A sound signal processing method and apparatus based on an acoustic model
By using signal processing methods based on acoustic models to filter and adjust the amplification factor of pure tone signals, the problem of low sound resolution in hearing aids is solved, thereby improving the auditory perception ability of hearing-impaired patients.
Patent Information
- Authority / Receiving Office
- CN · China
- Patent Type
- Applications(China)
- Current Assignee / Owner
- ZUODIAN IND (HUBEI) CO LTD
- Filing Date
- 2024-03-05
- Publication Date
- 2026-06-05
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Figure CN122160694A_ABST
Abstract
Description
Technical Field
[0001] This invention patent relates to the field of air conduction hearing aid technology, specifically to a sound signal processing method and device based on an acoustic model. Background Technology
[0002] A hearing aid is a small amplification device used by people with hearing loss to compensate for their hearing loss. A microphone module inputs amplified sound signals into an amplifier module, and a diaphragm module converts the amplified sound signals into sound energy, which is then output to the hearing-impaired person through headphones or an in-ear speaker. Adding a filtering module to the hearing aid system can reduce noise interference and improve the auditory environment. However, hearing-impaired patients have low sound perception, and hearing aids, by indiscriminately amplifying sounds within a specific frequency range, cannot effectively improve sound resolution. Therefore, this invention provides a sound signal processing method and apparatus based on an acoustic model to solve the above-mentioned technical problems.
[0003] Invention Patent Content To address the shortcomings of existing technologies, this invention provides a sound signal processing method and apparatus based on an acoustic model to improve the resolution of hearing aids.
[0004] According to a first aspect of the present disclosure, a preferred embodiment of the present invention provides a sound signal processing method based on an acoustic model for an air conduction hearing aid, the method comprising: The system simulates and replicates available sound signals from the ambient sound signals in which the hearing aid is located, and the number of available sound signals is at least two. Based on existing acoustic models, the available acoustic signals are subjected to differentiated filtering to form different pure tone signals; and The pure tone signals are differentiated according to their categories, and the processed pure tone signals are then integrated and output.
[0005] In one embodiment, a simulated copy of available sound signals from the ambient sound signal where the hearing aid is located is made, and the number of available sound signals is at least two, including: Acquire ambient sounds around the hearing aid and convert the ambient sounds into ambient sound signals; The ambient sound signal is filtered out to remove redundant signals and obtain usable sound signals. The redundant signals include noise signals and invalid signals that are beyond the range of human hearing. Multiple available acoustic signals are generated synchronously, and all available acoustic signals are sorted.
[0006] In one embodiment, the available acoustic signals are subjected to differential filtering based on an existing acoustic model to form different pure tone signals, including: The available acoustic signals are analyzed, and all the acoustic models contained therein are matched. Generate all absolute complements containing all combinations of the acoustic models, wherein the number of complements is consistent with the number of available acoustic signals; Sort the complement set and associate the sorted complement set with each available acoustic signal; Based on the acoustic models within different complements, the available acoustic signals associated with the complements are filtered to obtain multiple pure tone signals.
[0007] In one embodiment, the differential processing of the pure tone signal according to category is preferably an adjustment of the amplification factor of the pure tone signal, and the amplification factor is a preset value.
[0008] According to a second aspect of the present disclosure, the present invention provides a sound signal processing device based on an acoustic model for an air conduction hearing aid, the device comprising: The copying module is used to simulate and copy available sound signals from the ambient sound signals in which the hearing aid is located, and the number of available sound signals is at least two. A differentiation module is used to perform differential filtering on the available acoustic signals based on an existing acoustic model to form different pure tone signals; and The adjustment module is used to differentiate the pure tone signals according to their categories and then integrate and output the processed pure tone signals.
[0009] In one embodiment, the copying module includes: The acquisition module is used to acquire ambient sounds around the hearing aid and convert the ambient sounds into ambient sound signals; The preprocessing module is used to filter out redundant signals in the ambient sound signal to obtain a usable sound signal, wherein the redundant signals include noise signals and invalid signals that are beyond the range of human hearing. A generation module is used to synchronously generate multiple available sound signals and sort all the available sound signals.
[0010] In one embodiment, the differentiation module includes: The analysis module is used to analyze the available acoustic signals and match all the acoustic models contained therein; A combination module is used to generate all absolute complements containing all combinations of the acoustic models, wherein the number of complements is consistent with the number of available acoustic signals; The association module is used to sort the complement set and establish an association between the sorted complement set and each available acoustic signal; The filtering module is used to filter the available acoustic signals associated with the complements according to the acoustic models within different complements, so as to obtain multiple pure tone signals.
[0011] In one embodiment, the adjustment module's differential processing of the pure tone signal according to category is preferably an adjustment of the amplification factor of the pure tone signal, and the amplification factor is a preset value.
[0012] According to a third aspect of the present disclosure, the present invention provides a sound signal processing apparatus based on an acoustic model, 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 sound signal processing method and device based on acoustic models provided by this invention analyzes the available sound signals in the environment, matches relevant acoustic model combinations, and masks the remaining signals in the available sound signals using the acoustic models, thereby obtaining pure tone signals of different sound types. Combined with the degree of hearing impairment of hearing-impaired patients, the pure tone signals are amplified and output differently, which can increase the ratio of a specific sound to the other sounds, effectively improve the resolution of the specific sound, compensate for the hearing impairment of hearing-impaired patients to a certain extent, and greatly improve the performance of hearing aids.
[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 sound signal processing method based on an acoustic model provided for this invention patent; Figure 2 A flowchart of step S10 in an acoustic model-based sound signal processing method provided by this invention patent; Figure 3 A flowchart of step S20 in an acoustic model-based sound signal processing method provided by this invention patent; Figure 4 A block diagram of a sound signal processing device based on an acoustic model provided for this invention patent; Figure 5 A block diagram of a replication module in an acoustic model-based sound signal processing device provided for this invention patent; Figure 6 A block diagram of a differentiation module in an acoustic model-based sound signal processing device provided for this invention patent. Figure 7 This invention provides a block diagram of another acoustic model-based sound signal processing device. 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 a sound signal processing method based on an acoustic model, applied to an air conduction hearing aid terminal. This terminal 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 sound signal processing method based on an acoustic model for use with air conduction hearing aids, such as... Figure 1 As shown, the method includes the following steps S10-S30: In step S10, the available sound signals in the ambient sound signals of the hearing aid are simulated and copied, and the number of available sound signals is at least two. In this implementation, copying available acoustic signals can establish multiple signal processing channels.
[0020] In step S20, the available acoustic signals are subjected to differential filtering based on existing acoustic models to form different pure tone signals; and In this implementation, different types of sounds are selected from different signal processing channels, which greatly improves the purity of the sound.
[0021] In step S30, the pure tone signals are differentiated according to their categories, and the processed pure tone signals are integrated and output. In this implementation, the magnitude, frequency, amplitude, and duration of different types of sounds are adjusted to enhance or weaken the perception intensity of specific types of sounds by hearing-impaired patients. The output can be either integrated by a signal integrator and played through a speaker, or the individual signals can be played synchronously through a speaker.
[0022] Among them, such as Figure 2 As shown, in step S10, the available sound signals in the ambient sound signal where the hearing aid is located are simulated and copied, and the number of available sound signals is at least two, including: In step S11, the ambient sound around the hearing aid is acquired and converted into an ambient sound signal; In this embodiment, the microphone converts ambient sounds around the hearing aid into electrical signals.
[0023] In step S12, redundant signals in the ambient sound signal are filtered out to obtain a usable sound signal, wherein the redundant signals include noise signals and invalid signals that are beyond the range of human hearing. In this implementation, a filter is used to remove signals of specific frequencies from the ambient sound signal to obtain a usable sound signal. Only the usable sound signal is processed, which reduces the burden of data processing.
[0024] In step S13, multiple available sound signals are generated synchronously, and all available sound signals are sorted.
[0025] In this implementation, a signal generator is used to generate multiple usable sound signals, and sorting the usable sound signals serves as a marking function.
[0026] In one embodiment, such as Figure 3 As shown, in step S20, the available acoustic signals are subjected to differential filtering based on existing acoustic models to form different pure tone signals, including: In step S21, the available acoustic signals are analyzed and all the acoustic models contained therein are matched. In this implementation, the acoustic model is one of the most important parts of the speech recognition system. By identifying specific features in the available acoustic signals, the corresponding acoustic model can be matched.
[0027] In step S22, all absolute complements containing all the acoustic model combinations are generated, wherein the number of complements is consistent with the number of available acoustic signals; In step S23, the complement is sorted, and the sorted complement is associated with each available acoustic signal; In this implementation, if the number of acoustic models in step S21 is n, then the number of acoustic models in the complement set is n-1, and each complement set is different.
[0028] In step S24, the available acoustic signals associated with the complements are filtered according to the acoustic models within the different complements to obtain multiple pure tone signals; In this implementation, each complement corresponds to the missing acoustic model in the acoustic model combination and the pure tone signal generated after filtering, thereby determining the sound type of the pure tone signal.
[0029] In one embodiment, the differential processing of the pure tone signal according to the category is preferably to adjust the amplification factor of the pure tone signal, and the amplification factor is a preset value; In this implementation, assuming that a certain ambient sound signal contains a pure voice signal and a pure wind signal, adjusting the amplification factor of the pure voice signal to be much greater than that of the pure wind signal can effectively highlight the voice, reduce communication barriers for users, and reduce wind noise. It is worth noting that the amplification factor of the pure voice signal should not be adjusted to zero to prevent excessively reducing the user's perception of environmental changes.
[0030] The following are embodiments of the apparatus disclosed herein, which can be used to execute embodiments of the method disclosed herein.
[0031] Figure 4 This invention provides a block diagram of a sound signal processing device based on an acoustic model. 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, used for an air conduction hearing aid, includes: The copying module 100 is used to simulate and copy available sound signals in the ambient sound signals of the hearing aid, and the number of available sound signals is at least two. Differentiation module 200 is used to perform differential filtering on the available acoustic signals based on an existing acoustic model to form different pure tone signals; and The adjustment module 300 is used to differentiate the pure tone signals according to their categories and then integrate and output the processed pure tone signals.
[0032] This disclosure analyzes available sound signals in the environment to match relevant acoustic model combinations. By masking other signals in the available sound signals using acoustic models, pure tone signals of different sound types can be obtained. Combined with the degree of hearing impairment of hearing-impaired patients, the pure tone signals are amplified and output differentially, which can increase the ratio of a specific sound to other sounds, effectively improve the resolution of the specific sound, compensate for the hearing impairment of hearing-impaired patients to a certain extent, and greatly improve the performance of hearing aids.
[0033] In one embodiment, such as Figure 5 As shown, the copying module 100 includes: Acquisition module 101 is used to acquire ambient sounds around the hearing aid and convert the ambient sounds into ambient sound signals; The preprocessing module 102 is used to filter out redundant signals in the ambient sound signal to obtain a usable sound signal, wherein the redundant signals include noise signals and invalid signals that are beyond the range of human hearing. The generation module 103 is used to synchronously generate multiple available sound signals and sort all the available sound signals.
[0034] In one embodiment, such as Figure 6 As shown, the differentiation module 200 includes: Analysis module 201 is used to analyze the available acoustic signals and match all the acoustic models contained therein; Combination module 202 is used to generate all absolute complements containing all the combinations of the acoustic models, wherein the number of complements is consistent with the number of available acoustic signals; The association module 203 is used to sort the complement set and establish an association between the sorted complement set and each available acoustic signal; The filtering module 204 is used to filter the available acoustic signals associated with the complements according to the acoustic models in different complements to obtain multiple pure tone signals.
[0035] In one embodiment, the adjustment module 300's differential processing of the pure tone signal according to category is preferably adjusting the amplification factor of the pure tone signal, and the amplification factor is a preset value.
[0036] 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.
[0037] This disclosure also provides another sound signal processing apparatus based on an acoustic model: Figure 7 This is a block diagram illustrating an acoustic model-based sound signal processing apparatus 800 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.
[0038] Reference Figure 7 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.
[0039] 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.
[0040] 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.
[0041] 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.
[0042] 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.
[0043] 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 also includes a speaker for outputting audio signals.
[0044] 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.
[0045] 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.
[0046] 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.
[0047] 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 another exemplary embodiment, the communication component 816 further 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.
[0048] 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.
[0049] 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.
[0050] 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.
[0051] 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 sound signal processing method based on an acoustic model, characterized in that, For use in air conduction hearing aids, the method includes: The system simulates and replicates available sound signals from the ambient sound signals in which the hearing aid is located, and the number of available sound signals is at least two. Based on existing acoustic models, the available acoustic signals are subjected to differentiated filtering to form different pure tone signals; and The pure tone signals are differentiated according to their categories, and the processed pure tone signals are then integrated and output.
2. The method according to claim 1, characterized in that, Simulates and replicates available sound signals from the ambient sound signal where the hearing aid is located, and the number of available sound signals is at least two, including: Acquire ambient sounds around the hearing aid and convert the ambient sounds into ambient sound signals; The ambient sound signal is filtered out to remove redundant signals and obtain usable sound signals. The redundant signals include noise signals and invalid signals that are beyond the range of human hearing. Multiple available acoustic signals are generated synchronously, and all available acoustic signals are sorted.
3. The method according to claim 1, characterized in that, Based on existing acoustic models, the available acoustic signals are subjected to differentiated filtering to form different pure tone signals, including: The available acoustic signals are analyzed, and all the acoustic models contained therein are matched. Generate all absolute complements containing all combinations of the acoustic models, wherein the number of complements is consistent with the number of available acoustic signals; Sort the complement set and associate the sorted complement set with each available acoustic signal; Based on the acoustic models within different complements, the available acoustic signals associated with the complements are filtered to obtain multiple pure tone signals.
4. The method according to claim 1, characterized in that, The preferred method for differentiating pure tone signals according to category is to adjust the amplification factor of the pure tone signals, where the amplification factor is a preset value.
5. A sound signal processing device based on an acoustic model, characterized in that, For use in air conduction hearing aids, the device includes: The copying module is used to simulate and copy available sound signals from the ambient sound signals in which the hearing aid is located, and the number of available sound signals is at least two. A differentiation module is used to perform differential filtering on the available acoustic signals based on an existing acoustic model to form different pure tone signals; and The adjustment module is used to differentiate the pure tone signals according to their categories and then integrate and output the processed pure tone signals.
6. The apparatus according to claim 5, characterized in that, The replication module includes: The acquisition module is used to acquire ambient sounds around the hearing aid and convert the ambient sounds into ambient sound signals; The preprocessing module is used to filter out redundant signals in the ambient sound signal to obtain a usable sound signal, wherein the redundant signals include noise signals and invalid signals that are beyond the range of human hearing. A generation module is used to synchronously generate multiple available sound signals and sort all the available sound signals.
7. The apparatus according to claim 5, characterized in that, The differentiation module includes: The analysis module is used to analyze the available acoustic signals and match all the acoustic models contained therein; A combination module is used to generate all absolute complements containing all combinations of the acoustic models, wherein the number of complements is consistent with the number of available acoustic signals; The association module is used to sort the complement set and establish an association between the sorted complement set and each available acoustic signal; The filtering module is used to filter the available acoustic signals associated with the complements according to the acoustic models within different complements, so as to obtain multiple pure tone signals.
8. The apparatus according to claim 5, characterized in that, The preferred method for the adjustment module to differentiate the pure tone signal according to its category is to adjust the amplification factor of the pure tone signal, and the amplification factor is a preset value.
9. A sound signal processing device based on an acoustic model, 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.