Ambient sound gain method and device based on two-dimensional sound source localization technology

Abstract

The invention provides an environmental sound gain method based on a two-dimensional sound source positioning technology, which is applied to a digital hearing aid, and comprises the following steps: calculating sound information collected by the digital hearing aid based on the two-dimensional sound source positioning technology to obtain azimuth information of sound sources of different types of information; caching the azimuth information of the sound sources of different types of information, and judging the motion state of the sound sources relative to the digital hearing aid according to the change of the azimuth information of the sound sources of the same type of information; if the sound source is in a moving state relative to the digital hearing aid, performing dynamic high-gain-intensity amplification processing on a sound signal corresponding to the sound source; through independent processing of the environment sound, the situation that important voice information is covered by background sound is effectively avoided, differential amplification can be performed on a static sound source and a mobile sound source according to different motion states of the sound source relative to the digital hearing aid, and the identification degree of the mobile sound source can be improved.

Description

Technical Field

[0001] This invention patent relates to the field of digital hearing aid technology, specifically to an environmental sound gain method and device based on two-dimensional sound source localization technology. Background Technology

[0002] Sound source localization locates objects by measuring the sound they emit. Unlike localization methods that use sonar, radar, or wireless communication, the former uses ordinary sound as the source, which is a broadband signal, while the latter uses a narrowband signal. Based on the characteristics of sound signals, different sound source localization algorithms have been proposed. However, due to the presence of signal quality, noise, and reverberation, the localization accuracy of existing sound source localization algorithms is relatively low. Sound source localization technology based on sound sensors can effectively perform spatial filtering and noise reduction on the picked-up noisy signals, which can improve the performance of sound source localization systems to a certain extent. Chinese Patent Publication No. CN109286790B discloses a directional monitoring system and method based on sound source localization. It includes a monitoring device deployed at least once within the monitoring range, a signal processing device communicatively connected to the monitoring device, and a PC pre-loaded with control and recording software. The control and recording software communicates with the signal processing device via the PC. The monitoring device consists of a multi-channel microphone array. The signal processing device adjusts the phase of the received sound signal from the monitoring device based on a beamforming theory-based sound source localization algorithm, resulting in strong directivity. This ensures that the sound from a specified direction is stronger while maintaining sound quality, and it suppresses interference signals from unrelated angles to a certain extent, providing strong anti-interference capabilities and improving the quality of the monitored sound. Furthermore, it allows for wider coverage with fewer monitoring points and active monitoring. However, the above technology only achieves directional anti-interference. When the sound environment of a hearing-impaired patient is complex, it can hinder the patient's perception of their surroundings. Therefore, this invention provides an environmental sound gain method and device based on two-dimensional sound source localization technology to solve the above problems.

[0003] Invention Patent Content To address the shortcomings of existing technologies, this invention provides an environmental sound gain method and device based on two-dimensional sound source localization technology, in order to solve the deficiencies of digital hearing aids in directional hearing aid technology.

[0004] According to a first aspect of the present disclosure, a preferred embodiment of the present invention provides an environmental sound gain method based on two-dimensional sound source localization technology, applied to a digital hearing aid, the method comprising: The sound information collected by the digital hearing aid is calculated and processed based on two-dimensional sound source localization technology to obtain the location information of the sound source of different types of information. The location information of the sound source with different types of information is cached, and the motion state of the sound source relative to the digital hearing aid is determined based on the change in the location information of the sound source with the same type of information. If the sound source is moving relative to the digital hearing aid, the sound signal corresponding to the sound source undergoes dynamic high-gain amplification; and If the sound source is stationary relative to the digital hearing aid, the sound signal corresponding to the sound source is amplified at a low gain.

[0005] In one embodiment, the sound information collected by the digital hearing aid is processed using two-dimensional sound source localization technology to obtain the directional information of the sound source of different types, including: The sound information around the digital hearing aid is acquired by an array of audio acquisition devices, and the type information of all sound sources contained in the sound information is identified according to an acoustic model. Calculate the time difference between different audio acquisition devices acquiring the same type of information from sound sources, and calculate the orientation information of the sound sources of different types of information relative to the origin of the coordinate system with the audio acquisition device array as the time difference positioning method.

[0006] In one embodiment, if the sound source is moving relative to the digital hearing aid, the sound signal corresponding to the sound source performs a high gain; if the sound source is stationary relative to the digital hearing aid, the sound signal corresponding to the sound source performs a low gain, including: Calculate the displacement of the sound source of the same type of information relative to the origin of the coordinate system with the digital hearing aid as the origin; If the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is positive, obtain the direct proportional function relationship between the average value of the absolute value of the displacement per unit time and the gain intensity. If the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is negative, obtain the inverse proportional function relationship between the average value of the absolute value of the displacement per unit time and the gain intensity. The gain intensity is calculated based on the functional relationship and the average value of the absolute value of the displacement per unit time, and the acoustic information of the sound source is amplified with the corresponding gain intensity.

[0007] In one embodiment, the sound information collected by the digital hearing aid is processed based on two-dimensional sound source localization technology to obtain the sound information in the location information of the sound source of different types of information as the filtered ambient sound, which does not contain speech sound signals and noise signals.

[0008] According to a second aspect of the present disclosure, the present invention provides an environmental sound gain device based on two-dimensional sound source localization technology, applied to a digital hearing aid, the device comprising: The sound source localization module is used to calculate and process the sound information collected by the digital hearing aid based on two-dimensional sound source localization technology to obtain the directional information of the sound source of different types of information. The judgment module is used to cache the location information of the sound source of different types of information, and determine the motion state of the sound source relative to the digital hearing aid based on the change of the location information of the sound source of the same type of information. A dynamic gain module is used to perform dynamic high-gain amplification of the sound signal corresponding to the sound source if the sound source is moving relative to the digital hearing aid; and A static gain module is used to perform low-gain amplification of the sound signal corresponding to the sound source if the sound source is stationary relative to the digital hearing aid.

[0009] In one embodiment, the sound source localization module includes: The identification module is used to acquire sound information around the digital hearing aid through an array of audio acquisition devices, and to identify the type information of all sound sources contained in the sound information according to an acoustic model. The positioning submodule is used to calculate the time difference between different audio acquisition devices acquiring the same type of information from sound sources, and to calculate the orientation information of the sound sources of different types of information relative to the origin of the coordinate system of the audio acquisition device array based on the time difference positioning method.

[0010] In one embodiment, the dynamic gain module includes: The measurement module is used to calculate the displacement of the sound source of the same type of information relative to the origin of the coordinate system with the digital hearing aid as the origin; The first acquisition module is used to acquire the positive proportional function relationship between the average value of the absolute value of the displacement and the gain intensity if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is positive. The second acquisition module is used to acquire the inverse proportional function relationship between the absolute value of the displacement and the gain intensity if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is negative. The output module is used to calculate the gain intensity based on the functional relationship and the average value of the absolute value of the displacement over a unit time, and to amplify the acoustic information of the sound source with the corresponding gain intensity.

[0011] In one embodiment, the acoustic information in the sound source localization module is filtered ambient sound, which does not contain speech signals or noise signals.

[0012] According to a third aspect of the present disclosure, the present invention provides an environmental sound gain device based on two-dimensional sound source localization technology, 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 environmental sound gain method and device based on two-dimensional sound source localization technology provided by this invention patent can effectively avoid the situation where background sound masks important speech information by processing environmental sound separately. According to the different motion states of the sound source relative to the digital hearing aid, it can differentiate the amplification of stationary sound sources and moving sound sources, which can improve the recognition of moving sound sources. Outdoors, it can help hearing-impaired patients hear sounds such as vehicle driving and birdsong, improve environmental perception and safety. Moreover, by monitoring the movement path of the sound source, it can suppress the loudness changes of sound sources that are rapidly approaching or moving away, which can prevent the impact caused by excessively rapid loudness changes and prevent distortion or distortion from affecting the auditory experience.

[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 an environmental sound gain method based on two-dimensional sound source localization technology provided for this invention patent; Figure 2 A flowchart illustrating step S100 in an environmental sound gain method based on two-dimensional sound source localization technology provided for this invention patent; Figure 3 A flowchart illustrating step S300 in an environmental sound gain method based on two-dimensional sound source localization technology provided for this invention patent; Figure 4 A block diagram of an environmental sound gain device based on two-dimensional sound source localization technology provided for this invention patent; Figure 5A block diagram of another ambient sound gain device based on two-dimensional sound source localization technology provided for this invention patent. 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 1 This invention provides a flowchart of an environmental sound gain method based on two-dimensional sound source localization technology. This method is applied to a digital hearing aid terminal, which can display images, videos, text messages, WeChat messages, etc. 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 device, medical device, fitness equipment, or personal digital assistant. This embodiment provides an environmental sound gain method based on two-dimensional sound source localization technology applied to a digital hearing aid, such as... Figure 1 As shown, the method includes the following steps S10-S40: In step S10, the sound information collected by the digital hearing aid is processed based on two-dimensional sound source localization technology to obtain the directional information of the sound source of different types of information. In this implementation, two-dimensional sound source localization technology is a technique used to determine the position of a sound source on a two-dimensional plane. The general principle is as follows: first, two or more microphones are used to form an array, then the time difference of the sound reaching each microphone is measured, and finally the approximate position of the sound source is calculated using the speed of sound and the time difference.

[0020] In step S20, the location information of the sound source with different types of information is cached, and the motion state of the sound source relative to the digital hearing aid is determined according to the change of the location information of the sound source with the same type of information. In this implementation, the location information of the sound source is continuously monitored and updated. Then, the newly acquired location information is compared with the location information in the cache. If the location information of the sound source of the same type changes, it means that the position of the sound source relative to the digital hearing aid has changed, that is, the sound source is in a moving state. If the location information of the sound source does not change, it means that the position of the sound source relative to the digital hearing aid remains unchanged, that is, the sound source is in a stationary state. It is worth noting that if the hearing-impaired patient is stationary, any stationary sound-producing object is a stationary sound source, and any moving sound-producing object is a moving sound source. If the hearing-impaired patient is moving, then any sound-producing object stationary relative to the hearing-impaired patient is a stationary sound source, and any sound-producing object moving relative to the hearing-impaired patient is a moving sound source. In driving scenarios, this can significantly improve safety.

[0021] In step S30, if the sound source is in a moving state relative to the digital hearing aid, the sound signal corresponding to the sound source undergoes dynamic high gain intensity amplification processing.

[0022] In step S40, if the sound source is stationary relative to the digital hearing aid, the sound signal corresponding to the sound source is amplified with low gain intensity. In this implementation, high-gain amplification helps users track and identify sounds from moving sound sources, while low-gain amplification helps maintain the naturalness and comfort of the sound from stationary sound sources. This intelligent sound processing strategy can significantly improve the performance of hearing aids, especially in complex and dynamic sound environments. It is worth noting that the above-mentioned gain intensity is intended to achieve the amplification factor of the sound signal corresponding to different sound sources. The specific parameters need to be determined according to the degree of hearing impairment, which will not be elaborated in this application.

[0023] Among them, such as Figure 2 As shown, in step S10, the sound information collected by the digital hearing aid is processed based on two-dimensional sound source localization technology to obtain the directional information of the sound source of different types of information, including the following steps S11-S12: In step S11, sound information around the digital hearing aid is acquired by an array of audio acquisition devices, and the type information of all sound sources contained in the sound information is identified according to an acoustic model. In this implementation, the acoustic model contains audio features corresponding to sound sources of different types of information. By identifying specific audio features in the sound information, the type information of the sound source can be distinguished. Furthermore, by using a deep learning network to analyze the features of the sound source, the accuracy of sound source identification can be improved.

[0024] In step S12, the time difference value of the sound sources of the same type of information acquired by different audio acquisition devices is calculated, and the orientation information of the sound sources of different types of information relative to the origin of the coordinate system of the audio acquisition device array is calculated according to the time difference positioning method. In this implementation, the digital hearing aid is equipped with multiple audio acquisition devices (such as microphones). These devices are arranged in a certain geometry (such as linear, circular, or other array configurations) to capture sound from different directions. Each microphone in the array independently acquires sound signals from the surrounding environment. Due to the different positions of the microphones, the sound signals received by each microphone will differ in time and intensity. The spatial location of each sound source is determined by analyzing the time difference and intensity difference of the sound signals received by different microphones in the array. The specific process is as follows: extract features that can be used for time difference measurement from specific audio features of the sound information, such as the starting point, peak point, or specific frequency components of the signal. Since the speed of sound is known, the distance difference from the sound source to different devices can be calculated based on the time difference and the speed of sound. Using a geometric positioning algorithm, such as polygonal measurement, the azimuth information of the sound source relative to the microphone array is calculated based on the distance difference. The calculated sound source position is converted into coordinates relative to the coordinate system of the microphone array. The center of the microphone array or a certain reference point is set as the origin of the coordinate system, which can determine the approximate range or precise location of the sound source.

[0025] In one embodiment, such as Figure 3 As shown, in step S30, if the sound source is moving relative to the digital hearing aid, the sound signal corresponding to the sound source is subjected to high gain; if the sound source is stationary relative to the digital hearing aid, the sound signal corresponding to the sound source is subjected to low gain. This includes the following steps S31-S34: In step S31, the location information of the sound source of the same type of information is calculated relative to the displacement of the coordinate system with the digital hearing aid as the origin; In this implementation, for sound sources of the same type of information, their position coordinates at consecutive time points are calculated. The displacement of the sound source is determined by comparing the position coordinates at adjacent time points. The Pythagorean theorem can be used to calculate the displacement distance between two time points. By analyzing the displacement at consecutive time points, the movement trend of the sound source, such as velocity and acceleration, can be determined.

[0026] In step S32, if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is positive, the average value of the absolute value of the displacement per unit time is obtained as a direct proportional function relationship with the gain intensity. In step S33, if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is negative, the inverse proportional function relationship between the average value of the absolute value of the displacement per unit time and the gain intensity is obtained. In step S34, the gain intensity is calculated based on the functional relationship and the average value of the absolute value of the displacement per unit time, and the acoustic information of the sound source is amplified with the corresponding gain intensity. In this implementation, a positive displacement of the sound source's location relative to the origin of the digital hearing aid's coordinate system indicates that the sound source is moving away from the hearing aid. As the sound source's moving speed increases, the gain intensity also increases appropriately. This helps prevent sound distortion caused by the sound source moving away too quickly. Conversely, a negative displacement of the sound source's location relative to the origin of the digital hearing aid's coordinate system indicates that the sound source is approaching the hearing aid. This means that as the sound source's approach speed increases, the gain intensity decreases. This helps prevent sound distortion caused by the sound source approaching too quickly, thus preventing hearing damage to hearing-impaired patients. This intelligent amplification strategy helps improve the performance of hearing aids, especially in dynamic environments. It can help users more accurately locate and identify surrounding sounds, thereby improving the user's auditory experience and environmental perception. This is particularly important in scenarios such as preventing car accidents and assisting in hunting. It is worth noting that the above functional relationship aims to establish a proportional relationship between the sound source's displacement direction, speed, and gain intensity. Specific parameters need to be determined according to the degree of hearing impairment, which will not be elaborated in this application.

[0027] In one embodiment, the acoustic information in step S10 is filtered ambient sound, which does not contain speech signals or noise signals; In this implementation, the acoustic model can effectively eliminate speech and noise signals in specific frequency bands of the sound signal, which improves the performance of sound source localization technology to a certain extent. Furthermore, the speech signal is output by a separate gain system, which can effectively prevent background noise from masking important speech information.

[0028] The following are embodiments of the apparatus disclosed herein, which can be used to execute embodiments of the method disclosed herein.

[0029] Figure 4 This invention patent provides a block diagram of an environmental sound gain device based on two-dimensional sound source localization technology. This device can be implemented as part or all of an electronic device through software, hardware, or a combination of both. Figure 4 As shown, the device, applied to a digital hearing aid, includes: The sound source localization module 100 is used to calculate and process the sound information collected by the digital hearing aid based on two-dimensional sound source localization technology to obtain the directional information of the sound source of different types of information. The judgment module 200 is used to cache the location information of the sound source of different types of information, and determine the motion state of the sound source relative to the digital hearing aid based on the change of the location information of the sound source of the same type of information. The dynamic gain module 300 is used to perform dynamic high-gain amplification processing on the sound signal corresponding to the sound source if the sound source is in a moving state relative to the digital hearing aid; and The static gain module 400 is used to perform low-gain amplification processing on the sound signal corresponding to the sound source if the sound source is stationary relative to the digital hearing aid.

[0030] This disclosure effectively avoids the situation where background sounds mask important speech information by processing ambient sounds separately. Based on the different motion states of the sound source relative to the digital hearing aid, it can differentiate between stationary and moving sound sources, which can improve the recognition of moving sound sources. Outdoors, it helps hearing-impaired patients hear sounds such as vehicle driving and birdsong, improving their environmental perception and safety. Furthermore, by monitoring the movement path of the sound source, it can suppress loudness changes of rapidly approaching or moving sound sources, preventing the impact of excessively rapid loudness changes and preventing distortion or discrepancies that could affect the auditory experience.

[0031] In one embodiment, such as Figure 4 As shown, the sound source localization module 100 includes: The identification module 101 is used to acquire sound information around the digital hearing aid through an array of audio acquisition devices, and to identify the type information of all sound sources contained in the sound information according to an acoustic model. The positioning submodule 102 is used to calculate the time difference value of the sound sources of the same type of information acquired by different audio acquisition devices, and to calculate the orientation information of the sound sources of different types of information relative to the origin of the coordinate system of the audio acquisition device array according to the time difference positioning method.

[0032] In one embodiment, such as Figure 4 As shown, the dynamic gain module 300 includes: The measurement module 301 is used to calculate the displacement of the sound source of the same type of information relative to the origin of the coordinate system with the digital hearing aid as the origin. The first acquisition module 302 is used to acquire the positive proportional function relationship between the absolute value of the displacement and the gain intensity if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is positive. The second acquisition module 303 is used to acquire the inverse proportional function relationship between the absolute value of the displacement and the gain intensity if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is negative. The output module 304 is used to calculate the gain intensity based on the functional relationship and the average value of the absolute value of the displacement over a unit time, and to amplify the acoustic information of the sound source with the corresponding gain intensity.

[0033] In one embodiment, the sound information in the sound source localization module 100 is filtered ambient sound, which does not contain speech sound signals or noise signals.

[0034] 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.

[0035] This disclosure also provides another environmental sound gain device based on two-dimensional sound source localization technology: Figure 5 This is a block diagram illustrating an ambient sound enhancement device 800 based on two-dimensional sound source localization technology according to an exemplary embodiment. For example, device 800 may be a mobile phone, computer, digital broadcasting terminal, messaging device, game console, tablet device, medical device, fitness equipment, personal digital assistant, etc.

[0036] 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.

[0037] 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.

[0038] 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.

[0039] 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.

[0040] 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.

[0041] 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.

[0042] 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.

[0043] 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.

[0044] 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.

[0045] 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.

[0046] 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.

[0047] 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.

[0048] 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.

[0049] 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. An environmental sound gain method based on two-dimensional sound source localization technology, applied to digital hearing aids, characterized in that, The method includes: The sound information collected by the digital hearing aid is calculated and processed based on two-dimensional sound source localization technology to obtain the location information of the sound source of different types of information. The location information of the sound source with different types of information is cached, and the motion state of the sound source relative to the digital hearing aid is determined based on the change in the location information of the sound source with the same type of information. If the sound source is moving relative to the digital hearing aid, the sound signal corresponding to the sound source undergoes dynamic high-gain amplification; and If the sound source is stationary relative to the digital hearing aid, the sound signal corresponding to the sound source is amplified at a low gain.

2. The method according to claim 1, characterized in that, The sound information collected by the digital hearing aid is processed using two-dimensional sound source localization technology to obtain the location information of the sound source in different types, including: The sound information around the digital hearing aid is acquired by an array of audio acquisition devices, and the type information of all sound sources contained in the sound information is identified according to an acoustic model. Calculate the time difference between different audio acquisition devices acquiring the same type of information from sound sources, and calculate the orientation information of the sound sources of different types of information relative to the origin of the coordinate system with the audio acquisition device array as the time difference positioning method.

3. The method according to claim 1, characterized in that, If the sound source is moving relative to the digital hearing aid, the sound signal corresponding to the sound source is subjected to high gain; if the sound source is stationary relative to the digital hearing aid, the sound signal corresponding to the sound source is subjected to low gain, including: Calculate the displacement of the sound source of the same type of information relative to the origin of the coordinate system with the digital hearing aid as the origin; If the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is positive, obtain the direct proportional function relationship between the average value of the absolute value of the displacement per unit time and the gain intensity. If the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is negative, obtain the inverse proportional function relationship between the average value of the absolute value of the displacement per unit time and the gain intensity. The gain intensity is calculated based on the functional relationship and the average value of the absolute value of the displacement per unit time, and the acoustic information of the sound source is amplified with the corresponding gain intensity.

4. The method according to claim 1, characterized in that, Based on two-dimensional sound source localization technology, the sound information collected by the digital hearing aid is calculated and processed to obtain the sound information in the location information of the sound source of different types of information. The sound information is the filtered ambient sound, which does not contain speech sound signals and noise signals.

5. An environmental sound gain device based on two-dimensional sound source localization technology, applied to digital hearing aids, characterized in that, The device includes: The sound source localization module is used to calculate and process the sound information collected by the digital hearing aid based on two-dimensional sound source localization technology to obtain the directional information of the sound source of different types of information. The judgment module is used to cache the location information of the sound source of different types of information, and determine the motion state of the sound source relative to the digital hearing aid based on the change of the location information of the sound source of the same type of information. A dynamic gain module is used to perform dynamic high-gain amplification of the sound signal corresponding to the sound source if the sound source is moving relative to the digital hearing aid; and A static gain module is used to perform low-gain amplification of the sound signal corresponding to the sound source if the sound source is stationary relative to the digital hearing aid.

6. The apparatus according to claim 5, characterized in that, The sound source localization module includes: The identification module is used to acquire sound information around the digital hearing aid through an array of audio acquisition devices, and to identify the type information of all sound sources contained in the sound information according to an acoustic model. The positioning submodule is used to calculate the time difference between different audio acquisition devices acquiring the same type of information from sound sources, and to calculate the orientation information of the sound sources of different types of information relative to the origin of the coordinate system of the audio acquisition device array based on the time difference positioning method.

7. The apparatus according to claim 5, characterized in that, The dynamic gain module includes: The measurement module is used to calculate the displacement of the sound source of the same type of information relative to the origin of the coordinate system with the digital hearing aid as the origin; The first acquisition module is used to acquire the positive proportional function relationship between the average value of the absolute value of the displacement and the gain intensity if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is positive. The second acquisition module is used to acquire the inverse proportional function relationship between the absolute value of the displacement and the gain intensity if the displacement of the sound source relative to the origin of the coordinate system with the digital hearing aid as the origin is negative. The output module is used to calculate the gain intensity based on the functional relationship and the average value of the absolute value of the displacement over a unit time, and to amplify the acoustic information of the sound source with the corresponding gain intensity.

8. The apparatus according to claim 5, characterized in that, The sound information in the sound source localization module is the ambient sound after filtering, and the ambient sound does not contain speech sound signals or noise signals.

9. An environmental sound gain device based on two-dimensional sound source localization technology, 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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