delay compensation, and ensure audio synchronous playback. According to the transmission delay of each device, calculate personalized playback delay compensation value to ensure audio playback synchronization on all devices; accurate delay compensation value calculation ensures the consistency of playback time of each device, eliminates the phenomenon of asynchronism, and provides seamless audio playback experience through delay compensation to improve the user's audio sharing experience. Through resampling and buffer adjustment, the audio streams on all devices are ensured to be played synchronously, realizing true multi-device audio sharing; through precise resampling and buffer adjustment, audio distortion is reduced, audio quality is maintained, audio streams are dynamically adjusted, and network conditions change, ensuring the stability and consistency of audio playback.

[0050] Preferably, step S4 comprises the following steps:

[0051] Step S41: designing an optimal transmission path based on the dynamic network topology diagram, establishing a virtual transmission network, and thus obtaining an audio data transmission network;

[0052] Step S42: segmenting and encoding the synchronous audio stream data into data packets, and assigning transmission priorities to the data packets according to the network status and user priorities in the dynamic network topology diagram, thereby obtaining a data packet queue with priorities;

[0053] Step S43: selecting and configuring a suitable transmission protocol according to the network characteristics of the audio data transmission network, and initializing the distributed consensus protocol, thereby obtaining the initial state data of the consensus protocol;

[0054] Step S44: broadcasting the data packets with priority to all participating devices in the network according to the priority of the data packet queue, and each receiving device sends a data packet reception confirmation information, thereby obtaining a reception confirmation state table;

[0055] Step S45: Execute the consensus protocol according to the reception confirmation status table and the consensus protocol initial status data, and vote on the integrity and sequence of the data packets to verify whether a consensus is reached, thereby obtaining a consistency verification report;

[0056] Step S46: reorganizing and decoding the data packet queue according to the consistency verification report to obtain consistent audio stream data;

[0057] Step S47: performing information verification based on the hash fingerprint on the consistent audio stream data to obtain hash verification result data;

[0058] Step S48: Selectively retransmit trigger or forward error correction is performed on the consistent audio stream data according to the hash verification result data to achieve multi-user sharing of audio.

[0059] The present invention designs the optimal transmission path according to the dynamic network topology diagram, ensures the efficient transmission of data packets in the network, reduces transmission delay and packet loss rate; establishes a virtual transmission network, can flexibly adjust the transmission path, improve network resource utilization, and enhance overall transmission efficiency; the optimal path design and the use of virtual networks improve the reliability and stability of the system, and ensure the smooth transmission of audio data. The audio stream data is segmented and encoded into packets to facilitate management and control, ensuring the orderliness and integrity of data transmission; the transmission priority is allocated according to the network status and user priority to ensure that key data is transmitted first, optimize resource usage, and the priority packet queue can effectively reduce network congestion and improve the stability and quality of audio data transmission. Select and configure a suitable transmission protocol to ensure that audio data can be efficiently transmitted under different network conditions and improve the transmission success rate; initialize the distributed consensus protocol to provide a basis for subsequent data consistency verification and ensure the reliability of data transmission. The initial state data of the consensus protocol ensures that each device works together on the same basis to improve the coordination of the overall system. Broadcast data packets ensure that all participating devices can receive audio data, enhancing the coverage of audio sharing; receiving confirmation information provides real-time feedback on data packet transmission, facilitating monitoring and management, timely discovering and solving transmission problems, and receiving confirmation status table helps confirm the reception status of data packets to ensure the reliability and integrity of data transmission. Through consensus protocol execution and voting mechanism, the integrity and order of data packets are verified to ensure the consistency of data transmission. The consistency verification report provides the verification results of the transmitted data, improving the security and credibility of data transmission. Through the verification of the consensus mechanism, errors and losses in the data transmission process are reduced, and the accuracy of data transmission is improved. Data packets are reorganized according to the consistency verification report to ensure the correct order and integrity of audio data; the reorganized data packets are decoded to restore the original audio stream data to ensure the quality and consistency of audio data; through reorganization and decoding, consistent audio stream data is provided to enhance the user's audio sharing experience. Consistent audio stream data is verified through hash fingerprints to ensure that the data has not been tampered with during transmission and ensure data integrity. The hash verification result data provides data verification information to improve the security of data transmission. The verification method based on hash fingerprints is fast and can detect data integrity in real time, reducing verification time. Based on the hash verification results, data retransmission or forward error correction is performed selectively to ensure the correct transmission of audio data. Selective retransmission and forward error correction improve the reliability of data transmission, reduce data loss and errors, and optimize the use of network resources through selective retransmission, reduce unnecessary retransmissions, and improve transmission efficiency.

[0060] The present invention also provides a multi-user audio sharing system based on a le-audio Bluetooth headset, which is used to execute the multi-user audio sharing method based on a le-audio Bluetooth headset. The multi-user audio sharing system based on a le-audio Bluetooth headset comprises:

[0061] The network discovery and channel allocation module is used to obtain the connection status and topology of LE-Audio devices in real time through the network discovery function of the Bluetooth protocol stack, thereby generating a dynamic network topology map of Bluetooth; based on the dynamic network topology map, the available bandwidth is divided into multiple small-bandwidth channels, and an independent small-bandwidth channel is allocated to each user for audio transmission, forming multiple parallel audio stream channels;

[0062] The audio adaptation module is used to model the multi-user audio sharing scenario according to the audio stream channel, and obtain a channel response matrix describing the multi-user audio transmission characteristics; according to the channel response matrix, the audio transmitted in each small bandwidth channel is adaptively and dynamically adjusted to generate adaptive audio stream data;

[0063] The clock synchronization and delay compensation module is used to synchronize the clocks between the devices participating in the sharing according to the dynamic network topology diagram using the IEEE 1588PTP protocol, measure the transmission delay from each device to the audio source, and perform delay compensation on the adaptive audio stream data, thereby obtaining the synchronous audio stream data;

[0064] The consistency transmission and error correction module is used to build an audio data transmission network based on a dynamic network topology diagram, and to perform consistency confirmation transmission of synchronous audio stream data through a consensus mechanism, thereby obtaining consistent audio stream data; to perform information verification based on hash fingerprints on the consistent audio stream data to obtain hash verification result data, and to perform selective retransmission triggering or forward error correction on the consistent audio stream data based on the hash verification result data to achieve multi-user sharing of audio.

[0065] The present invention obtains the device connection status and topology in real time, ensures the timely update and adjustment of network information, improves the flexibility and adaptability of the system, divides the available bandwidth into multiple small bandwidth channels, realizes the refined management and efficient use of bandwidth, ensures the stability and high quality of audio transmission, and allocates independent small bandwidth channels to each user to form multiple parallel audio stream channels, effectively avoids mutual interference between channels, and improves the clarity and integrity of audio transmission. By modeling the multi-user audio sharing scenario, the audio transmission characteristics are accurately described, and a scientific basis is provided for channel adjustment and optimization; the audio transmission is dynamically adjusted according to the channel response matrix, and the network changes can be responded to in real time, ensuring the stability and efficiency of audio data transmission. The generation of adaptive audio stream data improves the audio transmission quality, reduces distortion and interference, and improves the user experience. The IEEE 1588PTP protocol is used to achieve high-precision clock synchronization between devices, ensuring that the audio data is played at the same time on each device to avoid the problem of asynchronous; the transmission delay is measured and calculated, and delay compensation is performed to ensure that the transmission and playback delay of audio data is minimized, and the audio synchronization and consistency are improved; the combination of clock synchronization and delay compensation improves the coordination and consistency of audio playback, ensuring the high-quality experience of multi-user audio sharing. Consistency confirmation is performed through a consensus mechanism to ensure that the audio stream data remains consistent during transmission, improve the reliability of data transmission, and verify information based on hash fingerprints to ensure the integrity and security of data during transmission, prevent data tampering, and perform selective retransmission or forward error correction based on the hash verification results to improve the correctness and success rate of data transmission and ensure the high quality and continuity of audio sharing. These four modules together build an efficient and reliable multi-user audio sharing system. The network discovery and channel allocation module ensures the optimization of device connection and bandwidth utilization; the audio adaptation module improves the stability and quality of audio transmission through dynamic adjustment; the clock synchronization and delay compensation module ensures the synchronization and consistency of audio; the consistency transmission and error correction module improves the reliability and security of transmission through data verification and error correction. Through the collaborative work of these modules, the system can provide high-quality audio sharing services in a complex multi-user environment. BRIEF DESCRIPTION OF THE DRAWINGS

[0066] Other features, objects and advantages of the present invention will become more apparent from the detailed description of non-limiting embodiments thereof made with reference to the following drawings:

[0067] Figure 1 It is a schematic diagram of the steps of the multi-user audio sharing method based on the le-audio Bluetooth headset of the present invention;

[0068] Figure 2 for Figure 1 Detailed step flow diagram of step S1;

[0069] Figure 3 for Figure 1 Detailed step flow chart of step S2 in FIG. DETAILED DESCRIPTION

[0070] The technical method of the present invention is described clearly and completely below in conjunction with the accompanying drawings. Obviously, the described embodiments are part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by technicians in this field without creative work are within the scope of protection of the present invention.

[0071] In addition, the accompanying drawings are only schematic illustrations of the present invention and are not necessarily drawn to scale. The same reference numerals in the figures represent the same or similar parts, and their repeated description will be omitted. Some of the block diagrams shown in the accompanying drawings are functional entities and do not necessarily correspond to physically or logically independent entities. The functional entities can be implemented in software form, or implemented in one or more hardware modules or integrated circuits, or implemented in different networks and / or processor methods and / or microcontroller methods.

[0072] It should be understood that, although the terms "first", "second", etc. may be used herein to describe various units, these units should not be limited by these terms. These terms are used only to distinguish one unit from another unit. For example, without departing from the scope of the exemplary embodiments, the first unit may be referred to as the second unit, and similarly the second unit may be referred to as the first unit. The term "and / or" used herein includes any and all combinations of one or more of the listed associated items.

[0073] To achieve this, please refer to Figures 1 to 3 The present invention provides a multi-user audio sharing method based on le-audio Bluetooth headset, the method comprising the following steps:

[0074] Step S1: The connection status and topology of the LE-Audio device are obtained in real time through the network discovery function of the Bluetooth protocol stack, thereby generating a dynamic network topology map of Bluetooth; based on the dynamic network topology map, the available bandwidth is divided into multiple small bandwidth channels, and an independent small bandwidth channel is allocated to each user for audio transmission, forming multiple parallel audio stream channels;

[0075] Step S2: Modeling a multi-user audio sharing scenario according to the audio stream channel to obtain a channel response matrix describing the multi-user audio transmission characteristics; adaptively and dynamically adjusting the audio transmitted in each small bandwidth channel according to the channel response matrix to generate adaptive audio stream data;

[0076] Step S3: using the IEEE 1588PTP protocol to synchronize the clocks between the devices participating in the sharing according to the dynamic network topology, measuring the transmission delay from each device to the audio source, and performing delay compensation on the adaptive audio stream data, thereby obtaining synchronous audio stream data;

[0077] Step S4: construct an audio data transmission network according to the dynamic network topology diagram, and perform consistency confirmation transmission on the synchronous audio stream data through a consensus mechanism to obtain consistent audio stream data; perform information verification based on hash fingerprint on the consistent audio stream data to obtain hash verification result data, and perform selective retransmission triggering or forward error correction on the consistent audio stream data according to the hash verification result data to achieve multi-user sharing of audio.

[0078] In the embodiment of the present invention, reference Figure 1 The present invention is a schematic diagram of the steps of a multi-user audio sharing method based on a le-audio Bluetooth headset. In this example, the multi-user audio sharing method based on a le-audio Bluetooth headset includes the following steps:

[0079] Step S1: The connection status and topology of the LE-Audio device are obtained in real time through the network discovery function of the Bluetooth protocol stack, thereby generating a dynamic network topology map of Bluetooth; based on the dynamic network topology map, the available bandwidth is divided into multiple small bandwidth channels, and an independent small bandwidth channel is allocated to each user for audio transmission, forming multiple parallel audio stream channels;

[0080] The embodiment of the present invention uses the network discovery function of the Bluetooth protocol stack (such as GAP, Generic Access Profile) to scan the surrounding LE-Audio devices and obtain the connection status of the devices in real time; through the connection relationship between the devices, a dynamic network topology diagram of the Bluetooth device is generated to display the device nodes and connection edges. According to the network topology diagram and the total available bandwidth, the bandwidth is divided into multiple small bandwidth channels; for example, if the total bandwidth is 2MHz, it can be divided into 20 small bandwidth channels of 100kHz. An independent small bandwidth channel is allocated to each user to ensure that each channel transmits audio data, forming multiple parallel audio stream channels, and using algorithms such as time division multiple access (TDMA) or frequency division multiple access (FDMA) for allocation to ensure independence and interference-free between channels.

[0081] Step S2: Modeling a multi-user audio sharing scenario according to the audio stream channel to obtain a channel response matrix describing the multi-user audio transmission characteristics; adaptively and dynamically adjusting the audio transmitted in each small bandwidth channel according to the channel response matrix to generate adaptive audio stream data;

[0082] The embodiment of the present invention models the multi-user audio sharing scenario, considers factors such as the number of users, device location, environmental noise, etc., and generates a channel response matrix that describes the multi-user audio transmission characteristics. The channel bandwidth, signal-to-noise ratio (SNR) and bit error rate (BER) are analyzed to obtain channel characteristic parameter data. According to the channel response matrix, the audio stream is adaptively and dynamically adjusted; for example, the gain and filtering parameters of the audio signal are adjusted according to the real-time channel conditions using an adaptive equalizer and dynamic power control technology; through noise reduction, dynamic range compression and other processing, audio stream data that adapts to the current channel conditions is generated to improve the quality and stability of audio transmission.

[0083] Step S3: using the IEEE 1588PTP protocol to synchronize the clocks between the devices participating in the sharing according to the dynamic network topology, measuring the transmission delay from each device to the audio source, and performing delay compensation on the adaptive audio stream data, thereby obtaining synchronous audio stream data;

[0084] The embodiment of the present invention initializes the IEEE 1588PTP protocol stack on each device and configures relevant parameters, such as the role configuration of the master clock device and the slave clock device; uses the PTP protocol to exchange synchronization messages between the master and slave devices, measures and calculates the clock offset and network delay, and performs clock synchronization adjustment; calculates the transmission delay from each device to the audio source based on the timestamp data in the synchronization message, and generates network delay estimation data. Based on the channel response matrix and the network delay estimation data, delay compensation is performed on the adaptive audio stream data, and the buffering and resampling technology is used to adjust the audio playback time to ensure the synchronization of audio playback of each device.

[0085] Step S4: construct an audio data transmission network according to the dynamic network topology diagram, and perform consistency confirmation transmission on the synchronous audio stream data through a consensus mechanism to obtain consistent audio stream data; perform information verification based on hash fingerprint on the consistent audio stream data to obtain hash verification result data, and perform selective retransmission triggering or forward error correction on the consistent audio stream data according to the hash verification result data to achieve multi-user sharing of audio.

[0086] According to the dynamic network topology diagram, the embodiment of the present invention designs the optimal transmission path, establishes a virtual audio data transmission network, selects a suitable transmission protocol (such as RTP, Real-time Transport Protocol) and configures parameters. The synchronous audio stream data is segmented and encoded, and the transmission priority is assigned to the data packet according to the network status and user priority to form a data packet queue with priority. Initialize the distributed consensus protocol (such as Paxos or Raft), broadcast the data packet with priority to all participating devices in the network, and each receiving device sends a reception confirmation message to generate a reception confirmation status table. According to the reception confirmation status table and the consensus protocol, the integrity and order of the data packet are voted and verified to ensure the consistency of the data packet and generate a consistency verification report. According to the consistency verification report, the data packet queue is reorganized and decoded to generate consistent audio stream data. The consistent audio stream data is verified based on the hash fingerprint information, and the selective retransmission or forward error correction is performed according to the information verification result to ensure the integrity and reliability of the audio data and realize multi-user sharing of audio.

[0087] The present invention utilizes the network discovery function of the Bluetooth protocol stack to obtain the connection status and topology of the LE-Audio device in real time, ensuring the dynamics and accuracy of the network. The available bandwidth is divided into multiple small bandwidth channels, and an independent channel is allocated to each user, which can avoid channel interference, optimize bandwidth utilization, and ensure the stability and integrity of audio transmission. By forming multiple parallel audio stream channels, multiple users are supported to perform audio transmission at the same time, meet the audio sharing needs of multiple users, and improve the scalability and flexibility of the system. The multi-user audio sharing scenario is modeled to obtain a channel response matrix, which can accurately describe the multi-user audio transmission characteristics and improve the prediction and optimization capabilities of audio transmission. According to the channel response matrix, the audio is adaptively and dynamically adjusted to ensure high quality and low latency of audio transmission, adapt to transmission requirements in different environments, and enhance user experience. Through the adaptive adjustment mechanism, the transmission path and transmission parameters of audio stream data are optimized, the transmission efficiency and stability are improved, and the transmission delay and error are reduced. The IEEE 1588PTP protocol can achieve accurate clock synchronization between devices participating in sharing, ensure that each device works under the same time reference, and avoid audio asynchrony problems caused by clock deviation. By measuring the transmission delay from each device to the audio source and performing delay compensation, the delay difference between devices can be eliminated, the synchronous transmission of audio streams can be achieved, and the synchronous audio playback between multiple devices can be ensured. Synchronous audio stream data ensures the time consistency of audio playback on each device, avoids audio delay and distortion caused by asynchronous audio playback on multiple devices, and improves the user experience in multi-user sharing scenarios. An audio data transmission network is constructed based on a dynamic network topology diagram to ensure the optimal path of audio data transmission and the stability of the network structure, thereby improving the reliability and efficiency of data transmission. The synchronous audio stream data is transmitted with consistency confirmation through a consensus mechanism to ensure that the consistent audio stream data received by each device is the same, thereby avoiding data inconsistency during data transmission. Information verification based on hash fingerprints can effectively detect errors and tampering in audio data transmission to ensure data integrity and security. Selective retransmission triggering or forward error correction mechanism can promptly repair transmission errors and improve the reliability and quality of audio data transmission. Through the combination of the above steps and mechanisms, multi-user sharing of audio is achieved, ensuring the consistency and high quality of audio experience of multiple users in sharing scenarios.

[0088] Preferably, step S1 comprises the following steps:

[0089] Step S11: Scan the surrounding LE-Audio devices using the scanning function of the Bluetooth protocol stack, and collect the scanned device information to obtain the original device scanning data;

[0090] Step S12: authenticating the scanned device according to the original device scan data and establishing a secure pairing connection, thereby obtaining a list of verified devices;

[0091] Step S13: obtaining the connection relationship between devices in the verified device list through the network discovery function of the Bluetooth protocol stack, determining the master and slave device roles, identifying the communication path between devices, and obtaining initial topology structure data;

[0092] Step S14: establishing a graphical representation of device nodes and connection edges according to the initial topology data, and adding device attribute information, thereby obtaining a dynamic network topology diagram of Bluetooth;

[0093] Step S15: performing link quality analysis on the dynamic network topology diagram and estimating the total available bandwidth, thereby obtaining bandwidth resource evaluation data;

[0094] Step S16: performing small bandwidth channel division according to the bandwidth resource evaluation data and the number of user equipments, and assigning a unique identifier to each channel, thereby obtaining channel division data;

[0095] Step S17: allocating an independent small-bandwidth channel for audio transmission to each user according to the channel division data, thereby forming multiple parallel audio stream channels.

[0096] As an embodiment of the present invention, refer to Figure 2 As shown, Figure 1 Detailed step flow diagram of step S1 in the embodiment of the present invention, step S1 includes the following steps:

[0097] Step S11: Scan the surrounding LE-Audio devices using the scanning function of the Bluetooth protocol stack, and collect the scanned device information to obtain the original device scanning data;

[0098] The embodiment of the present invention uses the scanning function provided by the Bluetooth protocol stack to set scanning parameters such as the scanning window and the scanning interval; executes the device scanning command to start searching for the surrounding LE-Audio devices. When the device is scanned, the unique identifier (such as MAC address), signal strength (RSSI), device type and other information of each device are obtained; this information is stored in a data structure, such as a list or a database, as part of the original device scanning data.

[0099] Step S12: authenticating the scanned device according to the original device scan data and establishing a secure pairing connection, thereby obtaining a list of verified devices;

[0100] The embodiments of the present invention use the collected device information to perform identity authentication to ensure the legitimacy and security of each device; the authentication process may include checking the device's digital certificate, PIN code verification or other security mechanisms. A secure pairing connection is established for the authenticated device; this may involve the authentication stage of the Bluetooth pairing process, such as entering a PIN code or confirming with a physical button.

[0101] Step S13: obtaining the connection relationship between devices in the verified device list through the network discovery function of the Bluetooth protocol stack, determining the master and slave device roles, identifying the communication path between devices, and obtaining initial topology structure data;

[0102] The embodiment of the present invention utilizes the network discovery function of the Bluetooth protocol stack to discover the connection relationship between devices based on the verified device list; determines the role of each device in the network, such as master device or slave device, and establishes a communication path between devices. A graphical representation of device nodes and connection edges is established, and graph theory or similar data structures are used to describe the physical connection and logical relationship between devices; attribute information is added to each device node, such as device type, MAC address, connection status, etc.

[0103] Step S14: establishing a graphical representation of device nodes and connection edges according to the initial topology data, and adding device attribute information, thereby obtaining a dynamic network topology diagram of Bluetooth;

[0104] The embodiment of the present invention uses a graphical representation (such as a graph or adjacency matrix) to establish a Bluetooth dynamic network topology graph based on the initial topology data; each device node represents an LE-Audio device, and each connection edge represents a communication link between devices. Detailed attribute information is added to each device node, including the physical location of the device, signal strength, supported audio codecs, etc. These attribute information are crucial for subsequent link quality analysis and bandwidth allocation.

[0105] Step S15: performing link quality analysis on the dynamic network topology diagram and estimating the total available bandwidth, thereby obtaining bandwidth resource evaluation data;

[0106] The embodiment of the present invention performs link quality analysis on each connection edge in the topology diagram, including signal strength (RSSI), packet loss rate, signal-to-noise ratio (SNR), etc., and evaluates the reliability and transmission performance of each connection based on the analysis results. Combined with the link quality analysis data and device quantity information, the available bandwidth resources for each small bandwidth channel are calculated, and the maximum transmission rate and stability of each channel are determined for subsequent audio traffic allocation.

[0107] Step S16: performing small bandwidth channel division according to the bandwidth resource evaluation data and the number of user equipments, and assigning a unique identifier to each channel, thereby obtaining channel division data;

[0108] The embodiment of the present invention divides the total available bandwidth into multiple small bandwidth channels according to the bandwidth resource evaluation data; each channel has a certain bandwidth capacity suitable for specific audio transmission requirements. A unique identifier or frequency range is assigned to each small bandwidth channel to ensure independence and non-interference between channels in a dynamic network environment, and these identifiers are used for subsequent audio traffic allocation and management.

[0109] Step S17: allocating an independent small-bandwidth channel for audio transmission to each user according to the channel division data, thereby forming multiple parallel audio stream channels.

[0110] The embodiment of the present invention allocates an independent small-bandwidth channel to each user or device based on the channel division data obtained in the previous steps; each user transmits audio through the allocated channel, ensuring data isolation and stable performance between users. The allocated independent small-bandwidth channels are used to form parallel audio stream transmission channels for multiple users or devices; these parallel audio stream channels support multiple users to share and transmit audio at the same time, optimizing the overall efficiency of the system and user experience.

[0111] The present invention can discover the surrounding LE-Audio devices in real time through the scanning function of the Bluetooth protocol stack, ensuring that all devices can participate in the audio sharing network. The scanned device information, including device type, signal strength, etc., is collected to provide basic data for subsequent device identity authentication and network topology construction. The wide scanning range ensures that no possible participating devices are missed, and the coverage and user experience of the audio sharing network are improved. By authenticating the scanned devices, it is ensured that only authorized devices can join the audio sharing network, preventing unauthorized devices from accessing, and improving network security. A secure pairing connection is established to ensure the confidentiality and integrity of data transmission, avoid data leakage and tampering; through the verified device list, unnecessary interference and conflict can be avoided, and the stability and quality of audio transmission can be improved. The connection relationship between verified devices is obtained, and the communication path of each device is clarified, laying the foundation for building a stable network topology. The master and slave device roles are determined, network resources are reasonably allocated, and network efficiency and data transmission reliability are improved. Identifying the communication path between devices helps to optimize the data transmission path, reduce delays and transmission errors, and improve the overall transmission performance. By means of graphical representation, the device nodes and connection edges in the network are intuitively displayed, which is convenient for monitoring and managing the network topology. Adding device attribute information, such as device type, signal strength, etc., helps to further analyze and optimize network performance. The dynamic network topology map can be updated in real time to reflect changes in the network and the status of the device, improving the adaptability and flexibility of the network. Through link quality analysis, possible bottlenecks and weaknesses in the network can be identified, providing a basis for subsequent optimization. Estimation of the total available bandwidth, accurate assessment of the network's bandwidth resources, ensure the rationality and effectiveness of bandwidth allocation, and avoid resource waste and conflicts. Through accurate bandwidth assessment, the quality and stability of audio data transmission are guaranteed, and the user's auditory experience is improved. Small bandwidth channels are divided according to bandwidth resource assessment data and the number of user devices to ensure the optimal bandwidth utilization of each channel; unique identifiers are assigned to each channel to avoid interference between channels and improve the stability and quality of audio transmission; the division method of small bandwidth channels can flexibly adapt to different numbers of user devices, with good scalability and adaptability. An independent small-bandwidth channel is allocated to each user to ensure the independence and integrity of audio data transmission and avoid mutual interference; multiple parallel audio stream channels can support the audio transmission needs of multiple users at the same time, realize multi-user audio sharing, and improve user experience; through parallel transmission, the overall audio transmission efficiency and quality are improved, and transmission delay and packet loss rate are reduced.

[0112] Preferably, step S2 comprises the following steps:

[0113] Step S21: Analyze the bandwidth, signal-to-noise ratio, and bit error rate of each audio stream channel to obtain channel characteristic parameter data;

[0114] Step S22: constructing a multi-user interference model based on user space distribution according to the channel characteristic parameter data, thereby obtaining scenario model parameter data;

[0115] Step S23: Calculate the interference coefficient of each user to other users according to the scenario model parameter data, thereby generating a time-varying channel response matrix;

[0116] Step S24: using a singular value decomposition algorithm to improve the sparsity and computational efficiency of the time-varying channel response matrix, thereby obtaining a channel response matrix;

[0117] Step S25: designing an adaptive equalizer for compensating channel distortion and suppressing interference according to the channel response matrix, thereby obtaining equalizer coefficient data;

[0118] Step S26: Adaptively and dynamically adjust the audio transmitted in each small bandwidth channel according to the channel response matrix and the equalizer coefficient data to generate adaptive audio stream data.

[0119] As an embodiment of the present invention, refer to Figure 3 As shown, Figure 1 Detailed step flow diagram of step S2 in the embodiment of the present invention, step S2 includes the following steps:

[0120] Step S21: Analyze the bandwidth, signal-to-noise ratio, and bit error rate of each audio stream channel to obtain channel characteristic parameter data;

[0121] The embodiment of the present invention measures and analyzes the bandwidth of each small bandwidth channel, using appropriate tools and algorithms, such as spectrum analysis or measurement through data packet transmission rate, to obtain the bandwidth parameters of each channel; determine the maximum data transmission rate that each channel can support for subsequent audio traffic adjustment and allocation. Measure the signal strength and background noise level of each channel, and calculate the signal-to-noise ratio, which can be obtained through the signal strength indication (RSSI) and environmental noise measurement at the receiving end; analysis of the signal-to-noise ratio helps evaluate the noise conditions of the channel to determine the clarity of the signal and the transmission quality. Test the bit error rate during data transmission on each channel, which is measured by sending a known data pattern and verifying the consistency of the received data with the original data at the receiving end; analyzing the bit error rate can evaluate the reliability of the channel and the accuracy of data transmission.

[0122] Step S22: constructing a multi-user interference model based on user space distribution according to the channel characteristic parameter data, thereby obtaining scenario model parameter data;

[0123] The embodiment of the present invention establishes a multi-user interference model based on the spatial distribution of user devices in the scene, taking into account factors such as the location between devices, signal strength attenuation, reflection and multipath effects; mathematical modeling techniques (such as probability distribution, geometric model or statistical analysis) are used to derive the interference relationship between users. The scene model parameters are obtained from actual measurement data or simulation results, and these parameters include information such as the location coordinates, transmission power, and antenna direction of the user device.

[0124] Step S23: Calculate the interference coefficient of each user to other users according to the scenario model parameter data, thereby generating a time-varying channel response matrix;

[0125] The embodiment of the present invention calculates the interference coefficient of each user to other users based on the scenario model parameters and the location distribution of the user equipment; the interference coefficient is usually estimated based on the transmission distance, transmission power and signal propagation model, reflecting the interference intensity between devices. A time-varying channel response matrix is ​​constructed based on the calculated interference coefficient, which describes the characteristics of signal transmission between each pair of devices, taking into account signal attenuation, multipath effects and dynamic interference changes.

[0126] Step S24: using a singular value decomposition algorithm to improve the sparsity and computational efficiency of the time-varying channel response matrix, thereby obtaining a channel response matrix;

[0127] The embodiment of the present invention uses a singular value decomposition algorithm to decompose the time-varying channel response matrix to improve the sparsity and computational efficiency of the matrix; SVD can decompose the channel response matrix into singular values ​​and left and right singular vectors, thereby simplifying the analysis and processing of channel characteristics.

[0128] Step S25: designing an adaptive equalizer for compensating channel distortion and suppressing interference according to the channel response matrix, thereby obtaining equalizer coefficient data;

[0129] The embodiment of the present invention designs an adaptive equalizer to compensate for channel distortion and suppress interference based on the optimized channel response matrix. The design of the equalizer takes into account the frequency response, transmission delay and noise characteristics of the channel, and achieves signal optimization by adjusting the equalizer coefficients; the coefficients of the equalizer are calculated and optimized to adapt to dynamically changing channel conditions and differences between different user devices.

[0130] Step S26: Adaptively and dynamically adjust the audio transmitted in each small bandwidth channel according to the channel response matrix and the equalizer coefficient data to generate adaptive audio stream data.

[0131] The embodiment of the present invention adaptively adjusts the audio data transmitted in each small bandwidth channel according to the channel response matrix and the equalizer coefficient; the adjustment process includes dynamically changing transmission power, spectrum shaping and time domain equalization to optimize the transmission quality and reception effect of the audio signal. The adaptively adjusted audio data stream is generated as the final adaptive audio stream data for subsequent audio transmission and sharing scenarios.

[0132] The present invention comprehensively evaluates the performance and quality of the channel by analyzing the bandwidth, signal-to-noise ratio and bit error rate of each channel, and ensures the reliability of audio transmission; discovers potential problems in the channel in advance, such as insufficient bandwidth, low signal-to-noise ratio or high bit error rate, to provide a basis for subsequent optimization and adjustment; accurately understands the bandwidth and quality of the channel, which helps to optimize the allocation and utilization of bandwidth and improve overall transmission efficiency. By constructing a multi-user interference model, the interference between different users can be identified and quantified, providing data support for subsequent interference suppression and channel equalization; modeling based on user spatial distribution can more accurately reflect the interference situation in the actual use scenario, improve the accuracy and practicality of the model; the dynamic adjustment ability of the model parameter data can adapt to different use scenarios and environmental changes, and improve the stability and quality of audio transmission. By calculating the interference coefficient, the interference effect of each user on other users is quantified, providing a clear interference situation, and providing a basis for interference suppression and channel optimization; generating a time-varying channel response matrix, it can track and reflect the changes in the channel state in real time, and improve the dynamic adaptability and accuracy of the channel model; by accurately calculating and tracking the interference coefficient, it can more effectively suppress interference and improve the quality and stability of audio transmission. The singular value decomposition algorithm is used to perform sparse processing on the matrix, reduce the computational complexity, and improve the computational efficiency and real-time performance of the channel response matrix; sparse processing can simplify the channel model, reduce the amount of data and computing resources occupied, and improve the overall performance and response speed of the system; singular value decomposition can retain the key information in the channel response matrix, while removing redundant and noise data, and improving the accuracy and reliability of the channel model. Designing an adaptive equalizer based on the channel response matrix can effectively compensate for the distortion in channel transmission and improve the quality of audio signals; the adaptive equalizer can dynamically adjust parameters, effectively suppress interference between different users, and ensure the clarity and stability of audio transmission; through precise equalizer design and adjustment, the auditory effect of the audio signal is optimized and the user's audio experience is improved. Adaptive dynamic adjustment is performed according to the real-time changing channel response matrix and equalizer coefficient data to ensure the stability and continuity of audio transmission; dynamic adjustment can optimize the audio signal in real time according to the channel conditions, reduce distortion and interference, and improve the clarity and quality of audio; the adaptive dynamic adjustment mechanism can adapt to the complex transmission environment in multi-user audio sharing scenarios, ensuring that each user can obtain a high-quality audio experience.

[0133] Preferably, step S26 comprises the following steps:

[0134] Step S261: dynamically adjusting the transmission power of each user according to the channel response matrix and the equalizer coefficient data, thereby obtaining power allocation scheme data;

[0135] The embodiment of the present invention uses the previously calculated channel response matrix, including information such as signal strength, multipath effect and interference level, to analyze the characteristics of each small bandwidth channel, such as signal attenuation rate and background noise level. Considering the adaptive equalizer coefficients designed in the previous steps, these coefficients can be adjusted according to the channel characteristics and the differences between devices to determine the optimal transmission power range for each user device to ensure the stability and quality of the signal during transmission. Based on the channel response matrix and equalizer coefficient data, a dynamic power adjustment strategy is implemented, which includes adjusting the transmission power of each user device according to the channel quality to maximize the signal coverage and minimize the interference effect; using an adaptive algorithm or feedback control mechanism to adjust the transmission power in real time to adapt to changes in the network environment.

[0136] Step S262: Selecting an audio coding scheme for each user according to the channel quality in the channel response matrix and the available power in the power allocation scheme data, thereby obtaining a user coding scheme mapping table;

[0137] The embodiment of the present invention determines the transmission capacity and reliability of each small bandwidth channel based on the signal quality information in the channel response matrix, including signal-to-noise ratio, bit error rate, signal attenuation rate, etc., so as to select a suitable audio coding scheme. Considering the power allocation scheme data generated in step S261, the maximum transmission power range available for each small bandwidth channel is determined, and according to the available power and channel quality, an appropriate audio coding scheme is selected to maximize the transmission efficiency and audio quality.

[0138] Step S263: performing noise reduction compression on the audio transmitted in each small bandwidth channel according to the user coding scheme mapping table, thereby obtaining noise-reduced audio data;

[0139] The embodiment of the present invention allocates a suitable audio coding scheme to each small bandwidth channel according to the user coding scheme mapping table generated in step S262; the coding scheme should take into account channel characteristics, bandwidth limitations and user priorities to achieve the best audio transmission effect. Use a suitable noise reduction compression algorithm, such as digital signal processing technology based on signal spectrum characteristics, to reduce noise and distortion during transmission and improve audio clarity and reception quality.

[0140] Step S264: Adaptively adjust the noise reduction audio data according to the channel response matrix, the equalizer coefficient data and the power allocation scheme data, so as to obtain adaptive audio stream data, wherein the adaptive adjustment includes power control, spectrum shaping and time domain equalization.

[0141] The embodiment of the present invention combines the channel response matrix and equalizer coefficient data in steps S261 and S263 to implement adaptive adjustment, which includes real-time power control, spectrum shaping and time domain equalization to adapt to the dynamic changes of the channel and the differences between different user devices. The audio data that has been processed by noise reduction and compression is combined with the adaptively adjusted transmission power and coding scheme to generate the final adaptive audio stream data, which is prepared for subsequent audio transmission and multi-user sharing scenarios to ensure high-quality audio transmission experience under different network conditions.

[0142] The present invention dynamically adjusts the transmission power according to the channel response and the equalizer data, can reasonably allocate power resources, avoid too high or too low power settings, and improve power utilization efficiency; by dynamically adjusting the power, the interference between different users is reduced, and the signal quality and transmission stability of the overall system are improved; the effective power allocation strategy can reduce unnecessary power consumption, extend the battery life of the device, and improve the user experience. According to the channel quality and power allocation scheme, the optimal audio coding scheme is selected to ensure that high-quality audio signals can be transmitted under different channel conditions; the appropriate coding scheme is selected to optimize the transmission efficiency and reduce the bandwidth occupancy under the premise of ensuring the sound quality; the adaptive coding selection mechanism can improve the system's adaptability to different channel conditions and enhance the robustness of audio transmission. Through the noise reduction compression technology, the noise in the transmission process is effectively removed, and the clarity and quality of the audio signal are improved; the noise reduction compression can reduce the redundant part of the audio data, reduce the bandwidth occupancy, and improve the transmission efficiency; in the scenario shared by multiple users, the noise reduction compression can ensure that each user can obtain clear audio and avoid the degradation of sound quality due to insufficient bandwidth. Through comprehensive adaptive adjustment of power control, spectrum shaping and time domain equalization, the audio signal can be optimized in multiple dimensions and the sound quality can be improved; the adaptive adjustment mechanism can respond to channel changes in real time, ensure the continuity and stability of audio transmission, and avoid sound quality degradation caused by channel quality fluctuations; comprehensive optimization measures not only improve the quality of audio signals, but also adapt to the needs of different users and environments, providing a consistent and high-quality listening experience.

[0143] Preferably, step S3 comprises the following steps:

[0144] Step S31: Initializing the IEEE 1588PTP protocol stack on the device in the dynamic network topology diagram and configuring PTP parameters to obtain PTP initialization configuration data;

[0145] The embodiment of the present invention initializes the IEEE 1588 Precision Time Protocol (PTP) protocol stack on each device in the dynamic network topology diagram; this includes loading and configuring the PTP software, initializing the hardware clock, and setting the network interface parameters. According to the system requirements and network characteristics, the parameters of PTP are configured, such as clock accuracy, transmission interval, clock source selection mechanism, etc.; the parameter configuration may take into account the role of the device in the network topology (master-slave clock), network delay estimation requirements, etc.; after the configuration is completed, the PTP initialization configuration data is generated, including the PTP setting value and network identification of each device.

[0146] Step S32: Execute the best master clock algorithm according to the PTP initialization configuration data and the dynamic network topology diagram to select the best master clock source in the network, thereby obtaining the master clock device identification data;

[0147] The embodiment of the present invention uses the PTP initialization data configured in step S31 and the dynamic network topology diagram to execute the best master clock algorithm. The goal of the algorithm is to select the best master clock source from the network to ensure the best clock synchronization effect in the entire network. According to the result of the best master clock algorithm, a master clock identifier is assigned to each device, and the master clock identifier data is used to determine the clock synchronization hierarchy and message exchange in subsequent steps.

[0148] Step S33: constructing a master-slave clock hierarchical structure based on the network topology according to the master clock device identification data and the dynamic network topology diagram, thereby obtaining a clock synchronization hierarchical diagram;

[0149] The embodiment of the present invention uses the master clock device identification data and the dynamic network topology diagram to construct a clock synchronization hierarchical structure and determine the position of each device in the clock synchronization chain, including the master-slave relationship and the synchronization path.

[0150] Step S34: exchanging PTP messages between the master and slave devices according to the clock synchronization hierarchy diagram, thereby obtaining original timestamp data;

[0151] According to the clock synchronization hierarchical structure, the embodiment of the present invention exchanges PTP messages between master and slave devices, and the messages may include timestamps, synchronization event triggers, calibration requests, etc. During the message exchange process, the original timestamp data is collected from each device, and these timestamps reflect the time difference between each device and the master clock source.

[0152] Step S35: Calculate the network delay from each device to the master clock source according to the original timestamp data, thereby obtaining network delay estimation data;

[0153] The embodiment of the present invention calculates the network delay from each device to the master clock source based on the collected original timestamp data, taking into account the message transmission time, device processing delay and path characteristics in the network topology. Using the network delay estimation data generated in step S35, the adaptive audio stream data is delayed compensated and the audio playback time is adjusted to ensure that all devices remain synchronized in a multi-user shared audio scenario.

[0154] Step S36: Delay compensation is performed on the adaptive audio stream data according to the network delay estimation data, thereby obtaining synchronous audio stream data.

[0155] The embodiment of the present invention combines the delay compensation processing in step S35 to generate final synchronous audio stream data, which can be directly used in subsequent audio transmission and multi-user sharing scenarios to ensure the synchronization and quality of audio playback.

[0156] The present invention initializes the IEEE 1588PTP protocol stack and configures PTP parameters, ensuring that all devices can use the same time source for synchronization, which helps to achieve accurate time synchronization; through precise PTP parameter configuration, the accuracy of time synchronization between devices can be significantly improved, and clock deviation can be reduced; PTP initialization is performed in a dynamic network topology, so that the system can adapt to changes in network topology and ensure the stability and reliability of time synchronization. By executing the best master clock algorithm, the best master clock source is selected to ensure the accuracy and stability of time synchronization in the network; selecting the best master clock source can improve the robustness of the system and reduce the time synchronization error caused by the instability of the clock source; selecting the master clock source according to the dynamic network topology enables the system to dynamically adapt to network changes and maintain the continuity of time synchronization. Construct a master-slave clock hierarchical structure, clarify the clock hierarchical relationship of each device in the network, and ensure the orderliness of time synchronization; by constructing a clock hierarchy diagram, optimize the time synchronization path, improve the synchronization efficiency, and reduce the delay of time synchronization; support a multi-level clock synchronization structure, adapt to complex network topologies, and improve the coverage and accuracy of time synchronization. By exchanging PTP messages, the original timestamp data of each device can be obtained in real time, providing basic data for subsequent delay calculation and time synchronization; through standardized PTP message exchange, the accuracy and consistency of timestamp data can be ensured, and synchronization errors can be reduced; frequent PTP message exchange can support high-frequency time synchronization requirements and improve the accuracy of time synchronization. According to the original timestamp data, the network delay from each device to the master clock source is accurately calculated to provide accurate delay data for subsequent time synchronization; through accurate delay calculation, the time synchronization error caused by network delay is reduced and the synchronization accuracy is improved; the delay estimation data can be dynamically adjusted to adapt to changes in network status and maintain the accuracy of time synchronization. Delay compensation is performed based on the network delay estimation data to ensure that the audio playback time on each device is consistent and synchronized playback is achieved; through accurate delay compensation, the delay difference in audio playback is eliminated to improve the user experience of audio sharing between multiple devices; the delay compensation is dynamically adjusted to adapt to changes in network delay to ensure the synchronization and stability of audio stream data.

[0157] Preferably, step S36 includes the following steps:

[0158] Step S361: Calculate the clock offset between each device and the master clock source according to the network delay estimation data and the original timestamp data, thereby obtaining clock offset value data;

[0159] The embodiment of the present invention uses the original timestamp data collected in step S34 and the network delay estimation data calculated in step S35 to calculate the clock offset between each device and the master clock source, and the clock offset reflects the time difference between the local clock of each device and the master clock source.

[0160] Step S362: adjusting the local clock of each device to align with the master clock source according to the clock offset value data, thereby obtaining device clock calibration data;

[0161] The embodiment of the present invention adjusts the local clock of each device according to the clock offset value data calculated in step S361 to synchronize it with the main clock source. The method of adjusting the local clock may include increasing or decreasing the frequency or phase of the local clock of the device, and recording the clock state and calibration parameters of each device after calibration. These data are used for subsequent audio transmission and delay compensation operations.

[0162] Step S363: measuring the end-to-end transmission delay from the audio source to each receiving device according to the device clock calibration data and the dynamic network topology diagram, thereby obtaining transmission delay matrix data;

[0163] The embodiment of the present invention uses device clock calibration data and a dynamic network topology diagram to measure the end-to-end transmission delay from an audio source to each receiving device, taking into account factors such as network topology, message transmission time, and device processing delay during the measurement process; and generates matrix data describing the transmission delay relationship between each device. Each element in the matrix represents the delay between the audio source and the receiving device, which can directly affect the synchronization and quality of audio playback.

[0164] Step S364: Calculate the audio playback delay compensation value required by each device according to the transmission delay matrix data, thereby obtaining delay compensation value data;

[0165] The embodiment of the present invention calculates the audio playback delay compensation value required by each device based on the transmission delay matrix data measured in step S363; the calculation of the delay compensation value takes into account the time difference of the audio stream during the transmission process to ensure that all devices remain synchronized when playing audio.

[0166] Step S365: resample and adjust the buffer of the adaptive audio stream data according to the delay compensation value data to obtain synchronous audio stream data.

[0167] The embodiment of the present invention resamples and adjusts the buffer of the adaptive audio stream data according to the delay compensation value data calculated in step S364; the resampling process adjusts the sampling rate or interpolation of the audio data according to the delay compensation value to adapt to the playback clock of different devices. The final synchronous audio stream data is generated by combining the resampled and buffered adjusted data; these data can be directly used in the audio playback system to ensure synchronization and high-quality audio experience in multi-user shared audio scenarios.

[0168] The present invention can accurately calibrate the device clock and reduce the time synchronization error by calculating the clock offset between each device and the main clock source; obtain accurate clock offset value, ensure the synchronization accuracy of the device clock, and reduce the delay difference during playback; calculate the clock offset in real time, adapt to changes in network conditions, and maintain continuous synchronization accuracy. Adjust the local clock of each device to eliminate the clock error between the device and the main clock source to ensure time synchronization; calibrate the device clock to ensure the time consistency of all devices in the system, improve the stability of the overall system, ensure the clock synchronization between multiple devices, and achieve high-precision collaborative work between multiple devices. Use the calibrated device clock to accurately measure the end-to-end transmission delay from the audio source to each receiving device and obtain accurate delay data; provide accurate delay estimation through detailed transmission delay measurement, reduce delay compensation error, provide a detailed transmission delay matrix, provide a basis for subsequent delay compensation, and ensure audio synchronous playback. According to the transmission delay of each device, calculate personalized playback delay compensation value to ensure audio playback synchronization on all devices; accurate delay compensation value calculation ensures the consistency of playback time of each device, eliminates the phenomenon of asynchronism, and provides seamless audio playback experience through delay compensation to improve the user's audio sharing experience. Through resampling and buffer adjustment, the audio streams on all devices are ensured to be played synchronously, realizing true multi-device audio sharing; through precise resampling and buffer adjustment, audio distortion is reduced, audio quality is maintained, audio streams are dynamically adjusted, and network conditions change, ensuring the stability and consistency of audio playback.

[0169] Preferably, step S4 comprises the following steps:

[0170] Step S41: designing an optimal transmission path based on the dynamic network topology diagram, establishing a virtual transmission network, and thus obtaining an audio data transmission network;

[0171] The embodiment of the present invention uses the dynamic network topology map obtained in step S14 to analyze the connection relationship, network status and bandwidth status between various devices; designs the optimal transmission path according to the network topology information to ensure that the audio data can be transmitted with the best efficiency and minimum delay during the transmission process. Based on the designed optimal transmission path, a virtual transmission network is established, that is, a data transmission channel between each device is defined; ensure that the virtual transmission network can support the parallel transmission of multiple audio streams, and take into account the priorities and needs of different users.

[0172] Step S42: segmenting and encoding the synchronous audio stream data into data packets, and assigning transmission priorities to the data packets according to the network status and user priorities in the dynamic network topology diagram, thereby obtaining a data packet queue with priorities;

[0173] The embodiment of the present invention performs data packet segmentation on the synchronized audio stream data, and divides the audio data into data packets of appropriate size; encodes each data packet to ensure the integrity and reliability of the data. According to the network status of the device and the priority of the user in the dynamic network topology diagram, an appropriate transmission priority is assigned to each data packet; high-priority data packets may receive faster response and processing in network transmission, ensuring that audio data is transmitted first in a high-quality network environment.

[0174] Step S43: selecting and configuring a suitable transmission protocol according to the network characteristics of the audio data transmission network, and initializing the distributed consensus protocol, thereby obtaining the initial state data of the consensus protocol;

[0175] The embodiment of the present invention selects a suitable transmission protocol, such as TCP, UDP or a custom protocol, according to the characteristics of the audio data transmission network (such as bandwidth, delay requirements, etc.); configures the parameters of the transmission protocol to ensure that the requirements of real-time and reliability of audio data can be met. If it is necessary to ensure the consistency and integrity of the data, a distributed consensus protocol can be initialized, such as a consensus algorithm based on blockchain or other distributed protocols; the initial state data of the consensus protocol includes the identity authentication information of the participating devices and the initial state of the shared data.

[0176] Step S44: broadcasting the data packets with priority to all participating devices in the network according to the priority of the data packet queue, and each receiving device sends a data packet reception confirmation information, thereby obtaining a reception confirmation state table;

[0177] The embodiment of the present invention broadcasts the data packets with priority allocation to all participating devices in the audio data transmission network according to the set transmission path and protocol, ensuring that the data packets can be quickly and effectively transmitted to the target device. After receiving the data packet, each receiving device sends a reception confirmation message to the sending device; after the sending device receives a sufficient number of reception confirmations, it can confirm that the data packet has been successfully transmitted in the network.

[0178] Step S45: Execute the consensus protocol according to the reception confirmation status table and the consensus protocol initial status data, and vote on the integrity and sequence of the data packets to verify whether a consensus is reached, thereby obtaining a consistency verification report;

[0179] The embodiment of the present invention executes a consensus algorithm based on the received confirmation status table and the initialized consensus protocol status data to ensure that all participating devices reach an agreement on the integrity and order of the data packet; each device verifies the correctness and order of the data packet by voting to ensure the consistency of the audio data. A consistency verification report is generated, including the voting results in the consensus process and detailed information on the verification data packet; the report reflects the consistency and integrity of the data packet during the transmission process.

[0180] Step S46: reorganizing and decoding the data packet queue according to the consistency verification report to obtain consistent audio stream data;

[0181] The embodiment of the present invention reassembles the received data packets according to the consistency verification report to ensure that the data packets are assembled into complete audio stream data in the correct order. The reassembled data packets are decoded to restore them to the original audio stream data, ensuring that the decoding process does not introduce additional distortion or information loss.

[0182] Step S47: performing information verification based on the hash fingerprint on the consistent audio stream data to obtain hash verification result data;

[0183] The embodiment of the present invention performs information verification based on hash fingerprint on the decoded consistent audio stream data, calculates the hash value of the audio data, compares it with the expected hash verification result data, and verifies the integrity of the data.

[0184] Step S48: Selectively retransmit trigger or forward error correction is performed on the consistent audio stream data according to the hash verification result data to achieve multi-user sharing of audio.

[0185] The embodiments of the present invention determine whether it is necessary to selectively retransmit certain data packets or apply a forward error correction algorithm based on the hash verification result data; these methods can help correct possible errors or lost data during data transmission to ensure that the end user receives high-quality audio stream data.

[0186] The present invention designs the optimal transmission path according to the dynamic network topology diagram, ensures the efficient transmission of data packets in the network, reduces transmission delay and packet loss rate; establishes a virtual transmission network, can flexibly adjust the transmission path, improve network resource utilization, and enhance overall transmission efficiency; the optimal path design and the use of virtual networks improve the reliability and stability of the system, and ensure the smooth transmission of audio data. The audio stream data is segmented and encoded into packets to facilitate management and control, ensuring the orderliness and integrity of data transmission; the transmission priority is allocated according to the network status and user priority to ensure that key data is transmitted first, optimize resource usage, and the priority packet queue can effectively reduce network congestion and improve the stability and quality of audio data transmission. Select and configure a suitable transmission protocol to ensure that audio data can be efficiently transmitted under different network conditions and improve the transmission success rate; initialize the distributed consensus protocol to provide a basis for subsequent data consistency verification and ensure the reliability of data transmission. The initial state data of the consensus protocol ensures that each device works together on the same basis to improve the coordination of the overall system. Broadcast data packets ensure that all participating devices can receive audio data, enhancing the coverage of audio sharing; receiving confirmation information provides real-time feedback on data packet transmission, facilitating monitoring and management, timely discovering and solving transmission problems, and receiving confirmation status table helps confirm the reception status of data packets to ensure the reliability and integrity of data transmission. Through consensus protocol execution and voting mechanism, the integrity and order of data packets are verified to ensure the consistency of data transmission. The consistency verification report provides the verification results of the transmitted data, improving the security and credibility of data transmission. Through the verification of the consensus mechanism, errors and losses in the data transmission process are reduced, and the accuracy of data transmission is improved. Data packets are reorganized according to the consistency verification report to ensure the correct order and integrity of audio data; the reorganized data packets are decoded to restore the original audio stream data to ensure the quality and consistency of audio data; through reorganization and decoding, consistent audio stream data is provided to enhance the user's audio sharing experience. Consistent audio stream data is verified through hash fingerprints to ensure that the data has not been tampered with during transmission and ensure data integrity. The hash verification result data provides data verification information to improve the security of data transmission. The verification method based on hash fingerprints is fast and can detect data integrity in real time, reducing verification time. Based on the hash verification results, data retransmission or forward error correction is performed selectively to ensure the correct transmission of audio data. Selective retransmission and forward error correction improve the reliability of data transmission, reduce data loss and errors, and optimize the use of network resources through selective retransmission, reduce unnecessary retransmissions, and improve transmission efficiency.

[0187] The present invention also provides a multi-user audio sharing system based on a le-audio Bluetooth headset, which is used to execute the multi-user audio sharing method based on a le-audio Bluetooth headset. The multi-user audio sharing system based on a le-audio Bluetooth headset comprises:

[0188] The network discovery and channel allocation module is used to obtain the connection status and topology of LE-Audio devices in real time through the network discovery function of the Bluetooth protocol stack, thereby generating a dynamic network topology map of Bluetooth; based on the dynamic network topology map, the available bandwidth is divided into multiple small-bandwidth channels, and an independent small-bandwidth channel is allocated to each user for audio transmission, forming multiple parallel audio stream channels;

[0189] The audio adaptation module is used to model the multi-user audio sharing scenario according to the audio stream channel, and obtain a channel response matrix describing the multi-user audio transmission characteristics; according to the channel response matrix, the audio transmitted in each small bandwidth channel is adaptively and dynamically adjusted to generate adaptive audio stream data;

[0190] The clock synchronization and delay compensation module is used to synchronize the clocks between the devices participating in the sharing according to the dynamic network topology diagram using the IEEE 1588PTP protocol, measure the transmission delay from each device to the audio source, and perform delay compensation on the adaptive audio stream data, thereby obtaining the synchronous audio stream data;

[0191] The consistency transmission and error correction module is used to build an audio data transmission network based on a dynamic network topology diagram, and to perform consistency confirmation transmission of synchronous audio stream data through a consensus mechanism, thereby obtaining consistent audio stream data; to perform information verification based on hash fingerprints on the consistent audio stream data to obtain hash verification result data, and to perform selective retransmission triggering or forward error correction on the consistent audio stream data based on the hash verification result data to achieve multi-user sharing of audio.

[0192] The present invention obtains the device connection status and topology in real time, ensures the timely update and adjustment of network information, improves the flexibility and adaptability of the system, divides the available bandwidth into multiple small bandwidth channels, realizes the refined management and efficient use of bandwidth, ensures the stability and high quality of audio transmission, and allocates independent small bandwidth channels to each user to form multiple parallel audio stream channels, effectively avoids mutual interference between channels, and improves the clarity and integrity of audio transmission. By modeling the multi-user audio sharing scenario, the audio transmission characteristics are accurately described, and a scientific basis is provided for channel adjustment and optimization; the audio transmission is dynamically adjusted according to the channel response matrix, and the network changes can be responded to in real time, ensuring the stability and efficiency of audio data transmission. The generation of adaptive audio stream data improves the audio transmission quality, reduces distortion and interference, and improves the user experience. The IEEE 1588PTP protocol is used to achieve high-precision clock synchronization between devices, ensuring that the audio data is played at the same time on each device to avoid the problem of asynchronous; the transmission delay is measured and calculated, and delay compensation is performed to ensure that the transmission and playback delay of audio data is minimized, and the audio synchronization and consistency are improved; the combination of clock synchronization and delay compensation improves the coordination and consistency of audio playback, ensuring the high-quality experience of multi-user audio sharing. Consistency confirmation is performed through a consensus mechanism to ensure that the audio stream data remains consistent during transmission, improve the reliability of data transmission, and verify information based on hash fingerprints to ensure the integrity and security of data during transmission, prevent data tampering, and perform selective retransmission or forward error correction based on the hash verification results to improve the correctness and success rate of data transmission and ensure the high quality and continuity of audio sharing. These four modules together build an efficient and reliable multi-user audio sharing system. The network discovery and channel allocation module ensures the optimization of device connection and bandwidth utilization; the audio adaptation module improves the stability and quality of audio transmission through dynamic adjustment; the clock synchronization and delay compensation module ensures the synchronization and consistency of audio; the consistency transmission and error correction module improves the reliability and security of transmission through data verification and error correction. Through the collaborative work of these modules, the system can provide high-quality audio sharing services in a complex multi-user environment.

[0193] Therefore, the embodiments should be regarded as illustrative and non-restrictive from all points, and the scope of the present invention is limited by the appended claims rather than the above description, and it is therefore intended that all changes falling within the meaning and range of equivalent elements of the application documents are included in the present invention.

[0194] The above description is only a specific embodiment of the present invention, so that those skilled in the art can understand or implement the present invention. Various modifications to these embodiments will be apparent to those skilled in the art, and the general principles defined herein may be implemented in other embodiments without departing from the spirit or scope of the present invention. Therefore, the present invention will not be limited to the embodiments shown herein, but should conform to the widest scope consistent with the principles and novel features invented herein.

Claims

1. A multi-user audio sharing method based on le-audio Bluetooth headset, characterized in that: The following steps are involved: Step S1: Obtain the connection status and topology of the LE-Audio device in real time through the network discovery function of the Bluetooth protocol stack, thereby generating a dynamic network topology map of Bluetooth; Based on the dynamic network topology, the available bandwidth is divided into multiple small bandwidth channels, and an independent small bandwidth channel is allocated to each user for audio transmission, forming multiple parallel audio stream channels; Step S2: Modeling a multi-user audio sharing scenario according to the audio stream channel to obtain a channel response matrix describing the multi-user audio transmission characteristics; adaptively and dynamically adjusting the audio transmitted in each small bandwidth channel according to the channel response matrix to generate adaptive audio stream data; Step S3: using the IEEE 1588PTP protocol to synchronize the clocks between the devices participating in the sharing according to the dynamic network topology, measuring the transmission delay from each device to the audio source, and performing delay compensation on the adaptive audio stream data, thereby obtaining synchronous audio stream data; Step S4: construct an audio data transmission network according to the dynamic network topology diagram, and perform consistency confirmation transmission on the synchronous audio stream data through a consensus mechanism to obtain consistent audio stream data; perform information verification based on hash fingerprint on the consistent audio stream data to obtain hash verification result data, and perform selective retransmission triggering or forward error correction on the consistent audio stream data according to the hash verification result data to achieve multi-user sharing of audio.

2. The multi-user audio sharing method based on le-audio Bluetooth headset according to claim 1 is characterized in that: Step S1 includes the following steps: Step S11: Scan the surrounding LE-Audio devices using the scanning function of the Bluetooth protocol stack, and collect the scanned device information to obtain the original device scanning data; Step S12: authenticating the scanned device according to the original device scan data and establishing a secure pairing connection, thereby obtaining a list of verified devices; Step S13: obtaining the connection relationship between devices in the verified device list through the network discovery function of the Bluetooth protocol stack, determining the master and slave device roles, identifying the communication path between devices, and obtaining initial topology structure data; Step S14: establishing a graphical representation of device nodes and connection edges according to the initial topology data, and adding device attribute information, thereby obtaining a dynamic network topology diagram of Bluetooth; Step S15: performing link quality analysis on the dynamic network topology diagram and estimating the total available bandwidth, thereby obtaining bandwidth resource evaluation data; Step S16: performing small bandwidth channel division according to the bandwidth resource evaluation data and the number of user equipments, and assigning a unique identifier to each channel, thereby obtaining channel division data; Step S17: allocating an independent small-bandwidth channel for audio transmission to each user according to the channel division data, thereby forming multiple parallel audio stream channels.

3. The multi-user audio sharing method based on le-audio Bluetooth headset according to claim 2 is characterized in that: Step S2 includes the following steps: Step S21: Analyze the bandwidth, signal-to-noise ratio, and bit error rate of each audio stream channel to obtain channel characteristic parameter data; Step S22: constructing a multi-user interference model based on user space distribution according to the channel characteristic parameter data, thereby obtaining scenario model parameter data; Step S23: Calculate the interference coefficient of each user to other users according to the scenario model parameter data, thereby generating a time-varying channel response matrix; Step S24: using a singular value decomposition algorithm to improve the sparsity and computational efficiency of the time-varying channel response matrix, thereby obtaining a channel response matrix; Step S25: designing an adaptive equalizer for compensating channel distortion and suppressing interference according to the channel response matrix, thereby obtaining equalizer coefficient data; Step S26: Adaptively and dynamically adjust the audio transmitted in each small bandwidth channel according to the channel response matrix and the equalizer coefficient data to generate adaptive audio stream data.

4. The multi-user audio sharing method based on le-audio Bluetooth headset according to claim 3 is characterized in that: Step S26 includes the following steps: Step S261: dynamically adjusting the transmission power of each user according to the channel response matrix and the equalizer coefficient data, thereby obtaining power allocation scheme data; Step S262: selecting an audio coding scheme for each user according to the channel quality in the channel response matrix and the available power in the power allocation scheme data, thereby obtaining a user coding scheme mapping table; Step S263: performing noise reduction compression on the audio transmitted in each small bandwidth channel according to the user coding scheme mapping table, thereby obtaining noise-reduced audio data; Step S264: Adaptively adjust the noise reduction audio data according to the channel response matrix, the equalizer coefficient data and the power allocation scheme data, so as to obtain adaptive audio stream data, wherein the adaptive adjustment includes power control, spectrum shaping and time domain equalization.

5. The multi-user audio sharing method based on le-audio Bluetooth headset according to claim 4 is characterized in that: Step S3 includes the following steps: Step S31: Initializing the IEEE 1588PTP protocol stack on the device in the dynamic network topology diagram and configuring PTP parameters to obtain PTP initialization configuration data; Step S32: Execute the best master clock algorithm according to the PTP initialization configuration data and the dynamic network topology diagram to select the best master clock source in the network, thereby obtaining the master clock device identification data; Step S33: constructing a master-slave clock hierarchical structure based on the network topology according to the master clock device identification data and the dynamic network topology diagram, thereby obtaining a clock synchronization hierarchical diagram; Step S34: exchanging PTP messages between the master and slave devices according to the clock synchronization hierarchy diagram, thereby obtaining original timestamp data; Step S35: Calculate the network delay from each device to the master clock source according to the original timestamp data, thereby obtaining network delay estimation data; Step S36: Delay compensation is performed on the adaptive audio stream data according to the network delay estimation data, thereby obtaining synchronous audio stream data.

6. The multi-user audio sharing method based on le-audio Bluetooth headset according to claim 5 is characterized in that: Step S36 includes the following steps: Step S361: Calculate the clock offset between each device and the master clock source according to the network delay estimation data and the original timestamp data, thereby obtaining clock offset value data; Step S362: adjusting the local clock of each device to align with the master clock source according to the clock offset value data, thereby obtaining device clock calibration data; Step S363: measuring the end-to-end transmission delay from the audio source to each receiving device according to the device clock calibration data and the dynamic network topology diagram, thereby obtaining transmission delay matrix data; Step S364: Calculate the audio playback delay compensation value required by each device according to the transmission delay matrix data, thereby obtaining delay compensation value data; Step S365: resample and adjust the buffer of the adaptive audio stream data according to the delay compensation value data to obtain synchronous audio stream data.

7. The multi-user audio sharing method based on le-audio Bluetooth headset according to claim 6 is characterized in that: Step S4 includes the following steps: Step S41: designing an optimal transmission path based on the dynamic network topology diagram, establishing a virtual transmission network, and thus obtaining an audio data transmission network; Step S42: segmenting and encoding the synchronous audio stream data into data packets, and assigning transmission priorities to the data packets according to the network status and user priorities in the dynamic network topology diagram, thereby obtaining a data packet queue with priorities; Step S43: selecting and configuring a suitable transmission protocol according to the network characteristics of the audio data transmission network, and initializing the distributed consensus protocol, thereby obtaining the initial state data of the consensus protocol; Step S44: broadcasting the data packets with priority to all participating devices in the network according to the priority of the data packet queue, and each receiving device sends a data packet reception confirmation information, thereby obtaining a reception confirmation state table; Step S45: Execute the consensus protocol according to the reception confirmation status table and the consensus protocol initial status data, and vote on the integrity and sequence of the data packets to verify whether a consensus is reached, thereby obtaining a consistency verification report; Step S46: reorganizing and decoding the data packet queue according to the consistency verification report to obtain consistent audio stream data; Step S47: performing information verification based on the hash fingerprint on the consistent audio stream data to obtain hash verification result data; Step S48: Selectively retransmit trigger or forward error correction is performed on the consistent audio stream data according to the hash verification result data to achieve multi-user sharing of audio.

8. A multi-user audio sharing system based on le-audio Bluetooth headset, characterized in that: Used to execute the multi-user audio sharing method based on le-audio Bluetooth headset as claimed in claim 1, the multi-user audio sharing system based on le-audio Bluetooth headset comprises: The network discovery and channel allocation module is used to obtain the connection status and topology of LE-Audio devices in real time through the network discovery function of the Bluetooth protocol stack, thereby generating a dynamic network topology map of Bluetooth; based on the dynamic network topology map, the available bandwidth is divided into multiple small-bandwidth channels, and an independent small-bandwidth channel is allocated to each user for audio transmission, forming multiple parallel audio stream channels; The audio adaptation module is used to model the multi-user audio sharing scenario according to the audio stream channel, and obtain a channel response matrix describing the multi-user audio transmission characteristics; according to the channel response matrix, the audio transmitted in each small bandwidth channel is adaptively and dynamically adjusted to generate adaptive audio stream data; The clock synchronization and delay compensation module is used to synchronize the clocks between the devices participating in the sharing according to the dynamic network topology diagram using the IEEE 1588PTP protocol, measure the transmission delay from each device to the audio source, and perform delay compensation on the adaptive audio stream data, thereby obtaining the synchronous audio stream data; The consistency transmission and error correction module is used to build an audio data transmission network based on a dynamic network topology diagram, and to perform consistency confirmation transmission of synchronous audio stream data through a consensus mechanism, thereby obtaining consistent audio stream data; to perform information verification based on hash fingerprints on the consistent audio stream data to obtain hash verification result data, and to perform selective retransmission triggering or forward error correction on the consistent audio stream data based on the hash verification result data to achieve multi-user sharing of audio.

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