Smart hearing aid system based on remote sound high-fidelity capturing technology

Abstract

This invention relates to the field of hearing aid technology, specifically to an intelligent hearing aid system based on remote high-fidelity sound capture technology. The system includes a remote sound pickup module, a wireless transmission module, an intelligent processing module, and an audio output module. The remote sound pickup module uses a high-sensitivity microphone array and beamforming technology to collect sound. The signal is transmitted wirelessly to the intelligent processing module, where it undergoes noise reduction, enhancement, and adaptive equalization processing before being delivered to the user by the audio output module. This system achieves remote high-fidelity sound capture, effectively improving sound recognition and processing capabilities in complex environments and enhancing the user's hearing experience.

Description

Technical Field

[0001] This invention relates to the field of hearing aid technology, specifically to an intelligent hearing aid system based on remote high-fidelity sound capture technology. Through innovative sound acquisition, transmission, processing, and output technologies, it provides hearing-impaired patients with a clearer, more natural, and personalized auditory experience, and is suitable for various complex acoustic environments. Background Technology

[0002] Traditional hearing aids suffer from significant shortcomings in sound acquisition and processing. On one hand, the design of a single microphone or a simple microphone array struggles to distinguish between the target sound source and ambient noise. In noisy environments such as conference rooms and shopping malls, users often cannot clearly capture conversations due to background noise interference. For example, in environments with background noise levels above 60dB, the speech recognition accuracy of ordinary hearing aids is less than 50%, severely impacting the user experience. On the other hand, existing noise reduction algorithms often use fixed parameters, making them unable to adapt to dynamically changing noise environments, easily leading to speech signal distortion and loss of high-frequency details. Furthermore, traditional hearing aids lack the ability to effectively capture distant sound sources. When the target sound source is more than 1 meter away, sound attenuation and signal-to-noise ratio decline become prominent issues, and the unoptimized microphone array layout results in remote pickup latency as high as 50ms. As users' performance requirements for hearing aids in complex scenarios increase, there is an urgent need for a new system capable of achieving high-fidelity remote pickup and intelligent environmental adaptation. Summary of the Invention

[0003] This invention aims to solve the technical bottlenecks in sound acquisition, transmission and processing of traditional hearing aids. Through multi-module collaborative innovation, it achieves high-fidelity capture and intelligent processing of remote sound, improving the sound perception accuracy and ease of use for hearing-impaired patients in various environments.

[0004] The technical solution adopted by this invention to solve its technical problem is: an intelligent hearing aid system based on remote high-fidelity sound capture technology, comprising: The remote sound pickup module includes a microphone array consisting of multiple high-sensitivity microphones, used to collect sound signals from the target sound source and enhance the target sound source signal through beamforming algorithms while suppressing ambient noise. The wireless transmission module is used to transmit the sound signal collected by the remote pickup module to the intelligent processing module in a low-latency, high-fidelity manner; The intelligent processing module performs noise reduction, dynamic range compression, adaptive equalization adjustment, and deep learning-based speech enhancement on the received audio signal. The audio output module converts the sound signal processed by the intelligent processing module into audio output to the user; wherein, the beamforming algorithm calculates the weighting coefficients of each microphone signal based on the geometric position of the microphone array and the sound propagation characteristics, and the weighting coefficients are expressed by the formula... calculate, For the first The distance from each microphone to the target sound source For the first The time delay of each microphone relative to the reference microphone, The frequency of the sound.

[0005] Specifically, the remote sound pickup module employs MEMS microphones, with the microphone array using either a uniform circular array or a linear array layout. The uniform circular array offers excellent sound capture uniformity across a 360-degree horizontal range, making it suitable for scenarios requiring omnidirectional sound capture, such as conference discussions and multi-person conversations. Taking a 6-microphone uniform circular array with an 8cm diameter as an example, by optimizing the microphone spacing and layout, precise beamforming of the target sound source can be achieved over a wide frequency range, with a sound source localization error of less than 5 degrees within a 5-meter radius. The linear array, on the other hand, offers high sound gain and noise suppression capabilities in a specific direction, making it suitable for scenarios where the direction of the target sound source is clearly defined, such as lectures and classroom listening. Using a 4-microphone linear array with a 3cm microphone spacing, a gain improvement of over 18dB can be achieved in the direction of the target sound source, while the suppression of side noise reaches 22dB. Furthermore, in terms of manufacturing processes, the linear array layout facilitates integration with the hearing aid's shell, reducing production difficulty.

[0006] Specifically, the wireless transmission module adopts Bluetooth 5.3 or a dedicated 2.4GHz wireless transmission protocol, with a transmission bandwidth no less than the transmission requirements of 16-bit audio at a sampling rate of 48kHz. Bluetooth 5.3, with its mature technological ecosystem, boasts broad device compatibility, seamlessly connecting with smartphones, tablets, and other mobile terminals, allowing users to remotely control and adjust parameters of the hearing aid via an app. It supports LEAudio technology, enabling low-power, high-quality audio transmission. During audio stream transmission, audio latency can be controlled within 30ms, meeting general voice interaction needs. The dedicated 2.4GHz wireless transmission protocol is tailor-made for hearing aid systems. By optimizing channel coding and modulation methods, such as using Orthogonal Frequency Division Multiplexing (OFDM) technology, transmission latency can be further reduced to 15ms. Furthermore, in complex electromagnetic environments, a dynamic channel switching mechanism effectively avoids interference frequency bands, ensuring stable audio signal transmission. Even in crowded places with complex wireless signals, such as shopping malls and subway stations, the packet loss rate can still be controlled below 0.5%.

[0007] Specifically, the intelligent processing module includes a digital signal processor, the noise reduction processing adopts a Wiener filtering-based noise reduction algorithm, and the dynamic range compression is achieved through the formula... accomplish, This is the compression ratio coefficient. The compression curve is represented by an exponent. In practical applications, the Wiener filtering-based noise reduction algorithm adapts to constantly changing noise environments by updating the noise power spectrum estimate in real time. The system estimates and updates the noise power spectrum every 100ms. In sudden noise scenarios, such as car horns or door slamming, it can quickly respond and adjust noise reduction parameters, increasing noise suppression to over 25dB while maintaining the integrity of the speech signal. For the dynamic range compression section, the system presets multiple k and n combination modes to meet the different hearing needs of users. For example, for users with severe hearing loss, a more advanced k-n combination is used. , The combination of these two technologies has a more significant effect on enhancing weak sounds, raising the originally barely audible 40dB weak sound to about 60dB, while effectively compressing strong sounds (such as 90dB) to stabilize the output at 75dB, ensuring the user's auditory comfort.

[0008] Specifically, the audio output module is a bone conduction headphone or an in-ear speaker, and it features adaptive volume adjustment. Bone conduction headphones transmit sound to the auditory nerve by vibrating the skull, eliminating the need to block the ear canal. This ensures clear sound transmission while allowing users to perceive ambient sounds, improving safety and making them suitable for sports, outdoor activities, and other similar scenarios. The built-in vibration unit uses high-performance neodymium iron boron magnets, achieving a sound pressure level variation of less than 10dB within the 100-8000Hz frequency range, ensuring accurate sound reproduction. The in-ear speaker uses a customized balanced armature driver, and through a precisely tuned crossover network, it achieves a flat response of ±3dB in the high-frequency range (4000-8000Hz), providing users with clear and detailed sound. The adaptive volume adjustment function monitors ambient noise intensity in real time using a built-in environmental noise sensor. When the ambient noise intensity exceeds 60dB, the system automatically increases the volume by 3-5dB; when the noise intensity is below 40dB, the volume decreases by 2-3dB, ensuring a comfortable listening experience for users in various environments.

[0009] Specifically, it also includes a user interaction module for receiving user commands and adjusting parameters such as volume and noise reduction level of the hearing aid. The user interaction module uses a combination of physical buttons and touch control. The physical buttons are designed as raised buttons, allowing for blind operation without looking at the hearing aid, with clear button feedback and distinct tactile feedback. The touch area uses capacitive touch technology, supporting various gestures such as swiping and clicking. For example, users can swipe up and down on the touch area of ​​the hearing aid shell to adjust the volume, and swipe left and right to switch noise reduction levels. In addition, the user interaction module also supports voice control. Through a built-in voice recognition chip, it can recognize user voice commands such as "increase volume" and "turn on noise reduction mode," with a recognition accuracy of over 95%, providing users with a more convenient and diverse operating method.

[0010] Specifically, the intelligent processing module incorporates a deep learning-based speech recognition model capable of recognizing different scenarios and automatically switching sound processing modes. This deep learning-based speech recognition model employs an architecture combining convolutional neural networks (CNN) and long short-term memory networks (LSTM), trained with a large amount of speech data covering various scenarios such as conference rooms, shopping malls, streets, and homes. In practical applications, the system extracts features and identifies scenarios from the collected sound signals every second. When a conference room scenario is identified, it automatically switches to conference mode to enhance the speech signal and suppress environmental reverberation. When a shopping mall scenario is identified, a strong noise reduction mode is activated to focus on suppressing noisy human voices and background noise while improving the clarity of the target speech. Testing shows that the scene recognition accuracy exceeds 92% in different scenarios, and the sound processing mode switching is timely and accurate, effectively improving the user's listening experience in various environments.

[0011] Specifically, the system also includes a charging management module that uses wireless charging technology to power the system. The charging management module adopts the Qi wireless charging standard and is compatible with most wireless chargers on the market. During charging, it features overvoltage, overcurrent, and overtemperature protection functions. When the charging voltage exceeds 5.5V, it automatically cuts off the charging circuit; when the charging current exceeds 500mA, it reduces the charging power; and when the charging temperature exceeds 50℃, it pauses charging and activates a cooling mechanism. In terms of charging efficiency, using a 10W wireless charger, the system battery can be charged from 0% to 80% in 2 hours and fully charged in 3 hours. Furthermore, the wireless charging design avoids the problems of easily damaged or poorly contacted traditional charging interfaces, improving the system's durability and ease of use.

[0012] Specifically, the remote microphone module features a waterproof and dustproof structure with a protection rating of at least IP67. The module's housing is made of high-strength polycarbonate material, combined with precise sealing technology. Silicone rubber sealing rings are used to seal the seams, and the microphone area is covered with a waterproof and breathable membrane, ensuring normal sound transmission while preventing water and dust ingress. In waterproof testing, after immersing it in 1 meter of water for 30 minutes, the insulation resistance of the internal circuitry remained above 100MΩ, and all functions operated normally. In dustproof testing, after operating it in an environment with a sand concentration of 1000mg / m³ for 8 hours, no sand or dust entered, and the microphone sensitivity changed by less than 5%, ensuring stable and reliable operation of the system even in harsh environments.

[0013] Specifically, the system supports connection to mobile terminals via an app, enabling remote adjustment and personalized settings. The mobile app interface features a simple and intuitive design, allowing users to view real-time information such as the hearing aid's battery level and connection status. For remote adjustment, users can upload their hearing test data to the system via the app, and the system automatically generates a personalized sound processing solution based on the data. Users can also manually adjust equalization parameters, noise reduction levels, volume, and other settings, with the adjusted parameters being wirelessly transmitted to the hearing aid in real time. Furthermore, the app has a data statistics function, recording daily usage time, usage frequency in different scenarios, and generating usage reports to help users and professionals understand hearing aid usage and hearing change trends, providing a basis for further optimization of hearing solutions.

[0014] This invention achieves technological breakthroughs across multiple dimensions, including sound acquisition, processing, interaction, and durability, through innovative design of remote high-fidelity sound capture technology and intelligent processing algorithms. Compared to traditional hearing aids, it exhibits significant advantages in performance, user experience, and practicality, as detailed below: 1. High-fidelity remote sound pickup: Overcoming distance and noise limitations. Traditional hearing aids are limited by single microphones or simple array designs. When the distance to the target sound source exceeds 1 meter, the sound signal attenuation and environmental noise interference cause a significant decrease in clarity. The remote sound pickup module of this invention uses a high-sensitivity MEMS microphone array and beamforming technology to achieve high-fidelity sound acquisition within a radius of 5 meters. (The formula is used to...) By dynamically calculating weighting coefficients, the system can accurately enhance the target sound source signal while suppressing environmental noise.

[0015] Actual testing showed that in a 60dB background noise environment, the speech recognition accuracy of traditional hearing aids is less than 50%, while this system can improve the speech recognition accuracy to 82% and increase the sound acquisition signal-to-noise ratio by 35%. For example, in a conference room setting, even if the user is sitting far away from the speaker, they can clearly capture the dialogue, effectively solving the problems of traditional hearing aids being "unable to hear clearly and unable to hear far".

[0016] 2. Intelligent scene adaptation: Dynamically optimize sound processing strategies Traditional hearing aids typically employ fixed-parameter noise reduction and sound processing modes, which cannot adapt to complex and ever-changing acoustic environments. The intelligent processing module of this invention incorporates a deep learning-based speech recognition model. By combining a convolutional neural network (CNN) with a long short-term memory network (LSTM), it has been trained on 200,000 hours of speech data from eight typical scenarios, including shopping malls, conference rooms, and streets, achieving a scene recognition accuracy of up to 92%.

[0017] The system can automatically switch sound processing modes based on the real-time identified scene: in a shopping mall setting, it automatically activates a strong noise reduction mode to suppress noisy human voices and background noise while improving the clarity of the target speech; in a conference room setting, it enhances the speech signal and suppresses environmental reverberation. This intelligent adaptation mechanism reduces the frequency of manual adjustments by the user, ensuring that the hearing aid maintains optimal performance in different environments and significantly improves the user's hearing experience in complex scenarios.

[0018] 3. Multimodal interactive experience: Meeting diverse operational needs Traditional hearing aids have a limited operation method, typically relying solely on physical buttons, which is not user-friendly for elderly users or those unfamiliar with operation. The user interaction module of this invention employs a triple interaction method: physical buttons, touch control, and voice control, catering to the usage habits of different user groups.

[0019] The physical buttons feature a raised design with a textured, non-slip surface, facilitating blind operation. The buttons offer clear feedback and a distinct tactile feel. The capacitive touch area supports various gesture controls, such as swiping up and down to adjust volume and swiping left and right to switch noise cancellation levels, providing intuitive and convenient operation. Furthermore, the voice recognition function supports natural language commands like "increase volume" and "switch modes," achieving an accuracy rate of over 95%. Users can quickly adjust hearing aid settings without manual intervention. The system also connects to a mobile app, allowing users to fine-tune parameters and monitor status via their smartphones, further enhancing convenience and personalization.

[0020] 4. Durability and Convenience: Adaptable to all usage scenarios Traditional hearing aids' wired charging interfaces are prone to damage from frequent plugging and unplugging, and their poor waterproof and dustproof performance limits their usage scenarios. This invention's charging management module adopts the Qi wireless charging standard, avoiding the contact problems of traditional interfaces. It also features overvoltage (>5.5V), overcurrent (>500mA), and overtemperature (>50℃) protection functions, ensuring a safe and reliable charging process. With the support of a 10W wireless charger, the system can be charged to 80% in 2 hours and fully charged in 3 hours, significantly improving charging efficiency.

[0021] The remote microphone module achieves an IP67 protection rating and has undergone rigorous waterproof and dustproof testing: after being submerged in 1 meter of water for 30 minutes, the internal circuit insulation resistance remains above 100MΩ, and all functions operate normally; after operating for 8 hours in an environment with a dust concentration of 1000mg / m³, no dust entered the internal components, and the microphone sensitivity changed by less than 5%. This allows the system to adapt to various harsh environments such as rainy days, outdoor sports, and factory operations, significantly expanding its application scenarios and improving the product's durability and reliability.

[0022] 5. Data-driven optimization: supporting hearing health management Traditional hearing aids lack data recording and analysis capabilities, failing to provide effective support for users' hearing health management. This invention enables data-driven hearing optimization through a mobile app. The app can record hearing aid usage time, frequency of use in different scenarios, volume adjustment records, and other data in real time, generating a visual usage report.

[0023] This data not only helps users intuitively understand the usage of their hearing aids, but also provides hearing professionals with quantitative data to assess hearing change trends and adjust hearing intervention strategies. For example, by analyzing users' volume adjustment data in different environments, professionals can optimize the sound processing parameters of hearing aids to achieve more precise and personalized hearing solutions, driving the upgrade of hearing aids from simple sound amplification devices to intelligent hearing health management tools. Attached Figure Description

[0024] The present invention will be further described below with reference to the accompanying drawings and embodiments.

[0025] Figure 1 A block diagram of the intelligent hearing aid system based on remote high-fidelity sound capture technology provided by the present invention. Detailed Implementation

[0026] To make the technical means, creative features, objectives and effects of this invention easier to understand, the invention will be further described below in conjunction with specific embodiments.

[0027] like Figure 1 As shown, the intelligent hearing aid system based on remote high-fidelity sound capture technology of the present invention includes a remote sound pickup module, a wireless transmission module, an intelligent processing module, an audio output module, a user interaction module, a charging management module, and supports connection with a mobile terminal APP.

[0028] The remote microphone pickup module uses a MEMS microphone array, supporting either a uniform circular array (UCA) or a linear array (ULA) layout. Taking a 6-microphone uniform circular array (8cm in diameter) as an example, beamforming algorithms (formula...) are used... This algorithm achieves sound source localization and signal enhancement. It dynamically adjusts weighting coefficients based on microphone spacing, geometric position, and sound propagation delay, boosting the gain of the target sound source by 18dB within a 5-meter radius, with a directional localization error of less than 5°. The array design balances acoustic performance with device integration; a linear array (e.g., 4 microphones, 3cm spacing) can achieve 22dB noise suppression in a specific direction (±30°), suitable for classroom lectures, presentations, and other similar scenarios.

[0029] The wireless transmission module supports dual modes: Bluetooth 5.3 and a dedicated 2.4GHz protocol. Bluetooth 5.3 utilizes LEAudio technology to achieve low-power transmission with latency controlled within 30ms, making it compatible with smartphones, tablets, and other devices. The dedicated 2.4GHz protocol employs OFDM modulation technology and avoids interference bands through a dynamic channel switching mechanism, achieving a packet loss rate of less than 0.5% and a transmission latency of only 15ms in complex electromagnetic environments, ensuring smooth real-time voice interaction.

[0030] The intelligent processing module integrates a high-performance digital signal processor (DSP) and has built-in multi-dimensional processing algorithms: Noise reduction is achieved through a Wiener filtering-based dynamic noise reduction algorithm, which updates the noise power spectrum estimate every 100ms. This algorithm can quickly suppress interference exceeding 25dB in sudden noise scenarios while preserving the integrity of the speech signal. Dynamic range compression is implemented using the formula... To achieve adaptive adjustment, the system presets five parameter combinations (such as the severe hearing loss mode). , It can boost weak sounds from 40dB to 60dB and compress strong sounds from 90dB to 75dB. Deep learning speech enhancement uses a CNN-LSTM hybrid model, trained on 200,000 hours of multi-scene speech data, achieving a scene recognition accuracy of 92%, and automatically switching between eight processing modes such as meeting and shopping mall modes.

[0031] The audio output module offers both bone conduction and in-ear modes. The bone conduction headphones use neodymium iron boron vibrating units, with a sound pressure level fluctuation of <10dB in the 100-8000Hz frequency range, ensuring natural sound. The in-ear speakers are equipped with custom balanced armature units, with a high-frequency response of ±3dB and support for adaptive volume adjustment. The volume is adjusted in real time through an ambient noise sensor (increased by 3-5dB when noise >60dB, and decreased by 2-3dB when noise <40dB).

[0032] The user interaction module employs a triple interaction method: physical buttons, touch control, and voice control. The physical buttons are designed for blind operation, and the touch area supports swipe and click gestures. The voice recognition chip integrates keyword wake-up functionality, achieving a 95% accuracy rate in recognizing commands such as "increase volume" and "switch modes."

[0033] The charging management module is based on Qi standard wireless charging technology and features overvoltage (>5.5V), overcurrent (>500mA), and overtemperature (>50℃) protection. With 10W wireless charging, it can charge to 80% in 2 hours, completely solving the problem of easily damaged traditional interfaces.

[0034] The mobile app provides real-time status monitoring, personalized parameter adjustment, and usage data statistics. Users can upload audiograms to generate a customized equalization solution, and the app records daily usage time, scene distribution, and other data to help optimize hearing intervention strategies.

[0035] Example 1: Microphone Array Layout and Beamforming Optimization In this embodiment, the remote sound pickup module uses four MEMS microphones arranged in a square uniform array with a side length of 5cm. Experiments were conducted to test the impact of different microphone spacing and layout on beamforming effects, determining the optimal layout parameters. In the experiment, a target sound source was placed 2 meters away from the array, and noise sources were arranged around it to simulate a noisy environment. The formula was used... The weighting coefficients are calculated, and the beam direction is adjusted to point towards the target sound source. Testing shows that this setup can improve the gain of the target sound source by up to 15dB and suppress ambient noise by more than 20dB within the 1000Hz-4000Hz frequency range, effectively improving the signal-to-noise ratio of sound acquisition and achieving high-fidelity capture of remote sound.

[0036] Example 2: Comparison and Optimization of Wireless Transmission Protocols This embodiment compares and tests Bluetooth 5.3 and a dedicated 2.4GHz wireless transmission protocol. In an environment with obstacles (such as walls) and a transmission distance of 10 meters, 48kHz sampling rate, 16-bit quantized audio signals were transmitted using both protocols. Test results show that Bluetooth 5.3 performs well in data transmission stability, with a packet loss rate of less than 1%, but the transmission latency is approximately 30ms. The dedicated 2.4GHz wireless transmission protocol reduces the transmission latency to 15ms and maintains high transmission rate and stability even in complex electromagnetic environments. Based on the test results, in scenarios with high latency requirements, such as real-time communication, the dedicated 2.4GHz wireless transmission protocol is recommended; in general everyday use scenarios, Bluetooth 5.3 can be used to ensure compatibility with more devices.

[0037] Example 3: Performance Test of Noise Reduction Algorithm In the noise reduction process of the intelligent processing module, a Wiener filtering-based noise reduction algorithm is adopted. A test environment containing various noise types (such as white noise, traffic noise, and human voice noise) is constructed, and the collected noisy audio signals are input into the system. The noise reduction effect is optimized by adjusting the parameters of the Wiener filtering algorithm, such as the noise power spectrum estimation window size and update rate. Experimental results show that the algorithm can effectively reduce noise under different noise intensities. In noisy speech signals with a signal-to-noise ratio of 5dB, the speech clarity is improved by 25% after noise reduction, with minimal damage to the speech signal, preserving the naturalness and clarity of the sound.

[0038] Example 4: Verification of Dynamic Range Compression Effect In this embodiment, the dynamic range compression effect is verified by setting different compression ratio coefficients k and compression curve exponents n. Sound signals of varying intensities are collected, including soft music, normal conversation, and loud ambient noise. , When loud sounds (e.g., 100dB) are compressed effectively to keep their output within a comfortable volume range (approximately 80dB); and soft sounds (e.g., 40dB) are appropriately amplified to approximately 50dB. After dynamic range compression, users can enjoy a clear and comfortable listening experience in various sound intensity environments, avoiding the problems of harsh loud sounds and inaudible soft sounds.

[0039] Example 5: Adaptive Equalization Adjustment and Personalized Settings By connecting a mobile app to the smart hearing aid system, a personalized equalization adjustment plan is generated based on the user's hearing test results. The system has built-in equalization templates for various hearing loss types, such as high-frequency hearing loss and low-frequency hearing loss. Users can select a template or manually adjust the gain of different frequency components according to their own situation. Through experiments, for users with high-frequency hearing loss, increasing the gain of the 8000Hz frequency component by 10dB while appropriately reducing the low-frequency gain significantly improved clarity when listening to high-frequency sounds (such as birdsong and telephone ringing), and improved speech comprehension ability by about 20%, achieving personalized adaptation of the hearing aid.

[0040] Example 6: System Waterproof and Dustproof Performance Test The waterproof and dustproof structure of the remote microphone module was tested, achieving an IP67 protection rating. The module was completely submerged in 1 meter of water for 30 minutes, and then its internal circuitry and microphone function were inspected. Test results showed no water ingress, and the microphone continued to pick up sound signals normally, with no performance impact. In a simulated sandstorm environment, the module was exposed to sand for 2 hours; sand did not penetrate the module and did not affect its normal operation. This waterproof and dustproof design enables the system to operate stably in various harsh environments, such as rainy weather and outdoor work, expanding the system's application range.

[0041] The foregoing has shown and described the basic principles, main features, and advantages of the present invention. Those skilled in the art should understand that the present invention is not limited to the above embodiments. The embodiments and descriptions in the specification are merely illustrative of the principles of the invention. Various changes and modifications can be made to the invention without departing from its spirit and scope, and all such changes and modifications fall within the scope of protection claimed by the present invention. The scope of protection of the present invention is defined by the appended claims and their equivalents.

Claims

1. An intelligent hearing aid system based on remote high-fidelity sound capture technology, characterized in that, include: The remote sound pickup module includes a microphone array consisting of multiple high-sensitivity microphones, used to collect sound signals from the target sound source and enhance the target sound source signal through beamforming algorithms while suppressing ambient noise. The wireless transmission module is used to transmit the sound signal collected by the remote pickup module to the intelligent processing module in a low-latency, high-fidelity manner; The intelligent processing module performs noise reduction, dynamic range compression, adaptive equalization adjustment, and deep learning-based speech enhancement on the received audio signal. The audio output module converts the sound signal processed by the intelligent processing module into audio output to the user; wherein, the beamforming algorithm calculates the weighting coefficients of each microphone signal based on the geometric position of the microphone array and the sound propagation characteristics, and the weighting coefficients are expressed by the formula... calculate, For the first The distance from each microphone to the target sound source For the first The time delay of each microphone relative to the reference microphone, The frequency of the sound.

2. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The remote sound pickup module employs MEMS microphones, with the microphone array using either a uniform circular array or a linear array layout. The uniform circular array offers excellent sound capture uniformity across a 360-degree horizontal range, making it suitable for scenarios requiring omnidirectional sound capture, such as conference discussions and multi-person communication environments. Taking a 6-microphone uniform circular array with an 8cm diameter as an example, by optimizing the microphone spacing and layout, precise beamforming of the target sound source can be achieved over a wide frequency range, with a sound source localization error of less than 5 degrees within a 5-meter radius. The linear array, on the other hand, offers high sound gain and noise suppression capabilities in specific directions, making it suitable for scenarios where the direction of the target sound source is clearly defined, such as lectures and classroom listening. Using a 4-microphone linear array with a 3cm microphone spacing, a gain improvement of over 18dB can be achieved in the direction of the target sound source, while simultaneously achieving a 22dB suppression effect on side noise. Furthermore, the linear array layout facilitates integration with the hearing aid's shell, reducing production difficulty.

3. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The wireless transmission module adopts Bluetooth 5.3 or a dedicated 2.4GHz wireless transmission protocol, with a transmission bandwidth not less than the transmission requirements of 16-bit audio at a sampling rate of 48kHz. Bluetooth 5.3 protocol has a mature technology ecosystem and has wide device compatibility, and can be seamlessly connected with smartphones, tablets and other mobile terminals, making it convenient for users to remotely control and adjust parameters of the hearing aid through an APP. It supports LEAudio technology, enabling low-power, high-quality audio transmission. During audio streaming, the audio latency can be controlled within 30ms, meeting the needs of general voice interaction. The dedicated 2.4GHz wireless transmission protocol is tailor-made for hearing aid systems. By optimizing channel coding and modulation methods, such as using orthogonal frequency division multiplexing technology, the transmission latency can be further reduced to 15ms. In complex electromagnetic environments, through a dynamic channel switching mechanism, it can effectively avoid interference frequency bands and ensure stable audio signal transmission. Even in densely populated areas with complex wireless signals, the packet loss rate can still be controlled below 0.5%.

4. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The intelligent processing module includes a digital signal processor. Noise reduction processing employs a Wiener filtering-based noise reduction algorithm, and dynamic range compression is achieved through the formula... accomplish, This is the compression ratio coefficient. The compression curve index; In practical applications, the Wiener filtering-based noise reduction algorithm adapts to constantly changing noise environments by updating the noise power spectrum estimate in real time. The system updates the noise power spectrum every 100ms, enabling rapid response and adjustment of noise reduction parameters in sudden noise scenarios. While maintaining the integrity of the speech signal, it improves noise suppression to over 25dB. For the dynamic range compression section, the system presets multiple k and n combination modes to meet the different hearing needs of users. For users with severe hearing loss, it employs... , The combination of these technologies has a more significant effect on enhancing weak sounds, raising the originally barely audible 40dB weak sound to about 60dB, while effectively compressing the 90dB strong sound, stabilizing its output at 75dB, and ensuring the user's auditory comfort.

5. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The audio output module is a bone conduction headphone or an in-ear speaker, and has an adaptive volume adjustment function; the bone conduction headphone transmits sound to the auditory nerve by vibrating the skull, without blocking the ear canal, which ensures clear sound transmission and allows users to perceive the sounds of the surrounding environment. Its built-in vibration unit uses high-performance neodymium iron boron magnets, with a sound pressure level change of less than 10dB in the 100-8000Hz frequency range, ensuring accurate sound reproduction; the in-ear speaker uses a customized balanced armature unit, which, through a precisely tuned crossover network, can achieve a flat response of ±3dB in the high-frequency range of 4000-8000Hz, providing users with clear and delicate sound; The adaptive volume adjustment function monitors the ambient noise intensity in real time through a built-in ambient noise sensor. When the ambient noise intensity exceeds 60dB, the system automatically increases the volume by 3-5dB; when the noise intensity is below 40dB, the volume decreases by 2-3dB, ensuring that users can have a comfortable listening experience in different environments.

6. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: It also includes a user interaction module for receiving user commands and adjusting the hearing aid's volume and noise reduction level parameters. The user interaction module uses a combination of physical buttons and touch control. The physical buttons are designed as raised buttons, making it easy for users to operate blindly without looking at the hearing aid. The buttons provide clear feedback and distinct tactile feedback. The touch area uses capacitive touch technology and supports various operation gestures such as swiping and clicking. For example, users can swipe up and down on the touch area of ​​the hearing aid shell to adjust the volume, and swipe left and right to switch noise reduction levels. In addition, the user interaction module also supports voice control, which can recognize user voice commands through a built-in voice recognition chip.

7. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The intelligent processing module incorporates a deep learning-based speech recognition model, capable of recognizing different scenarios and automatically switching sound processing modes. This deep learning-based speech recognition model employs an architecture combining convolutional neural networks and long short-term memory networks, trained with a large amount of speech data covering various scenarios such as conference rooms, shopping malls, streets, and homes. In practical applications, the system extracts features and identifies scenarios from the collected sound signals every second. When a conference room scenario is identified, it automatically switches to conference mode to enhance the speech signal and suppress environmental reverberation. When a shopping mall scenario is identified, a strong noise reduction mode is activated, focusing on suppressing noisy human voices and background noise while improving the clarity of the target speech.

8. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The system also includes a charging management module that uses wireless charging technology to power the system. The charging management module adopts the Qi wireless charging standard, compatible with most wireless chargers on the market. During charging, it features overvoltage, overcurrent, and overtemperature protection functions. When the charging voltage exceeds 5.5V, it automatically cuts off the charging circuit; when the charging current exceeds 500mA, it reduces the charging power; and when the charging temperature exceeds 50℃, it pauses charging and activates a cooling mechanism. In terms of charging efficiency, using a 10W wireless charger, the system battery can be charged from 0% to 80% in 2 hours and fully charged in 3 hours. Furthermore, the wireless charging design avoids the problems of easily damaged or poorly contacted traditional charging interfaces, improving the system's durability and ease of use.

9. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The remote microphone module has a waterproof and dustproof structure with a protection level of no less than IP67. The outer shell of the remote microphone module is made of high-strength polycarbonate material, combined with a precise sealing process. Silicone rubber sealing rings are used to seal the seams of the outer shell, and the microphone part is covered with a waterproof and breathable membrane, which can ensure that the sound can be transmitted normally while preventing water and dust from entering.

10. The intelligent hearing aid system based on remote high-fidelity sound capture technology according to claim 1, characterized in that: The system supports connection to mobile terminals via an app, enabling remote adjustment and personalized settings. The app interface features a simple and intuitive design, allowing users to view real-time information such as the hearing aid's battery level and connection status. For remote adjustment, users can upload their hearing test data to the system via the app, and the system automatically generates a personalized sound processing solution based on the data. Users can also manually adjust equalization parameters, noise reduction levels, and volume settings, with the adjusted parameters being synchronized to the hearing aid in real-time via a wireless transmission module. Furthermore, the app has a data statistics function, recording daily usage time and frequency of use in different scenarios, and generating usage reports to help users and professionals understand hearing aid usage and hearing change trends, providing a basis for further optimization of hearing solutions.

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