A wearable bone conduction hearing assistance system and device

Abstract

The application discloses a wearable bone conduction hearing assistance system and equipment, and relates to the technical field of bone conduction data processing. The wearable bone conduction hearing assistance system comprises an audio processing module, a data transmission module and an audio receiving module. The audio processing module performs standardized processing on continuously read audio files based on data condition parameters. The data transmission module transmits the data blocks after standardized processing to independent queues of each device, dynamically adjusts the queues based on the transportation state parameters of each device in the data block transmission process, and then transmits and controls the reliability of the UDP frames obtained by encapsulating the data blocks in the independent queues. After scanning the SSID information preset by the device, the audio receiving module performs network monitoring processing in combination with the SSID scanning result and the connection state parameters of the device, and then performs adaptive playing control on the UDP frames received by the device, thereby supporting the synchronous access of a large number of hearing-impaired users.

Description

Technical Field

[0001] This invention relates to the field of bone conduction data processing technology, and in particular to a wearable bone conduction hearing aid system and device. Background Technology

[0002] Bluetooth technology is a short-range wireless communication standard widely used for connecting audio devices. For music transmission, it uses the A2DP protocol to wirelessly transmit digital audio signals from audio source devices such as mobile phones and computers to playback devices such as headphones and speakers. Existing hearing aids each have their own characteristics but also limitations. Bone conduction technology directly stimulates the auditory nerve through skull vibrations, maintaining ambient sound perception while avoiding the physical damage risks of traditional headphones. However, its current driver units lack sufficient power, making it difficult to provide adequate vibration intensity in high-noise environments such as concerts. While cochlear implants can effectively reconstruct hearing function, signal processing is easily interfered with in noisy environments, affecting usability. Bluetooth technology, while enabling convenient wireless audio transmission, has limitations in the number of connected devices and encoding / decoding losses, restricting audio quality in group applications. To truly meet the needs of hearing-impaired individuals participating in large-scale events, there is an urgent need to develop higher-power drivers and optimize the adaptation scheme of bone conduction to the optimal vibration transmission position in the human body, thereby achieving both powerful and clearly discernible vibrational audio transmission in complex acoustic environments.

[0003] For example, patent application CN109933202B discloses a bone conduction-based intelligent input method and system, which includes: receiving bone conduction vibration signals from keys and extracting features to obtain the feature-extracted vibration signals; inputting the time sequence of the feature-extracted vibration signals into a trained neural network classification model to identify the key type corresponding to the bone conduction vibration signals; and determining the text information input by the user based on the identified key type.

[0004] However, in the process of implementing the inventive technical solution in the embodiments of this application, it was found that the above-mentioned technology has at least the following technical problems:

[0005] In existing technologies, the current Bluetooth audio transmission protocol has an inherent upper limit on the number of master and slave devices that can be connected. This technical limitation may lead to insufficient device connectivity in multi-user application scenarios. When the number of hearing-impaired users on site exceeds the maximum number of connections supported by the Bluetooth protocol, some users will be unable to access the system, resulting in insufficient coverage of hearing assistance services. Summary of the Invention

[0006] This application provides a wearable bone conduction hearing assistance system and device, which solves the problem that the existing Bluetooth audio transmission protocol has an inherent upper limit on the number of master and slave devices that can be connected. This limitation may lead to insufficient device connectivity in multi-user application scenarios. When the number of hearing-impaired users on site exceeds the maximum number of connections supported by the Bluetooth protocol, some users will be unable to access the system, resulting in insufficient coverage of hearing assistance services. This system supports simultaneous access for a large number of hearing-impaired users.

[0007] The wearable bone conduction hearing aid system provided in this application includes: an audio processing module, a data transmission module, and an audio receiving module. The audio processing module is used to perform file standardization evaluation on continuously read audio files based on data condition parameters, and to standardize the audio files based on the file standardization evaluation results. Standardization means that the application layer converts the continuously read audio files into a format conforming to preset data condition parameters, and then converts them into data blocks of a specified format. The data transmission module is used to transmit the standardized data blocks into independent queues of each device, evaluate the queue status based on the transportation status parameters of each device during data block transmission, and then... The system performs dynamic queue adjustment, and then encapsulates the data blocks in the independent queues to obtain UDP frames for transmission and reliability control. Dynamic queue adjustment means dynamically adjusting the data production rate during transportation based on transportation status parameters to reduce the risk of data block overflow in the independent queues. The audio receiving module is used to scan the device's preset SSID information, combine the SSID scanning results with the device's connection status parameters to perform network status assessment, and perform network monitoring processing based on the network status assessment results. Then, it performs adaptive playback control on the UDP frames received by the device after network monitoring processing. Network monitoring processing means performing real-time analysis and processing based on the connection status parameters of the device's receiving end to improve the reliability of network transmission and connection stability.

[0008] This application also provides a wearable bone conduction hearing aid device, comprising: a processor; a memory, wherein the memory stores computer-readable instructions, and when the computer-readable instructions are executed by the processor, the wearable bone conduction hearing aid system is realized.

[0009] One or more technical solutions provided in the embodiments of this application have at least the following technical effects or advantages:

[0010] 1. The acquired audio files are standardized according to preset specifications, converting them into audio data of the target format. These data are then divided into fixed-size blocks and allocated to independent transmission queues based on different target devices. During transmission, the system dynamically adjusts the data generation rate of each queue based on its transport status, effectively avoiding overflow risks and ensuring transmission stability. The processed data blocks are encapsulated using the UDP protocol and sent to the receiving end. Before receiving the data, the receiving end assesses network reliability through SSID scanning and connection status analysis. Upon receiving the data from the sender, it sequentially performs operations such as transmission integrity verification, byte alignment detection, and silence packet filtering, ultimately outputting high-quality, real-time audio data. While ensuring smooth audio transmission, it also has the capability to support simultaneous access for a large number of hearing-impaired users, significantly improving the inclusivity and user experience of the audio service.

[0011] 2. During audio data transmission, the system continuously monitors the operating status of each device's transmission channel and dynamically adjusts the data generation rate of the corresponding queue based on the operating status assessment results. This ensures that each independent queue is always kept at its optimal load, fully utilizing available bandwidth resources while significantly reducing the probability of data loss. Optimized data blocks are encapsulated into UDP data frames with sequence numbers and checksums and sent to the target device. After receiving an acknowledgment signal from the receiving end, the system can continuously send subsequent data frames, maximizing bandwidth utilization and ensuring continuous, smooth transmission and real-time response. If a timeout occurs without a response, the system automatically moves the data frame to the retransmission queue, effectively compensating for potential data loss through subsequent retransmissions, thereby comprehensively improving the integrity and reliability of audio transmission.

[0012] 3. At the receiving end, the system first comprehensively assesses the current network's transmission reliability using SSID scan results and real-time connection status parameters of the devices, and continuously monitors the network links dynamically based on these status parameters. Only stable networks that pass the verification are used for subsequent data transmission, thus ensuring the effectiveness of the underlying links. For UDP data frames received on these network links, the system sequentially performs transmission integrity verification, fine-grained byte alignment assessment, and silence packet filtering. This effectively eliminates transmission errors, format misalignments, and useless silence data, significantly improving the clarity, coherence, and real-time performance of audio output, ensuring stable reproduction of high-quality audio. Attached Figure Description

[0013] Figure 1 This is a schematic diagram of the structure of the wearable bone conduction hearing aid system provided in the embodiments of this application;

[0014] Figure 2 A flowchart illustrating the dynamic queue adjustment process of a wearable bone conduction hearing aid system provided in this application embodiment. Detailed Implementation

[0015] This application provides a wearable bone conduction hearing assistance system and device, which solves the problem that the existing Bluetooth audio transmission protocol has an inherent limit on the number of master and slave devices that can be connected. This limitation may lead to insufficient device connectivity in multi-user application scenarios. When the number of hearing-impaired users on site exceeds the maximum number of connections supported by the Bluetooth protocol, some users will be unable to access the system, resulting in insufficient coverage of hearing assistance services. The overall approach is as follows:

[0016] First, the original audio file is standardized according to the preset audio transmission specifications, converting it into audio data that meets the target format requirements. Then, the processed audio data is divided into fixed-size data blocks, which are allocated to their respective independent transmission queues based on the target device. During transmission, the system monitors network status parameters in real time to ensure that the amount of data in the queue remains within a reasonable range, effectively preventing data overflow. The queue-optimized data blocks are encapsulated using the UDP (User Datagram Protocol) protocol and sent to the receiving end. Before receiving data, the receiving end first assesses the reliability of the current network through SSID (the name of the default Wi-Fi network the device is connected to) scanning and connection status analysis. Only network links that meet the quality requirements are used for data transmission. For verified UDP data frames, the receiving end sequentially performs transmission integrity verification, byte alignment detection, and silence packet filtering to ensure that the final output audio data has high quality and good real-time performance, while also having the capability for large-scale simultaneous access by hearing-impaired users.

[0017] To better understand the above technical solutions, the following will provide a detailed explanation of the technical solutions in conjunction with the accompanying drawings and specific implementation methods.

[0018] like Figure 1The diagram shows a wearable bone conduction hearing aid system provided in this embodiment of the application. The wearable bone conduction hearing aid system includes an audio processing module, a data transmission module, and an audio receiving module. The audio processing module is used to perform file standardization evaluation on continuously read audio files based on data condition parameters, and to standardize the audio files based on the file standardization evaluation results. Standardization means that the application layer converts the continuously read audio files into a format that conforms to preset data condition parameters, and then converts them into data blocks of a specified format. This system has an intelligent audio input detection function, which can automatically identify and prioritize capturing sound sources from virtual audio devices such as BlackHole, and also supports manual switching to a physical microphone to meet different audio input needs. The data transmission module is used to transmit the standardized data through the main thread. The subsequent data block transmission into each device's independent queue performs queue status evaluation based on the transportation status parameters of each device during data block transmission. Dynamic queue adjustments are then made based on the evaluation results. Furthermore, the UDP frames encapsulated from data blocks in the independent queues undergo transmission and reliability control. Dynamic queue adjustment means dynamically adjusting the data production rate of the independent queues according to the transportation status parameters to reduce the risk of data block overflow. The audio receiving module scans the device's preset SSID information, combines the SSID scan results with the device's connection status parameters to perform network status evaluation, and performs network monitoring processing based on the evaluation results. Adaptive playback control is then applied to the UDP frames received by the device after network monitoring processing. Network monitoring processing means real-time analysis and processing based on the device's receiving connection status parameters to improve network transmission reliability and connection stability. SSID information refers to the specific WiFi network name that the device pre-sets to connect to.

[0019] In this embodiment, the device is an ESP32-S3 device. The main thread is the first thread automatically created by the operating system when the program starts, responsible for global task scheduling and resource management. Device threads are created separately for each ESP32-S3 device, dedicated to data transmission with the corresponding device. The device queue is a buffer channel between the main thread and device threads. Each ESP32-S3 device has its own independent UDP thread, responsible for retrieving data from the corresponding queue and sending it via UDP unicast. The queue depth is 4 to avoid excessively long queues. This invention addresses the performance limitations of traditional hearing aids in high-noise environments such as concerts. This system innovatively employs bone conduction technology, directly transmitting audio information through mechanical vibration, effectively avoiding interference from environmental noise. The vibration actuator uses the Apple 7 enhanced Taptic motor, whose high response speed and acceleration characteristics ensure users receive real-time and accurate vibration feedback. The Taptic motor and custom-designed housing model are based on 3D printing technology and adopt a split structure, consisting of a main body and a cover. The model is optimized using Onshape software to ensure structural strength while taking into account ease of maintenance, and to minimize volume to improve wearing comfort. Its split design achieves efficient vibration transmission through a housing structure that fits the motor securely, while avoiding discomfort that may be caused by direct contact between X-axis linear vibration (horizontal movement) and the human body.

[0020] Regarding the positioning of the vibration units, experiments showed that the sternum and clavicle regions are sensitive to vibration but also highly resistant to it; therefore, the system fixes the vibration units in these two key locations. To accommodate different user body types, the system uses an adjustable vest as the wearer: the sternum unit is integrated into the front of the vest and its position is adjusted via the shoulder straps; the clavicle unit is finely adjusted via a slider on the shoulder straps secured by nylon strap clips to ensure a snug fit. The mainboard, equipped with the ESP32-S3 chip, is placed in a mesh pocket in the front of the chest. The power supply is connected to the clavicle vibration unit via a long cable, which is protected by both heat-shrink tubing and nylon tubing to prevent cable breakage. The vest's built-in mesh pocket also reduces space usage and further enhances wearing comfort.

[0021] At the circuit design level, audio signal lines are soldered to the positive and negative terminals of the six pads on the Taptic motor. After soldering, hot melt adhesive is used to reinforce the pads to prevent them from detaching. The audio signal drives the vibration motor via a MAX98357 amplifier, ensuring precise matching between the vibration intensity and the audio signal. The entire circuit balances stability and maintainability, ensuring reliable operation even in complex usage scenarios.

[0022] As a core functional support, this system proposes a wearable bone conduction hearing assistance system, characterized by comprising three main modules: an audio processing module, a data transmission module, and an audio receiving module. The audio processing module is responsible for standardizing the continuously read audio files based on preset parameters and converting them into a compliant data block format through standardization. The data transmission module transmits the standardized data blocks into independent queues for each device, dynamically adjusting the queues based on transport status parameters to reduce the risk of data overflow, and achieving reliable transmission via UDP frames. The audio receiving module scans the device's preset SSID information, assesses the network status based on connection status parameters, implements network monitoring and processing, and performs adaptive playback control on the received UDP frames, thereby ensuring that the system can provide stable and high-quality hearing assistance services in different environments.

[0023] Of the six pads on the Taptic motor, audio signal lines need to be soldered to the positive and negative terminals. To prevent the pads from detaching, hot melt adhesive is used for reinforcement after soldering. The audio signal drives the vibration motor via a MAX98357 amplifier, ensuring precise matching between vibration intensity and the audio signal. The entire circuit design balances stability and maintainability, ensuring reliable operation even in complex usage scenarios.

[0024] Furthermore, the data condition parameters include sampling points, bit depth, sampling rate, and channels. The sampling points are 256, the bit depth is 16-bit, the sampling rate is 8kHz, and the channels are mono. The number of bytes in a data block is obtained based on the sampling points and bit depth of the audio file. This invention ensures that each sampling point is stored in 16-bit format, i.e., 2 bytes, so the number of bytes in a data block is the product of the sampling points and the bit depth, i.e., 512 bytes. The audio duration of a data block is obtained based on the sampling points and sampling rate of the audio file. This invention ensures that each data block read strictly adheres to an 8kHz sampling rate, so the audio duration of a data block is the ratio of the sampling points to the sampling rate, i.e., 32 milliseconds. Based on the data condition parameters, a file standardization evaluation is performed on the continuously read audio file. The steps include: Step 1, determining whether the number of bytes in each data block of the audio file meets the data block byte count requirement; if yes, proceed directly to Step 2; otherwise, mark the file as a bit depth abnormal file. Step 2, determining whether the audio duration of each data block of the audio file meets the data block audio duration requirement; if yes, proceed directly to Step 3; otherwise, mark the audio file as a sampling rate abnormal file. Step 3, determining whether the number of channels in the audio file meets the preset number of channels; if yes, proceed directly to Step 4; otherwise, mark the audio file as a channel abnormal file. Step 4, determining whether the total number of samples in the audio file is an integer multiple of the sampling points; if yes, mark the audio file as a normal file; otherwise, mark the audio file as a data alignment abnormal file.

[0025] Specifically, the steps for standardizing audio files based on file standardization evaluation results include: if the audio file is a bit depth aberration file, then the sampling points in the audio file whose numerical range does not meet the preset bit depth numerical range are converted to the preset bit depth numerical range. For example, if the original bit depth of 8-bit is lower than the preset bit depth of 16-bit, then the unsigned value of the 8-bit is subtracted by 128 to align the silence point to 0, and then bit depth expansion is performed to map the 8-bit step size to the 16-bit step size, ensuring dynamic range expansion; if the original bit depth is 2... If the 4-bit value exceeds the preset bit depth of 16-bit, the 24-bit value is arithmetically right-shifted by 8 bits, retaining the sign bit. If the check result exceeds the 16-bit range due to the original data being too large, it is forcibly truncated to the maximum or minimum value that 16-bit can represent, thus ensuring that no errors occur due to overflow after data conversion. If the original is a 32-bit floating-point number, it needs to be converted to a 16-bit integer. In this case, the input 32-bit floating-point number is restricted to the standard numerical range specified by digital signal processing to prevent exceeding the range. In the case of an audio file with an abnormal sampling rate, a scaling factor is calculated using a linear ratio based on the numerical range of the target integer type and the actual limit range of the floating-point number. The floating-point number is then multiplied by this factor and rounded to the nearest integer. Floating-point numbers exceeding the range are pre-trimmed to avoid overflow. If the audio file has an abnormal sampling rate, a resampling prompt is issued. If the audio file has an abnormal channel, the phase difference between the left and right channels is checked. If it is consistent, a single channel is extracted directly; otherwise, the average of the corresponding sample points of the left and right channels is taken. If the audio file has an abnormal data alignment, waveform analysis is used to determine if there are any missing data points (sudden zeroing or irregular jumps). If so, the data is repaired using linear interpolation; otherwise, it indicates that the audio file does not have missing data but is misaligned. Zeros are added to the end of the audio file until the total number of samples in the audio file is an integer multiple of the sample points. If the audio file is normal, the volume gain of the input audio signal is amplified to twice its normal value. The mono audio data is copied to the left and right channels to generate stereo-compatible audio data, and then the processed stereo audio data is divided into 256 data blocks.

[0026] In this embodiment, the present invention intelligently classifies files and automatically triggers optimal repair strategies through precise audio parameter diagnosis: it not only performs adaptive numerical conversion for bit depth anomalies, intelligently selects extraction or mixing strategies based on phase analysis for channel anomalies, and precisely repairs data alignment anomalies by distinguishing them based on waveform characteristics and using interpolation or zero padding, but also automatically performs volume normalization and stereo compatibility processing on normal files after ensuring basic quality, and segments data blocks for subsequent processing. The present invention improves the automation level, repair accuracy, and efficiency of audio processing, providing a reliable and complete front-end solution for subsequent audio data analysis and large-scale processing.

[0027] Furthermore, the transportation status parameters include queue blocking duration, queue occupancy rate, and enqueue failure rate. The steps for evaluating the queue status based on the transportation status parameters of each device during data block transmission include: obtaining preset transportation status parameter reference data and transportation status parameter contribution data. The transportation status parameter reference data includes critical queue blocking duration, critical queue occupancy rate, and critical enqueue failure rate. The transportation status parameter contribution data includes the contribution of queue blocking duration, the contribution of queue occupancy rate, and the contribution of enqueue failure rate. The transportation status parameter reference data is used to perform a proportion convergence calculation with the corresponding transportation status parameters of each device. Then, the corresponding proportion convergence calculation results are weighted and coupled using the transportation status parameter contribution data to obtain the transportation status evaluation index of each device. The transportation status evaluation index is a comprehensive indicator that quantitatively measures the transportation status of the device based on the transportation status parameters.

[0028] The transportation status assessment index for each piece of equipment is obtained in the following ways:

[0029] ;

[0030] In the formula, This represents the transportation status assessment index of the e-th device. This indicates the contribution of queue blocking time. Indicates the contribution of queue occupancy rate. Indicates the contribution of the team entry failure rate. This represents the queue blocking time of the e-th device, which is the total waiting time for data blocks to be enqueued due to the queue being full. A high-precision timer can be started before the data block is enqueued, and the timer stops after successful enqueueing. The difference between the successful enqueueing time and the enqueueing attempt time can be accumulated. Indicates the blocking duration of the critical queue. This represents the queue occupancy rate of the e-th device, and is the ratio of the number of data blocks in the current queue to the maximum capacity of the queue. Indicates the critical queue occupancy rate. This represents the enqueue failure rate for the e-th device, which is the ratio of the number of data blocks dropped due to a full queue to the total number of enqueue attempts per unit time. This represents the critical failure rate for enqueuing, where e is the device number, e = 1, 2, 3, ..., E, and E is the total number of devices.

[0031] like Figure 2The diagram shown is a flowchart of the dynamic queue adjustment process for a wearable bone conduction hearing aid system provided in an embodiment of this application. In the diagram, the index is the transportation status assessment index, the first threshold is the first threshold for transportation status assessment, and the second threshold is the second threshold for transportation status assessment. The steps for transmission and reliability control of UDP frames encapsulated from data blocks in independent queues include: comparing the transport status evaluation index of each device with preset transport status evaluation first threshold and transport status evaluation second threshold respectively; if the transport status evaluation index of any device is lower than the transport status evaluation threshold, the data production rate of that device is adjusted to a preset minimum production rate; if the transport status evaluation index of that device remains lower than the transport status evaluation first threshold for a period exceeding a preset time window after the data production rate of that device is adjusted to the preset minimum production rate, a backup queue is opened until the transport status evaluation index of that device is no lower than the transport status evaluation first threshold; if the transport status evaluation index of any device is not lower than the transport status evaluation first threshold but is lower than the transport status evaluation second threshold, the data production rate is dynamically adjusted according to the transport status evaluation index of that device based on the data production rate mapping table until the transport status evaluation index of that device is no lower than the transport status evaluation second threshold; if the transport status evaluation index of any device is not lower than the transport status evaluation second threshold, no additional processing is performed.

[0032] In this embodiment, the data production rate is the amount of data blocks the system prepares to put into the transmission queue per unit time. The production rate mapping table is a key decision table connecting the queue status assessment system and the data source control module, enabling dynamic adjustment of the data production rate. The production rate mapping table supports one-to-one or many-to-one rate matching modes. The data production rate value is strictly constrained within the preset minimum and maximum production rate range, and piecewise linear interpolation is used to achieve smooth adjustment in the range below the second threshold of the transportation status assessment. The backup queue is a temporary storage space independent of the main queue. When the main queue of any device cannot process data in time due to device performance bottlenecks, the backup queue can temporarily take over the overflow data. Transmission and reliability control are performed on the UDP frames encapsulated from data blocks in the independent queue. By monitoring the queue load in real time, the system bottleneck can be accurately identified and the data production rate can be automatically adjusted. This avoids resource waste or overload risks caused by static configuration, and ensures the continuity and low latency of data processing in extreme scenarios such as high concurrency and sudden traffic. At the same time, the overall system cost is optimized through elastic resource scheduling, ultimately achieving a dynamic balance between reliability, efficiency, and economy.

[0033] Furthermore, based on the independent queues of each device, the three-layer protocol stack encapsulates and processes the data blocks in the independent queues step by step to obtain UDP frame data. The UDP frame is the encapsulation unit of the data block by the data link layer. The three-layer protocol stack includes the transport layer, network layer, and data link layer. Among them: the transport layer configures fixed source port number and destination port number, adds UDP header to the data block to form a UDP unicast packet; the network layer adds IP header to the UDP unicast packet and sets the address field of the IP header according to the target device address; the data link layer adds link layer frame information to the IP data packet, generates a UDP frame, and obtains the target MAC address through the address resolution protocol. The process involves sending UDP frame data to the receiving device and determining whether an acknowledgment signal is received within a preset thread time threshold. If an acknowledgment signal is received within the time threshold, the subsequent UDP frame data transmission task continues. If no acknowledgment signal is received within the time threshold, the current UDP frame data is stored in the retransmission array, and the current data frame timestamp and retry count are recorded. The main thread continues to execute the subsequent UDP frame data transmission task, thereby maintaining high throughput while ensuring data transmission reliability. The steps for transmission and reliability control of UDP frames encapsulated from data blocks in independent queues also include: using a retransmission thread to monitor the UDP frame data in the retransmission array in real time; if the retry count of any UDP frame data exceeds a preset retry count threshold, the UDP frame data is re-encapsulated and processed step-by-step based on the three-layer protocol stack of each device's independent queue, and then sent to the receiving device.

[0034] In this embodiment, the present invention establishes a dual-thread collaborative architecture of main thread and retransmission thread. The main thread can continuously send subsequent frames without waiting for acknowledgment, which greatly improves the throughput and real-time performance of data transmission. At the same time, an independent intelligent retransmission thread monitors the retransmission array, triggers on-demand retransmission based on retry count and timestamp threshold, and uses a three-layer protocol stack with independent queues to perform complete frame reconstruction, ensuring the final reliability of critical data transmission on unreliable networks and balancing transmission efficiency and data reliability.

[0035] Furthermore, the connection status parameters include authentication status, received signal strength, and reconnection count. After scanning the device's preset SSID information, the network status assessment steps, combining the SSID scan results and the device's connection status parameters, include: if the device's preset SSID information is not found within a preset time period, the preset SSID information is re-scanned after the device restarts; if the device's preset SSID information is still not found within the preset time period, a network anomaly warning is issued; if the device's preset SSID information is found within the preset time period, the process proceeds directly to step five; step five determines whether the device's authentication status is normal. The system monitors the validity of the device's authentication credentials in real time, comparing the identity submitted by the device... The system verifies the authorization records stored on the server. If the device fails to provide a valid digital certificate, the token expires, or the signature verification fails, the device is marked as having an abnormal authentication status. Furthermore, the system records detailed reasons for each authentication failure, including timestamps, error codes, and the authentication method used. If the device's authentication status is normal, the system proceeds directly to step six. Step six determines whether the device's average received signal strength during the monitoring period reaches a preset signal strength threshold. If yes, the system proceeds directly to step seven; otherwise, the device is marked as having an abnormal signal. Step seven determines whether the device's reconnection count exceeds a preset threshold. The system maintains a connection attempt counter for each device, statistically analyzing abnormal disconnections and reconnections within a unit of time. When the reconnection frequency exceeds the threshold set by the security policy, the system automatically analyzes the regularity of the reconnection intervals to distinguish between network jitter and device failure. Simultaneously, it correlates with the device's operation logs to check for accompanying hardware errors or resource exhaustion, avoiding misjudging temporary network fluctuations as device failures. If so, the device is marked as having abnormal reconnection; otherwise, it is marked as a normal device.

[0036] Specifically, the steps for network monitoring based on network status assessment results include: if a device is an authentication failure device, a re-authentication prompt is issued, and a forced authentication request is sent to the device. After receiving the instruction, the device needs to resubmit authentication credentials for verification. The system will record the result of each authentication attempt in real time. If the number of re-authentication attempts exceeds the preset threshold, an authentication failure prompt is issued, and the administrator is notified to intervene and investigate the problem. First, check whether the device key configuration is consistent with the authentication server. Second, verify whether network access permissions are incorrectly restricted. Finally, check whether the device clock synchronization is accurate, which may cause certificate expiration failure. Administrators need to conduct in-depth analysis based on the failure logs and device status snapshots provided by the system. If necessary, they should manually reset the device authentication status or update security credentials until the root cause of the problem is identified and fixed. Only then will the system remove the device authentication restrictions and restore its normal communication permissions. If the device has abnormal signal, a frequency band switching process is triggered. First, all pre-configured backup frequency bands are scanned, and the received signal strength of each band is measured in real time. These bands are then sorted in descending order of signal strength, and the strongest band is selected as the switching target. Subsequently, the hardware layer immediately adjusts the radio frequency parameters, including reconfiguring the center frequency, optimizing channel bandwidth, and switching modulation methods. Simultaneously, the protocol stack layer rebuilds the network connection, ensuring network continuity by reacquiring IP addresses and migrating TCP sessions. At the application layer, the system activates data buffering and intelligent retransmission mechanisms to ensure uninterrupted business data during the switching process, achieving a smooth transition. The entire switching process is recorded in detail in the system log, and the connection stability of the new frequency band is continuously monitored. If an anomaly occurs during the handover process, the system will gradually attempt to fall back to other available frequency bands according to a preset backoff algorithm until a stable connection is established, thereby ensuring the continuity and reliability of communication services. If the device is an abnormal reconnection device, a cooling-off process is triggered. First, the automatic reconnection function of the device is forcibly suspended for 1 minute. During this period, a deep diagnostic is performed simultaneously, verifying the operating status of key components such as sensors and RF modules through hardware self-test programs, and scanning device driver logs and system processes to check for potential software faults. If no obvious hardware damage or software anomaly is found during the diagnostic process, the system will automatically lift the reconnection restriction after the cooling-off period, allowing the device to reconnect to the network at gradual intervals. Conversely, if serious problems such as hardware failure or software crashes are detected, the system will immediately trigger a multi-level alarm mechanism, notifying technical personnel to intervene through audible and visual prompts, work order pushes, etc. If necessary, prompting the replacement of faulty components or the entire device, and archiving detailed fault codes and diagnostic reports for subsequent analysis, thereby achieving accurate isolation and rapid recovery of faulty devices while ensuring system stability. If the device is a normal device, adaptive playback control is directly implemented.

[0037] Specifically, the steps for adaptive playback control of UDP frames received by the device after network monitoring processing include: decapsulating the UDP frames received by the device based on the LWIP protocol stack at each level. The decapsulation process includes: Step eight, performing transmission error verification on the UDP data frames received by the device. The LWIP protocol stack first performs hard verification of the IP data packets at the network layer, including checking if the IP version number conforms to the specification and recalculating the IP header checksum. If a version mismatch or checksum error is found, the protocol stack will directly discard the problematic data packet and issue a transmission error prompt. If the network layer automatic verification is successful, it enters the transport layer for secondary transmission verification, further checking the basic compliance of the UDP datagram and confirming that the data length is not less than the fixed 8 bytes of the UDP header. If an abnormality such as length abnormality is detected, the transmission error handling process is also triggered to ensure that only complete and compliant data packets enter step nine; Step nine, determining whether the length of the UDP frames received by the device conforms to the preset byte alignment requirements, extracting the effective payload from the UDP data packets that have passed the transmission verification, and obtaining the starting memory address of the payload. The total byte length is checked. If the payload's starting address is a multiple of 4, the length alignment is verified. The total payload length is calculated, and the length value is checked for divisibility by 4. If divisible, the process proceeds directly to step ten; otherwise, an alignment error message is issued. In step ten, the aligned audio data is converted into a sample value array. The audio sample value of the UDP frame is calculated using the root mean square algorithm. It is determined whether the audio sample value of the UDP frame received by the device is lower than a preset audio amplitude value. If so, the UDP frame received by the device is marked as a mute packet; otherwise, it is marked as valid data. The valid data is stored in a circular buffer, and an acknowledgment signal is returned. A circular buffer is a first-in, first-out (FIFO) data structure. Its physical structure is a continuous linear memory space, and its logical structure is a circular storage space with its ends connected. The core components of the circular buffer include a write pointer, a read pointer, and a buffer counter. The write pointer points to the next writable location, the read pointer points to the next readable location, and the buffer counter records the number of valid data blocks in real time. When writing, valid data is stored at the write pointer position. After storage, the write pointer moves forward one step to the next empty position. If the write pointer has moved to the last position of the buffer, it will loop back to the beginning of the buffer. Each time a data block is successfully written, the buffer counter increments by one. When reading, data is retrieved from the read pointer position. After retrieval, the read pointer moves forward one step to point to the next data block to be read. Similarly, if the read pointer has moved to the last position of the buffer, it will also loop back to the beginning and monitor the available data in the buffer in real time. Each time a data block is successfully read, the buffer counter decrements by one.Audio playback begins when the available buffer data in the circular buffer reaches a preset start threshold. If the available buffer data does not reach the preset first start threshold, a single data block output mode is used, with data transmission occurring at the first interval. If the available buffer data reaches the preset first start threshold but not the preset second start threshold, a double data block output mode is used, with data transmission occurring at the second interval. If the available buffer data reaches the preset second start threshold, a triple data block output mode is used, with data transmission occurring at the third interval. No additional processing is performed when the available buffer data in the circular buffer does not reach the preset start threshold. The start threshold is the minimum amount of buffer data required to start playback. When the available buffer data reaches the start threshold, it indicates that the circular buffer currently has enough data to start audio playback and can withstand a certain degree of network fluctuation, thus avoiding immediate stuttering.

[0038] Furthermore, during audio playback, this invention transmits the processed digital audio signal to a digital-to-analog converter and power amplifier via an I2S digital audio bus configured in master mode for digital-to-analog conversion and signal amplification output. The I2S master mode interface is configured to operate in master device mode, using 16-bit data precision and a sampling rate of 8kHz. The left and right channels transmit the same audio data, and data transmission is achieved through a DMA controller. The configuration is as follows: a single buffer length of 512 bytes, with four buffers forming a circular buffer queue. This configuration reduces processor load and minimizes data transmission congestion. During system operation, this invention provides real-time feedback on the system's operating status via multi-color LEDs: red indicates an abnormal network connection; green indicates a normal network connection and a muted state; blue indicates normal audio playback; and orange indicates a buffer data overload warning.

[0039] In this embodiment, the call interval refers to the time interval between two consecutive data output operations. Dynamically adjusting the call interval balances data transmission real-time performance and system resource utilization. The interval is related to the current output mode; the more data blocks output, the shorter the interval. The single-data-block output mode outputs one data block at a time, suitable for situations with insufficient buffered data, reducing processing frequency by extending the interval. The dual-data-block output mode outputs two data blocks at a time, suitable for situations with normal buffered data, maintaining a standard processing rate. The triple-data-block output mode outputs three data blocks at a time, suitable for situations with sufficient buffered data, increasing throughput by shortening the interval. This invention, by integrating multi-dimensional indicators such as authentication status, signal strength, and reconnection behavior, accurately classifies devices and triggers differentiated processing strategies. It not only automatically performs tiered verification and alarm escalation for abnormal authentication devices, intelligently triggers frequency band switching for devices with degraded signals to optimize link quality, and implements cooling and frequency reduction for abnormal reconnection devices to prevent system overload, but also ensures interference-free adaptive control of normal devices, significantly improving device management efficiency, system stability, and network resource utilization, providing core guarantees for the secure access and reliable operation of large-scale IoT terminals.

[0040] This application also provides a wearable bone conduction hearing aid device, including: a processor; a memory, the memory storing computer-readable instructions, which, when executed by the processor, realize the wearable bone conduction hearing aid system.

[0041] Those skilled in the art will understand that embodiments of the present invention can be provided as methods, systems, or computer program products. Therefore, the present invention can take the form of a completely hardware embodiment, a completely software embodiment, or an embodiment combining software and hardware aspects. Furthermore, the present invention can take the form of a computer program product embodied on one or more computer-usable storage media (including, but not limited to, disk storage, CD-ROM, optical storage, etc.) containing computer-usable program code.

[0042] This invention is described with reference to flowchart illustrations and / or block diagrams of systems, apparatus (systems), and computer program products according to embodiments of the invention. It will be understood that each block of the flowchart illustrations and / or block diagrams, and combinations of blocks in the flowchart illustrations and / or block diagrams, can be implemented by computer program instructions. These computer program instructions can be provided to a processor of a general-purpose computer, special-purpose computer, embedded processor, or other programmable data processing apparatus to produce a machine, such that the instructions, which execute via the processor of the computer or other programmable data processing apparatus, generate instructions for implementing the flowchart illustrations and / or block diagrams. Figure 1 One or more processes and / or boxes Figure 1 A device that provides the functions specified in one or more boxes.

[0043] These computer program instructions may also be stored in a computer-readable storage medium that can direct a computer or other programmable data processing device to function in a particular manner, such that the instructions stored in the computer-readable storage medium produce an article of manufacture including instruction means, which are implemented in a process Figure 1 One or more processes and / or boxes Figure 1 The function specified in one or more boxes.

[0044] These computer program instructions may also be loaded onto a computer or other programmable data processing equipment to cause a series of operational steps to be performed on the computer or other programmable equipment to produce a computer-implemented process, thereby providing instructions that execute on the computer or other programmable equipment for implementing the process. Figure 1 One or more processes and / or boxes Figure 1 The steps of the function specified in one or more boxes.

[0045] Although preferred embodiments of the invention have been described, those skilled in the art, upon learning the basic inventive concept, can make other changes and modifications to these embodiments. Therefore, the appended claims are intended to be interpreted as including both the preferred embodiments and all changes and modifications falling within the scope of the invention.

[0046] Obviously, those skilled in the art can make various modifications and variations to this invention without departing from its spirit and scope. Therefore, if these modifications and variations fall within the scope of the claims of this invention and their equivalents, this invention also intends to include these modifications and variations.

Claims

1. A wearable bone conduction hearing aid system, characterized in that, It includes an audio processing module, a data transmission module, and an audio receiving module; The audio processing module is used to perform file standardization evaluation on continuously read audio files based on data condition parameters, and to perform standardization processing on the audio files based on the file standardization evaluation results. The standardization processing means that the application layer converts the continuously read audio files into a format that conforms to preset data condition parameters, and then converts them into data blocks of a specified format. The data transmission module is used to transmit the data blocks obtained after standardization to the independent queues of each device, evaluate the queue status based on the transportation status parameters of each device during the data block transmission process, and make dynamic queue adjustments based on the queue status evaluation results. Then, it performs transmission and reliability control on the UDP frames encapsulated from the data blocks in the independent queues. The dynamic queue adjustment means that the data production rate of the independent queues is dynamically adjusted according to the transportation status parameters to reduce the risk of data block overflow in the independent queues. The audio receiving module is used to scan the device's preset SSID information, combine the SSID scanning results with the device's connection status parameters to perform network status assessment, and perform network monitoring processing based on the network status assessment results. Then, it performs adaptive playback control on the UDP frames received by the device after network monitoring processing. The network monitoring processing means performing real-time analysis and processing based on the connection status parameters of the device's receiving end to improve the reliability of network transmission and connection stability.

2. The wearable bone conduction hearing aid system as described in claim 1, characterized in that, The data condition parameters include sampling points, bit depth, sampling rate, and audio channels; The number of bytes in a data block is obtained based on the sampling points and bit depth of the audio file; The audio duration of a data block is obtained based on the sampling points and sampling rate of the audio file; The steps for evaluating the file standardization of continuously read audio files based on data condition parameters include: Step 1: Determine if the number of bytes in each data block of the audio file meets the data block byte count requirement. If yes, proceed directly to Step 2; otherwise, mark the file as a bit depth abnormal file. Step 2: Determine whether the audio duration of each data block in the audio file meets the data block audio duration requirement. If yes, proceed directly to Step 3; otherwise, mark the audio file as a file with an abnormal sampling rate. Step 3: Determine whether the number of channels in the audio file meets the preset number of channels. If yes, proceed directly to Step 4; otherwise, mark the audio file as a channel abnormality file. Step 4: Determine whether the total number of samples in the audio file is an integer multiple of the number of samples. If so, mark the audio file as a normal file; otherwise, mark the audio file as a file with data alignment error.

3. The wearable bone conduction hearing aid system as described in claim 2, characterized in that, The steps for standardizing audio files based on file standardization evaluation results include: If the audio file is a bit depth abnormal file, then the sample points in the audio file whose numerical range does not meet the preset bit depth numerical range will be converted to the preset bit depth numerical range. If the audio file has an abnormal sampling rate, a resampling prompt will be issued; If the audio file is a channel abnormal file, determine whether the phase difference between the left and right channels in the audio file is consistent. If it is, extract a single channel directly; otherwise, take the average value of the corresponding sampling points of the left and right channels of the audio file. If the audio file is a data alignment error file, determine whether there is missing data in the audio file. If so, repair the data by interpolation. Otherwise, pad the end of the audio file with zeros until the total number of samples in the audio file is an integer multiple of the number of samples. If the audio file is a normal file, the volume gain of the input audio signal is amplified to a preset multiple, the mono audio data is copied to the left and right channels to generate stereo-compatible audio data, and then the processed stereo audio data is divided into data blocks of a preset sample length.

4. The wearable bone conduction hearing aid system as described in claim 1, characterized in that, The transportation status parameters include queue blocking time, queue occupancy rate, and enqueue failure rate. The steps for evaluating the queue status based on the transportation status parameters of each device during data block transmission include: Obtain preset transportation status parameter reference data and transportation status parameter contribution data. The transportation status parameter reference data includes critical queue blocking time, critical queue occupancy rate, and critical enqueue failure rate. The transportation status parameter contribution data includes queue blocking time contribution, queue occupancy rate contribution, and enqueue failure rate contribution. The transportation status parameter reference data is used to calculate the proportion convergence with the transportation status parameters of each device. Then, the corresponding proportion convergence calculation results are weighted and coupled using the transportation status parameter contribution data to obtain the transportation status evaluation index of each device. The transportation status evaluation index is a comprehensive indicator that quantifies the transportation status of the device based on the transportation status parameters.

5. The wearable bone conduction hearing aid system as described in claim 4, characterized in that, The steps for dynamic queue adjustment based on queue status evaluation results include: The transportation status evaluation index of each device is compared with the preset first threshold and second threshold of transportation status evaluation. If the transportation status evaluation index of any device is lower than the first threshold of transportation status evaluation, the data production rate of the device is adjusted to the preset minimum production rate. If the transportation status evaluation index of the device is still lower than the first threshold of transportation status evaluation after the data production rate of the device is adjusted to the preset minimum production rate, and the duration of the state exceeds the preset time window, the standby queue is opened until the transportation status evaluation index of the device is not lower than the first threshold of transportation status evaluation. If the transportation status assessment index of any device is not lower than the first threshold of transportation status assessment, but is lower than the second threshold of transportation status assessment, then the data production rate is dynamically adjusted according to the transportation status assessment index of the device based on the data production rate mapping table, until the transportation status assessment index of the device is not lower than the second threshold of transportation status assessment. If the transportation status assessment index of any device is not lower than the second threshold for transportation status assessment, no additional processing will be performed.

6. The wearable bone conduction hearing aid system as described in claim 1, characterized in that, The steps for transmission and reliability control of UDP frames encapsulated from data blocks in independent queues include: Based on the three-layer protocol stack of each device's independent queue, the data blocks in the independent queue are encapsulated and processed step by step to obtain UDP frame data; The UDP frame data is sent to the receiving end of the device. It is determined whether the acknowledgment signal from the receiving end of the device is received within the preset thread time threshold. If the acknowledgment signal is received within the time threshold, the subsequent UDP frame data sending task is continued. If no response signal is received within the time threshold, the current UDP frame data is stored in the retransmission array, and the timestamp and retry count of the current data frame are recorded. The main thread continues to execute the subsequent UDP frame data transmission task. The step of performing transmission and reliability control on UDP frames encapsulated from data blocks in independent queues further includes: The retransmission thread monitors the UDP frame data of the retransmission array in real time. If the retry count of any UDP frame data exceeds the preset retry count threshold, the UDP frame data is re-encapsulated and processed step by step based on the three-layer protocol stack of each device's independent queue, and then sent to the device receiving end.

7. The wearable bone conduction hearing aid system as described in claim 1, characterized in that, The connection status parameters include authentication status, received signal strength, and number of reconnections. After obtaining the preset SSID information from the scanning device, the steps for evaluating the network status by combining the SSID scanning results and the device's connection status parameters include: If the device's default SSID information is not found within the preset time period, the device will be restarted and the default SSID information will be scanned again. If the device's default SSID information is still not found within the preset time period, a network error message will be issued. If the device's preset SSID information is found within the preset time period, proceed directly to step five; Step 5: Determine if the device's authentication status is normal. If it is, proceed directly to Step 6; otherwise, mark the device as having an abnormal authentication status. Step 6: Determine whether the average received signal strength of the device during the monitoring period reaches the preset signal strength threshold. If yes, proceed directly to Step 7; otherwise, mark the device as a device with abnormal signal. Step 7: Determine whether the number of reconnections by the device exceeds the preset threshold. If so, mark the device as an abnormal reconnection device; otherwise, mark the device as a normal device.

8. The wearable bone conduction hearing aid system as described in claim 7, characterized in that, The steps for network monitoring based on network status assessment results include: If the device is an authentication error device, a re-authentication prompt will be issued. If the number of re-authentication attempts exceeds the preset threshold, an authentication error prompt will be issued. If the device is a signal abnormal device, a signal abnormality prompt will be issued, triggering the frequency band switching process; If the device is an abnormal reconnection device, a reconnection error message will be issued, triggering the cooling process. If the device is a normal device, adaptive playback control will be implemented directly.

9. The wearable bone conduction hearing aid system as described in claim 1, characterized in that, The steps for adaptive playback control of UDP frames received by the device after network monitoring processing include: Based on the LWIP protocol stack, the device receives UDP frames after network monitoring processing through a step-by-step decapsulation process. The decapsulation process includes the following steps: Step 8: Perform transmission error verification on the UDP data frames received by the device. If the transmission error verification fails, issue a transmission error prompt; otherwise, proceed directly to step 9. Step 9: Determine whether the length of the UDP frame received by the device meets the preset byte alignment requirements. If yes, proceed directly to Step 10; otherwise, issue an alignment error message. Step 10: Determine whether the audio sample value of the UDP frame received by the device is lower than the preset audio amplitude value. If so, mark the UDP frame received by the device as a mute packet; otherwise, mark the UDP frame received by the device as valid data. Store valid data in a circular buffer and return an acknowledgment signal; The core components of the circular buffer include a write pointer, a read pointer, and a buffer counter. The write pointer points to the next writable location, the read pointer points to the next readable location, and the buffer counter records the number of valid data blocks in real time. When the amount of available buffered data in the circular buffer reaches the preset start threshold, if the amount of available buffered data does not reach the preset first start threshold, the single data block output mode is adopted and the first interval time is used for data transmission. If the available buffer data reaches the preset first start threshold but does not reach the preset second start threshold, then the dual data block output mode is adopted, and the data is transmitted at the second interval. If the amount of available buffered data reaches the preset second start threshold, then the three-data-block output mode is adopted, and the data is transmitted at the third interval. No additional processing is performed when the amount of available buffered data in the circular buffer does not reach the preset start threshold.

10. A wearable bone conduction hearing aid, characterized in that, The wearable bone conduction hearing aid includes: processor; A memory storing computer-readable instructions, which, when executed by the processor, implement the function of the wearable bone conduction hearing aid system as described in any one of claims 1 to 9.

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