Method and device for optimizing hearing aid environment

Abstract

The invention provides a method and a device for optimizing a hearing aid environment, the method and the device are applied to a digital hearing aid, and the method comprises the following steps: acquiring sound information around the digital hearing aid through audio acquisition equipment arranged in an array, and carrying out differential filtering processing on the sound information to obtain a plurality of pieces of audio information; judging the position of a sound source according to the time difference of acquiring the same audio information by different audio acquisition devices; different audio information is amplified differentially according to the position of the sound source position relative to a preset communication range, the sound source position is acquired by means of an audio acquisition device array, in the area within the preset communication range, the sound loudness is increased along with the increase of the distance from the hearing aid, and in the area outside the preset communication range, the sound loudness is increased along with the increase of the distance from the hearing aid. The sound loudness is weakened along with the increase of the distance with the hearing aid, the influence of operation noise, wind noise and friction noise of the hearing aid is weakened to the maximum extent, the filtering burden of hearing aid is reduced, and the rigid gain design of a traditional hearing aid is abandoned.

Description

Technical Field

[0001] This invention patent relates to the field of digital hearing aid technology, specifically to a method and apparatus for optimizing the hearing aid environment. Background Technology

[0002] Hearing aids are amplification devices that help people with hearing loss improve their hearing and thus enhance their verbal communication abilities. There are many types of hearing aids. The first generation of hearing aids simply amplified audio information without discrimination to help hearing-impaired patients. In complex environments, the sound of these hearing aids is noisy, making it difficult for hearing-impaired patients to communicate normally with the people around them. The second generation of hearing aids, on the other hand, amplified the sound of nearby sounds differently by amplifying the sound of distant sounds, thereby increasing the loudness of nearby sounds and weakening the loudness of distant sounds, which greatly improves the hearing environment for hearing-impaired patients. Chinese Patent Publication No. CN105764774B discloses a method for generating audio signals using configurable distance cues. It includes the following steps: receiving an input audio signal, the input audio signal having at least one signal characteristic, the at least one signal characteristic being the hearing distance perception of the hearing aid user; receiving a distance control signal; And according to the distance control signal representing the perceived distance, at least one characteristic is changed. However, the above technology can only achieve a lower loudness for distant sound sources than for nearby sound sources. It will over-gain all sound sources that are close to itself. This results in a lot of noise such as operating noise, wind noise, and friction noise in the hearing aid. It also requires high filtering performance of the hearing aid. Furthermore, when the sound source is too close, it may even damage the hearing of the hearing-impaired patient due to excessive loudness. Therefore, the present invention provides a hearing aid method and device that can improve the comfort of hearing aids.

[0003] Invention Patent Content In view of the deficiencies in the prior art, this invention provides a method and apparatus for optimizing the hearing aid environment to solve the defects of hearing aids.

[0004] According to a first aspect of the present disclosure, a preferred embodiment of the present invention provides a method for optimizing a hearing-aiding environment, applied to a digital hearing aid, the method comprising: The sound information around the digital hearing aid is acquired by an array of audio acquisition devices, and the sound information is subjected to differential filtering to obtain multiple audio information. The location of the sound source is determined based on the time difference in the acquisition of the same audio information by different audio acquisition devices. Different audio information is amplified differently based on the position of the sound source relative to the preset communication range, wherein the preset communication range is a circular interval surrounding the audio acquisition device.

[0005] In one embodiment, determining the sound source location based on the time difference in the acquisition of the same audio information by different audio acquisition devices includes: The audio features within different audio information are labeled according to the acoustic model, and the timestamps of the labeled audio features are obtained. Calculate the timestamp difference when different audio acquisition devices acquire the same audio feature; The distance between the sound source corresponding to different audio features and the audio acquisition device array is calculated using the time difference positioning method, and this distance is used as the sound source location.

[0006] In one embodiment, different audio information is amplified differentially based on the distance between the sound source location and a preset communication range, characterized by including: Based on the position of the sound source relative to the preset communication range, establish a functional relationship between the difference between the sound source position and the preset communication range and the audio amplification factor; Substitute the sound source location into the inverse proportional function between the difference between the sound source location and the preset AC range and the audio information amplification factor to obtain the required audio amplification factor; The audio information corresponding to the sound source location is amplified and output according to the audio amplification factor.

[0007] In one embodiment, in establishing the functional relationship between the sound source location and the preset AC range and the audio amplification factor, if the sound source location is outside the preset AC range, then the difference between the sound source location and the preset AC range and the audio amplification factor are inversely proportional; if the sound source location is within the preset AC range, then the difference between the sound source location and the preset AC range and the audio amplification factor are directly proportional.

[0008] According to a second aspect of the present disclosure, this invention provides an apparatus for optimizing the hearing aid environment, applied to a digital hearing aid, the apparatus comprising: The acquisition module is used to acquire sound information around the digital hearing aid through an array of audio acquisition devices, and to perform differential filtering on the sound information to obtain multiple audio information. The calculation module is used to determine the location of the sound source based on the time difference in the acquisition of the same audio information by different audio acquisition devices; The output module is used to amplify different audio information based on the position of the sound source relative to a preset communication range, wherein the preset communication range is a circular interval surrounding the audio acquisition device.

[0009] In one embodiment, the computing module includes: The identification module is used to mark the audio features in different audio information according to the acoustic model, and to obtain the timestamp of the marked audio features; Calculation submodule one is used to calculate the timestamp difference of the same audio feature acquired by different audio acquisition devices; The second calculation submodule is used to calculate the distance between the sound source corresponding to different audio features and the audio acquisition device array according to the time difference positioning method, and the distance is used as the sound source position.

[0010] In one embodiment, the output module includes: Module 1 is used to establish a functional relationship between the difference between the sound source location and the preset communication range and the audio amplification factor, based on the position of the sound source location relative to the preset communication range. Module 2 is used to substitute the sound source location into a function between the sound source location and the difference between the preset AC range and the audio information amplification factor to obtain the required audio amplification factor; The adjustment module is used to amplify and output the audio information corresponding to the sound source location according to the audio amplification factor.

[0011] In one embodiment, if the sound source location is outside the preset AC range in the determination module one, then the difference between the sound source location and the preset AC range has an inverse proportional function relationship with the audio amplification factor; if the sound source location is within the preset AC range, then the difference between the sound source location and the preset AC range has a direct proportional function relationship with the audio amplification factor.

[0012] According to a third aspect of the present disclosure, the present invention provides an apparatus for optimizing the hearing aid environment, comprising: processor; Memory used to store the processor's executable instructions; The processor is configured to perform the steps of the above method.

[0013] According to a fourth aspect of the present disclosure, the present invention provides a computer-readable storage medium having a computer program stored thereon, the computer program being executed by a processor of the steps of the above-described method.

[0014] As can be seen from the above technical solution, the method and device for optimizing the hearing aid environment provided by this invention patent obtains the sound source location by means of an audio acquisition device array. Within the preset communication range, the sound loudness increases with the increase of distance from the hearing aid, while outside the preset communication range, the sound loudness weakens with the increase of distance from the hearing aid. The influence of operating noise, wind noise, and friction noise of the hearing aid is reduced to the greatest extent, reducing the filtering burden of the hearing aid. It abandons the rigid gain design of traditional hearing aids and replaces the physical structure of the ear with digital technology gain. The preset communication range design conforms to communication habits, so that the hearing aid gain is maximized for close-range communication and communication barriers are minimized.

[0015] It should be understood that the above general description and the following detailed description are merely exemplary and do not limit this disclosure. Attached Figure Description

[0016] To more clearly illustrate the specific embodiments of this invention, the accompanying drawings used in the description of the specific embodiments or prior art will be briefly introduced below. In all the drawings, the elements or parts are not necessarily drawn to scale.

[0017] Figure 1 A flowchart of a method for optimizing the hearing aid environment provided by this invention patent; Figure 2 A flowchart of step S200 in a method for optimizing a hearing aid environment provided by this invention patent; Figure 3 The process of step S300 in the method for optimizing the hearing aid environment provided by this invention patent Figure 1 ; Figure 4 A block diagram of a device for optimizing the hearing aid environment provided for this invention patent; Figure 5 A block diagram of another device for optimizing the hearing aid environment provided by this invention. Detailed Implementation

[0018] The embodiments of the technical solution of this invention will now be described in detail with reference to the accompanying drawings. These embodiments are merely illustrative of the technical solution of this invention and are therefore intended to limit the scope of protection of this invention.

[0019] Figure 1This invention provides a flowchart of a method for optimizing the hearing-aided environment. This method is applied to a digital hearing aid terminal, which can display images, videos, text messages, WeChat messages, and other information. The terminal can be equipped with any terminal device with a display screen, such as a mobile phone, computer, digital broadcasting terminal, messaging device, game console, tablet, medical device, fitness equipment, or personal digital assistant. This embodiment provides a method for optimizing the hearing-aided environment applied to a digital hearing aid, such as... Figure 1 As shown, the method includes the following steps S10-S30: In step S10, sound information around the digital hearing aid is acquired by an array of audio acquisition devices, and the sound information is subjected to differential filtering to obtain multiple audio information. In this implementation, firstly, sound information around the hearing aid user is collected by an array of audio acquisition devices. These devices typically include multiple microphones that can capture sound from different directions. The collected sound information is then subjected to differential filtering. The purpose of this step is to decompose the acquired composite sound signal into multiple independent audio information for subsequent processing and analysis. Differential filtering involves signal processing techniques such as Fourier transform to convert the time-domain signal into a frequency-domain signal, thereby better identifying and processing the frequency components of the sound.

[0020] In step S20, the location of the sound source is determined based on the time difference in the acquisition of the same audio information by different audio acquisition devices; In this implementation, by analyzing the timestamp differences of the same audio feature obtained by different audio acquisition devices, the system can calculate the distance of the sound source relative to the audio acquisition device array. This method is usually called time difference localization, which uses the time difference of sound arrival at each microphone to determine the location of the sound source.

[0021] In step S30, different audio information is amplified differently according to the position of the sound source relative to the preset communication range, wherein the preset communication range is a circular interval surrounding the audio acquisition device; In this implementation, within the preset communication range, the sound loudness increases with the distance from the hearing aid, while outside the preset communication range, the sound loudness decreases with the distance from the hearing aid. The effects of the hearing aid's operating noise, wind noise, and friction noise are minimized, reducing the filtering burden on the hearing aid and providing the user with a more comfortable and natural auditory environment.

[0022] Among them, such as Figure 2 As shown, in step S20, the location of the sound source is determined based on the time difference in the acquisition of the same audio information by different audio acquisition devices, including the following steps S21-S23: In step S21, audio features within different audio information are marked according to the acoustic model, and the timestamps of the marked audio features are obtained. In this implementation, based on the acoustic model, the system identifies key audio features in the audio information. These features include the frequency components, amplitude, and duration of the sound, which are the basis for sound recognition and processing. In step S22, the timestamp difference between different audio acquisition devices acquiring the same audio feature is calculated; In this implementation, once an audio feature is identified, the system will mark it. This marking process involves assigning a unique identifier or classification label to the audio feature to facilitate subsequent processing and analysis. For each marked audio feature, the system will record its timestamp. The timestamp refers to the specific time point when the audio feature appears in the audio signal. This is crucial for subsequent calculation of the sound source location and time difference localization. The acquired timestamp information will be stored so that the calculation module can further analyze the data and calculate the position of the sound source relative to the hearing aid. In step S23, the distance between the sound source corresponding to different audio features and the audio acquisition device array is calculated according to the time difference positioning method, and this distance is used as the sound source location; In this implementation, time-difference localization is used to calculate the actual distance from the sound source to each microphone. This typically involves calculating the speed of sound (usually around 343 m / s in air) and the time difference, and the specific formula can be expressed as: Where d is the distance from the sound source to the microphone, c is the speed of sound, and Δt is the time difference between the sound reaching the two microphones. Once the distance from the sound source to each microphone is calculated, triangulation or other positioning algorithms can be used to determine the exact location of the sound source in space. This involves solving a system of multivariate equations, where each equation is based on the distance information of a microphone.

[0023] In one embodiment, such as Figure 3 As shown, in step S30, different audio information is amplified differently based on the distance between the sound source location and the preset communication range, including the following steps S31-S33: In step S31, a functional relationship between the difference between the sound source location and the preset AC range and the audio amplification factor is established based on the position of the sound source location relative to the preset AC range. Specifically, the circular interval surrounding the audio acquisition device is set according to the daily communication habits and needs of hearing aid users. After obtaining the distance of the sound source relative to the audio acquisition device array, the distance difference between the sound source position and the preset communication range boundary is calculated. This difference represents the distance of the sound source from the communication range. Then, a functional relationship is established between the difference between the sound source position and the preset communication range and the audio amplification factor.

[0024] In step S32, the sound source location is substituted into the inverse proportional function between the sound source location and the difference between the preset AC range and the audio information amplification factor to obtain the required audio amplification factor; In this implementation, taking an inverse proportional function as an example, it describes the relationship between the sound source location and the amplification factor, which can be expressed as: G(x) = k / x, where: G(x) is the audio amplification factor, x is the difference between the sound source location and the preset AC range, and K is a constant that depends on the design of the hearing aid and the user's hearing needs. In this function, x represents the distance difference between the sound source and the boundary of the hearing aid's preset AC range. If the sound source is outside the preset AC range, then x is a positive value, representing the distance between the sound source and the AC range boundary; if the sound source is within the preset AC range, then x can be considered a negative value. The value of x can be zero or zero, depending on the specific function definition and the actual application scenario. When we know the specific location of the sound source relative to the preset communication range, we can calculate the value of x and substitute it into the inverse proportional function mentioned above to calculate G(x), which is the required audio amplification factor. For example, if the sound source is located 10 meters away from the preset communication range, and the preset communication range of the hearing aid is 5 meters, then x can be 5 meters (i.e., 10 meters - 5 meters). If we set the value of K (this value needs to be determined based on the performance of the hearing aid and the user's hearing test results), we can calculate the specific value of G(x).

[0025] In step S33, the audio information corresponding to the sound source location is amplified and output according to the audio amplification factor; In this implementation, the calculated amplification factor G(x) is applied to the audio signal of the target sound source. This step can be achieved through digital signal processing (DSP) technology, where the amplitude of the audio signal is multiplied by the amplification factor G(x).

[0026] In one embodiment, in establishing the functional relationship between the sound source location and the preset AC range and the audio amplification factor, if the sound source location is outside the preset AC range, then the difference between the sound source location and the preset AC range and the audio amplification factor are inversely proportional; if the sound source location is within the preset AC range, then the difference between the sound source location and the preset AC range and the audio amplification factor are directly proportional. In this implementation, the functional relationship can be linear or nonlinear, depending on the design requirements and the user's hearing needs. The linear function can be a direct or inverse proportional function, or a straight line with different slopes, used to describe the relationship between the sound source location and the amplification factor within a certain range. When the sound source is within a preset AC range, a direct proportional function can be used to describe the relationship between the sound source location and the amplification factor. The general form of the direct proportional function is y=kx, where y represents the audio amplification factor, x represents the distance difference between the sound source and the AC range boundary, and k is a proportionality constant. In this case, as the sound source location approaches the user (i.e., the x value decreases), the amplification factor increases. The amplification factor y increases, making nearby sounds clearer. When the sound source is outside the preset AC range, an inverse proportional function can be used to describe the relationship between the sound source location and the amplification factor. The general form of the inverse proportional function is y=k / x, where y represents the audio amplification factor, x represents the distance difference between the sound source and the AC range boundary, and k is a proportionality constant. In this case, as the sound source moves further away from the user (i.e., the x value increases), the amplification factor y decreases, thereby reducing the loudness of distant sounds and reducing noise. Nonlinear functions can be more complex, including polynomial, exponential, or logarithmic functions, to describe more complex relationships between the sound source location and the amplification factor.

[0027] The following are embodiments of the apparatus disclosed herein, which can be used to execute embodiments of the method disclosed herein.

[0028] Figure 4 This invention provides a block diagram of a device for optimizing the hearing aid environment. This device can be implemented as part or all of an electronic device through software, hardware, or a combination of both. Figure 5 As shown, the device, applied to a digital hearing aid, includes: The acquisition module 100 is used to acquire sound information around the digital hearing aid through an array of audio acquisition devices, and to perform differential filtering on the sound information to obtain multiple audio information. The calculation module 200 is used to determine the location of the sound source based on the time difference in the acquisition of the same audio information by different audio acquisition devices; The output module 300 is used to amplify different audio information differentially according to the position of the sound source relative to a preset communication range, wherein the preset communication range is a circular interval surrounding the audio acquisition device.

[0029] When connected to Bluetooth mode, this invention can coordinate Bluetooth audio and ambient audio based on the distance between the hearing aid and the Bluetooth-connected mobile device. As the mobile device approaches the hearing aid, the Bluetooth audio gradually increases while the ambient audio gradually decreases until the hearing-impaired patient can enjoy a dedicated hearing environment in noisy environments. As the mobile device moves away from the hearing aid, the Bluetooth audio gradually decreases while the ambient audio gradually increases. This helps the patient respond to surrounding sounds in a timely manner, ensuring safe listening and is more in line with the human ear mechanism. It is suitable for Bluetooth-connected devices such as smartphones, TVs, and in-vehicle systems.

[0030] In one embodiment, such as Figure 4 As shown, the computing module 200 includes: The identification module 210 is used to mark the audio features in different audio information according to the acoustic model, and to obtain the timestamp of the marked audio features; Calculation submodule 220 is used to calculate the timestamp difference of the same audio feature acquired by different audio acquisition devices; The calculation submodule 230 is used to calculate the distance of the sound source corresponding to different audio features relative to the audio acquisition device array according to the time difference positioning method, and the distance is used as the sound source position.

[0031] In one embodiment, such as Figure 4 As shown, the output module 300 includes: Module 310 is used to establish a functional relationship between the difference between the sound source location and the preset AC range and the audio amplification factor based on the position of the sound source location relative to the preset AC range. Module 2 320 is used to substitute the sound source position into the function between the sound source position and the difference between the preset AC range and the audio information amplification factor to obtain the required audio amplification factor; The adjustment module 330 is used to amplify and output the audio information corresponding to the sound source location according to the audio amplification factor.

[0032] In one embodiment, if the sound source location is outside the preset AC range in the determination module one, then the difference between the sound source location and the preset AC range has an inverse proportional function relationship with the audio amplification factor; if the sound source location is within the preset AC range, then the difference between the sound source location and the preset AC range has a direct proportional function relationship with the audio amplification factor.

[0033] Regarding the apparatus in the above embodiments, the specific manner in which each module performs its operation has been described in detail in the embodiments related to the method, and will not be elaborated upon here.

[0034] This disclosure also provides another device for optimizing the hearing aid environment: Figure 5 This is a block diagram illustrating an apparatus 800 for optimizing the hearing aid environment according to an exemplary embodiment. For example, apparatus 800 may be a mobile phone, computer, digital broadcasting terminal, messaging device, game console, tablet device, medical device, fitness equipment, personal digital assistant, etc.

[0035] Reference Figure 5 The device 800 may include one or more of the following components: a processing component 802, a memory 804, a power supply component 806, a multimedia component 808, an audio component 810, an input / output (I / O) interface 812, a sensor component 814, and a communication component 816.

[0036] Processing component 802 typically controls the overall operation of device 800, such as operations associated with display, telephone calls, data communication, camera operation, and recording. Processing component 802 may include one or more processors 820 to execute instructions to perform all or part of the steps of the methods described above. Furthermore, processing component 802 may include one or more modules to facilitate interaction between processing component 802 and other components. For example, processing component 802 may include a multimedia module to facilitate interaction between multimedia component 808 and processing component 802.

[0037] Memory 804 is configured to store various types of data to support the operation of device 800. Examples of such data include instructions for any application or method operating on device 800, contact data, phonebook data, messages, pictures, videos, etc. Memory 804 can be implemented by any type of volatile or non-volatile storage device or a combination thereof, such as static random access memory (SRAM), electrically erasable programmable read-only memory (EEPROM), erasable programmable read-only memory (EPROM), programmable read-only memory (PROM), read-only memory (ROM), magnetic storage, flash memory, magnetic disk, or optical disk.

[0038] Power supply component 806 provides power to various components of device 800. Power supply component 806 may include a power management system, one or more power sources, and other components associated with generating, managing, and distributing power to device 800.

[0039] Multimedia component 808 includes a screen that provides an output interface between the device 800 and the user. In some embodiments, the screen may include a liquid crystal display (LCD) and a touch panel (TP). If the screen includes a touch panel, the screen may be implemented as a touchscreen to receive input signals from the user. The touch panel includes one or more touch sensors to sense touches, swipes, and gestures on the touch panel. The touch sensors may sense not only the boundaries of the touch or swipe action but also the duration and pressure associated with the touch or swipe operation. In some embodiments, multimedia component 808 includes a front-facing camera and / or a rear-facing camera. When the device 800 is in an operating mode, such as a shooting mode or a video mode, the front-facing camera and / or the rear-facing camera may receive external multimedia data. Each front-facing camera and rear-facing camera may be a fixed optical lens system or have focal length and optical zoom capabilities.

[0040] Audio component 810 is configured to output and / or input audio signals. For example, audio component 810 includes a microphone (MIC) configured to receive external audio signals when device 800 is in an operating mode, such as call mode, recording mode, and voice recognition mode. The received audio signals may be further stored in memory 804 or transmitted via communication component 816. In some embodiments, audio component 810 includes a speaker for outputting audio signals.

[0041] I / O interface 812 provides an interface between processing component 802 and peripheral interface modules, such as keyboards, click wheels, buttons, etc. These buttons may include, but are not limited to, home buttons, volume buttons, power buttons, and lock buttons.

[0042] Sensor assembly 814 includes one or more sensors for providing status assessments of various aspects of device 800. For example, sensor assembly 814 may detect the on / off state of device 800, the relative positioning of components such as the display and keypad of device 800, changes in the position of device 800 or a component of device 800, the presence or absence of user contact with device 800, the orientation or acceleration / deceleration of device 800, and temperature changes of device 800. Sensor assembly 814 may include a proximity sensor configured to detect the presence of nearby objects without any physical contact. Sensor assembly 814 may also include a light sensor, such as a CMOS or CCD image sensor, for use in imaging applications. In some embodiments, sensor assembly 814 may also include an accelerometer, a gyroscope, a magnetometer, a pressure sensor, or a temperature sensor.

[0043] The communication component 816 is configured to facilitate wired or wireless communication between the device 800 and other devices. The device 800 can access wireless networks based on communication standards, such as WiFi, 2G or 3G, or combinations thereof.

[0044] In one exemplary embodiment, the communication component 816 receives broadcast signals or broadcast-related information from an external broadcast management system via a broadcast channel. In one exemplary embodiment, the communication component 816 includes a near-field communication (NFC) module to facilitate short-range communication. For example, the NFC module may be implemented based on radio frequency identification (RFID) technology, Infrared Data Association (IrDA) technology, ultra-wideband (UWB) technology, Bluetooth (BT) technology, and other technologies.

[0045] In an exemplary embodiment, the apparatus 800 may be implemented by one or more application-specific integrated circuits (ASICs), digital signal processors (DSPs), digital signal processing devices (DSPDs), programmable logic devices (PLDs), field-programmable gate arrays (FPGAs), controllers, microcontrollers, microprocessors, or other electronic components to perform the methods described above.

[0046] In an exemplary embodiment, a non-transitory computer-readable storage medium including instructions is also provided, such as a memory 804 including instructions, which can be executed by a processor 820 of the device 800 to perform the above-described method. For example, the non-transitory computer-readable storage medium may be a ROM, random access memory (RAM), CD-ROM, magnetic tape, floppy disk, and optical data storage device, etc.

[0047] Other embodiments of this disclosure will readily occur to those skilled in the art upon consideration of the specification and practice of the invention disclosed herein. This application is intended to cover any variations, uses, or adaptations of this disclosure that follow the general principles of this disclosure and include common knowledge or customary techniques in the art not disclosed herein. The specification and examples are to be considered exemplary only, and the true scope and spirit of this disclosure are indicated by the following claims.

[0048] It should be understood that this disclosure is not limited to the precise structures described above and shown in the accompanying drawings, and various modifications and changes can be made without departing from its scope. The scope of this disclosure is limited only by the appended claims.

Claims

1. A method for optimizing the hearing aid environment, applied to a digital hearing aid, characterized in that, The method includes: The sound information around the digital hearing aid is acquired by an array of audio acquisition devices, and the sound information is subjected to differential filtering to obtain multiple audio information. The location of the sound source is determined based on the time difference in the acquisition of the same audio information by different audio acquisition devices. Different audio information is amplified differently based on the position of the sound source relative to the preset communication range, wherein the preset communication range is a circular interval surrounding the audio acquisition device.

2. The method according to claim 1, characterized in that, Determining the location of a sound source based on the time difference in how the same audio information is acquired by different audio acquisition devices includes: The audio features within different audio information are labeled according to the acoustic model, and the timestamps of the labeled audio features are obtained. Calculate the timestamp difference when different audio acquisition devices acquire the same audio feature; The distance between the sound source corresponding to different audio features and the audio acquisition device array is calculated using the time difference positioning method, and this distance is used as the sound source location.

3. The method according to claim 1, characterized in that, The method amplifies different audio information based on the distance between the sound source location and the preset communication range, characterized by including: Based on the position of the sound source relative to the preset communication range, establish a functional relationship between the difference between the sound source position and the preset communication range and the audio amplification factor; Substitute the sound source location into the inverse proportional function between the difference between the sound source location and the preset AC range and the audio information amplification factor to obtain the required audio amplification factor; The audio information corresponding to the sound source location is amplified and output according to the audio amplification factor.

4. The method according to claim 3, characterized in that, In establishing the functional relationship between the sound source location and the preset AC range and the audio amplification factor, if the sound source location is outside the preset AC range, then the difference between the sound source location and the preset AC range and the audio amplification factor are inversely proportional; if the sound source location is within the preset AC range, then the difference between the sound source location and the preset AC range and the audio amplification factor are directly proportional.

5. A device for optimizing the hearing aid environment, applied to a digital hearing aid, characterized in that, The method includes: The acquisition module is used to acquire sound information around the digital hearing aid through an array of audio acquisition devices, and to perform differential filtering on the sound information to obtain multiple audio information. The calculation module is used to determine the location of the sound source based on the time difference in the acquisition of the same audio information by different audio acquisition devices; The output module is used to amplify different audio information based on the position of the sound source relative to a preset communication range, wherein the preset communication range is a circular interval surrounding the audio acquisition device.

6. The apparatus according to claim 5, characterized in that, The computing module includes: The identification module is used to mark the audio features in different audio information according to the acoustic model, and to obtain the timestamp of the marked audio features; Calculation submodule one is used to calculate the timestamp difference of the same audio feature acquired by different audio acquisition devices; The second calculation submodule is used to calculate the distance between the sound source corresponding to different audio features and the audio acquisition device array according to the time difference positioning method, and the distance is used as the sound source position.

7. The apparatus according to claim 5, characterized in that, The output module includes: Module 1 is used to establish a functional relationship between the difference between the sound source location and the preset communication range and the audio amplification factor, based on the position of the sound source location relative to the preset communication range. Module 2 is used to substitute the sound source location into a function between the sound source location and the difference between the preset AC range and the audio information amplification factor to obtain the required audio amplification factor; The adjustment module is used to amplify and output the audio information corresponding to the sound source location according to the audio amplification factor.

8. The apparatus according to claim 7, characterized in that, In the determination module one, if the sound source location is outside the preset AC range, the difference between the sound source location and the preset AC range has an inverse proportional function relationship with the audio amplification factor; if the sound source location is within the preset AC range, the difference between the sound source location and the preset AC range has a direct proportional function relationship with the audio amplification factor.

9. A device for optimizing the hearing aid environment, characterized in that, include: processor; Memory used to store the processor's executable instructions; The processor is configured to perform the steps of the method of any one of claims 1 to 4.

10. A computer-readable storage medium having a computer program stored thereon, characterized in that, When the computer program is executed by the processor, it implements the steps of any one of claims 1 to 4.

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