Desktop type hearing-aid loudspeaker and intelligent human voice frequency band compensation system thereof

Abstract

The invention discloses a desktop type hearing-aid loudspeaker and an intelligent human voice frequency band compensation system thereof.The desktop type hearing-aid loudspeaker comprises a loudspeaker shell, an amplifier assembly is installed at one end of the loudspeaker shell in a clamped mode, a microphone is fixedly installed in the loudspeaker shell, and a circuit board is fixedly installed on one side of the bottom end of the inner wall of the loudspeaker shell; a storage battery is fixedly installed on the other side of the bottom end of the inner wall of the loudspeaker shell, the amplifier assembly comprises an amplifier shell and an amplifier body, the sound processing module is arranged, howling suppression, noise suppression, low frequency removal and millisecond frequency shift processing are conducted, the howling interference problem of traditional hearing-aid equipment is eliminated, and the sound processing efficiency is improved. The loudspeaker is more intelligent and humanized; by arranging the human voice frequency band compensation system, a non-intrusive mode is adopted to replace a traditional wearing type loudspeaking mode, the phenomenon that the ears of a user feel uncomfortable after the user wears the system for a long time is avoided, and acoustic optimization conforming to physiological features of hearing impairment is achieved through a self-adaptive algorithm.

Description

TECHNICAL FIELD

[0001] The present application relates to the technical field of hearing speaker, in particular to a desktop hearing speaker and an intelligent human voice frequency band compensation system thereof. BACKGROUND

[0002] The hearing speaker is one of the core components of the hearing aid, and mainly functions to convert the amplified electrical signal into sound waves into the ear canal to help the hearing impaired to perceive the sound more clearly. The working principle thereof is to collect the sound through the built-in microphone and convert it into an electrical signal, which is processed by the amplifier, and then the electrical signal drives the diaphragm of the speaker to generate sound waves, which are finally transmitted to the eardrum through the earplug or ear mold. The hearing speaker compensates for the hearing loss and enhances the weak sound to the perceptible range. Some high-end models can automatically adjust the volume and tone in different scenes such as TV, conversation and outdoor.

[0003] However, the conventional hearing speaker has the following disadvantages:

[0004] The conventional hearing speaker is generally in the form of earphones, and the user needs to wear it on the ear bone to clearly receive audio content such as TV broadcast and music. Long-term wearing of such hearing devices may cause the ear of the wearer to be tired, reducing the comfort of wearing the hearing speaker. SUMMARY

[0005] The present application aims to provide a desktop hearing speaker and an intelligent human voice frequency band compensation system thereof to solve the problem of the conventional hearing speaker being generally in the form of earphones, and the user needing to wear it on the ear bone to clearly receive audio content such as TV broadcast and music. Long-term wearing of such hearing devices may cause the ear of the wearer to be tired, reducing the comfort of wearing the hearing speaker.

[0006] To achieve the above-mentioned purpose, the present application provides the following technical scheme: a desktop hearing speaker, comprising a speaker housing, an amplifier assembly is clamped and installed at one end of the speaker housing, a microphone is fixedly installed inside the speaker housing, an electric circuit board is fixedly installed on one side of the bottom end of the inner wall of the speaker housing, a storage battery is fixedly installed on the other side of the bottom end of the inner wall of the speaker housing, the amplifier assembly comprises an amplifier housing and an amplifier main body, the middle part of one side of the amplifier housing is fixedly connected with one side of the amplifier main body, and a frequency modulation knob is rotatably connected with the top end of the speaker housing.

[0007] As a preferred technical scheme of the present application, a plurality of clamping shells are fixedly installed on one side of the inner wall of the speaker housing, a plurality of clamping columns are fixedly installed on the surface of the amplifier housing, and the plurality of clamping shells are correspondingly arranged with the plurality of clamping columns. The amplifier assembly is clamped on the clamping shell through the clamping column to complete the assembly of the amplifier assembly and the speaker housing.

[0008] As a preferred embodiment of the present invention, the amplifier housing has a plurality of grooves inside, and all of the grooves are fixedly set as hexagons. The concave-convex structure formed by the hexagonal grooves can effectively disperse impact energy and reduce vibration and noise transmission through buffering.

[0009] As a preferred embodiment of the present invention, a reinforcing plate is fixedly installed inside the speaker housing. The installation of the reinforcing plate increases the rigidity of the speaker housing itself and avoids damage to the speaker housing when it is struck by external force.

[0010] As a preferred embodiment of the present invention, the microphone, circuit board and amplifier body are all electrically connected to the battery. The microphone receives ambient sound waves and converts them into electrical signals. The circuit board performs frequency division, noise reduction and personalized compensation on the electrical signals. The amplifier body adjusts the gain according to the degree of hearing loss and adopts nonlinear amplification technology to greatly enhance weak sounds and compress strong sounds to avoid distortion.

[0011] As a preferred embodiment of the present invention, mounting bases are fixedly installed on both sides of the speaker housing, and a lifting ring is rotatably connected to the top of each of the two mounting bases. The user can carry the speaker housing by hooking the lifting ring with a shoulder strap with a hook.

[0012] The present invention provides an intelligent human voice frequency band compensation system for a desktop hearing aid speaker, comprising a human voice frequency band compensation system, wherein the human voice frequency band compensation system includes a hearing aid adjustment module, a sound processing module, and a frequency response setting module;

[0013] The hearing aid adjustment module performs certain adjustments to bring the hearing aid to the most suitable state.

[0014] The sound processing module performs howling suppression, noise suppression, low-frequency removal, and millisecond frequency shifting respectively;

[0015] The frequency response setting module intelligently adjusts the output sound pressure to the 90dB-105dB range across the entire frequency band from 500Hz to 20kHz.

[0016] As a preferred embodiment of the present invention, the hearing aid adjustment module includes a hearing detection submodule and a voice enhancement submodule;

[0017] The hearing detection submodule detects the hearing level of the left and right ears respectively to obtain hearing curves;

[0018] The human voice enhancement submodule uses a DSP algorithm to implement dynamic graded compensation in the key human voice frequency range of 500-3500Hz, and customizes parameters according to the compensation hearing curve.

[0019] The DSP algorithm convolution formula, used to process linear time-invariant systems of signal sequences, is calculated as follows:

[0020]

[0021] Where x[n] is the input signal sequence, h[n] is the filter coefficient, and y[n] is the output signal sequence; the formula for calculating the sequence length after convolution is:

[0022] M+N-1,

[0023] M is the length of the input sequence, N is the filter length, and the DSP algorithm calculation steps are as follows:

[0024] S1, Reverse Filter Coefficients: Reverse the filter coefficient sequence;

[0025] S2, Shift Input Sequence: Align the input sequence with the filter coefficients and multiply the corresponding positions;

[0026] S3. Accumulation result: The products are added in the order of displacement to obtain the output sequence.

[0027] As a preferred embodiment of the present invention, the sound processing module includes a loud noise protection submodule, a howling suppression submodule, a noise suppression submodule, a low-frequency removal submodule, and a millisecond frequency shifting submodule;

[0028] The loud noise protection submodule can control the output volume to protect the user's ears when a sudden impact sound occurs in the environment and the sound source with the maximum instantaneous volume of the impact sound is input.

[0029] The whistling suppression submodule reduces the occurrence of whistling.

[0030] The noise suppression submodule reduces noise that interferes with speech and provides better sound processing in noisy environments, thereby helping to obtain clearer speech signals for comfortable and better speech comprehension: the higher the number of segments, the better the noise reduction effect.

[0031] The low-frequency removal submodule prevents low-frequency noise from being excessively amplified. When low-frequency noise is detected, it suppresses the volume of the sound output to avoid discomfort caused by low-frequency noise when used in the external environment.

[0032] The millisecond frequency shifting submodule detects high-frequency subtones in real time and adjusts them to the low-frequency range in real time to improve semantic misunderstandings caused by unclear high-frequency sounds.

[0033] As a preferred embodiment of the present invention, the frequency response setting module includes a frequency response focusing submodule, a frequency response output submodule, and a frequency response range submodule;

[0034] The frequency response focusing submodule focuses on the human voice frequency band of 100Hz-20kHz;

[0035] The frequency response output submodule adjusts the output sound pressure to the 90dB-105dB range;

[0036] The frequency response range submodule receives frequencies ranging from 200Hz to 8kHz.

[0037] Compared with the prior art, the beneficial effects of the present invention are:

[0038] 1. By setting up a sound processing module, feedback suppression, noise suppression, low-frequency removal, and millisecond frequency shifting are performed respectively to eliminate feedback interference problems of traditional hearing aids, making the speaker more intelligent and user-friendly;

[0039] 2. By setting up a human voice frequency band compensation system, a non-invasive method is used to replace the traditional wearable speaker, avoiding ear discomfort for long-term wearers. Furthermore, an adaptive algorithm is used to achieve acoustic optimization that conforms to the physiological characteristics of hearing impairment.

[0040] 3. This device enables users with moderate to severe hearing loss to obtain a speech recognition experience superior to traditional hearing aids without relying on head-mounted devices, while maintaining a natural ability to perceive ambient sound fields. Attached Figure Description

[0041] Figure 1 This is a perspective view of the present invention;

[0042] Figure 2 This is a cross-sectional view of the speaker housing of the present invention;

[0043] Figure 3 This is a schematic diagram of the amplifier assembly architecture of the present invention;

[0044] Figure 4 This is a schematic diagram of the architecture of the human voice frequency band compensation system of the present invention;

[0045] Figure 5 This is a schematic diagram of the architecture of the hearing aid adjustment module of the present invention;

[0046] Figure 6 This is a schematic diagram of the architecture of the sound processing module of the present invention;

[0047] Figure 7 This is a schematic diagram of the architecture of the frequency response setting module of the present invention;

[0048] Figure 8 This is a flowchart of the DSP algorithm of the present invention;

[0049] Figure 9 This is a schematic diagram illustrating the hearing aid comparison of the present invention.

[0050] In the diagram: 1. Speaker housing; 2. Amplifier assembly; 21. Amplifier housing; 22. Clip; 23. Amplifier body; 24. Groove; 3. Tuning knob; 4. Hanging ring; 5. Mounting base; 6. Clip; 7. Reinforcing plate; 8. Microphone; 9. Circuit board; 10. Battery. Detailed Implementation

[0051] The technical solutions of the present invention will be clearly and completely described below with reference to the embodiments of the present invention. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those of ordinary skill in the art without creative effort are within the scope of protection of the present invention.

[0052] Please see Figures 1-9 The present invention provides a desktop hearing aid speaker, including a speaker housing 1, an amplifier assembly 2 is snapped onto one end of the speaker housing 1, a microphone 8 is fixedly installed inside the speaker housing 1, a circuit board 9 is fixedly installed on one side of the bottom of the inner wall of the speaker housing 1, and a battery 10 is fixedly installed on the other side of the bottom of the inner wall of the speaker housing 1. The amplifier assembly 2 includes an amplifier housing 21 and an amplifier body 23. The middle part of one side of the amplifier housing 21 is fixedly connected to one side of the amplifier body 23, and a frequency tuning knob 3 is rotatably connected to the top of the speaker housing 1.

[0053] Several retaining clips 6 are fixedly installed on one side of the inner wall of the speaker housing 1. Several retaining posts 22 are fixedly installed on the surface of the amplifier housing 21. The retaining clips 6 are respectively set with the retaining posts 22. The amplifier assembly 2 is fastened to the retaining clips 6 through the retaining posts 22, thus completing the assembly of the amplifier assembly 2 and the speaker housing 1.

[0054] The amplifier housing 21 has several grooves 24 inside, and all grooves 24 are fixedly set as hexagons. The concave and convex structure formed by the hexagonal grooves 24 can effectively disperse impact energy and reduce vibration and noise transmission through buffering.

[0055] A reinforcing plate 7 is fixedly installed inside the speaker housing 1. The installation of the reinforcing plate 7 increases the rigidity of the speaker housing 1 itself and prevents damage when the speaker housing 1 is hit by external force.

[0056] Microphone 8, circuit board 9 and amplifier body 23 are all electrically connected to battery 10. Microphone 8 receives ambient sound waves and converts them into electrical signals. Circuit board 9 performs frequency division, noise reduction and personalized compensation on the electrical signals. Amplifier body 23 adjusts the gain according to the degree of hearing loss and uses nonlinear amplification technology to greatly enhance weak sounds and compress strong sounds to avoid distortion.

[0057] The speaker housing 1 is fixedly mounted on both sides with mounting bases 5. The top of each mounting base 5 is rotatably connected to a hanging ring 4. The user can carry the speaker housing 1 by hooking the hanging ring 4 with a shoulder strap with a hook.

[0058] The present invention provides an intelligent human voice frequency band compensation system for a desktop hearing aid speaker, comprising a human voice frequency band compensation system, which includes a hearing aid adjustment module, a sound processing module, and a frequency response setting module.

[0059] The hearing aid adjustment module is adjusted to bring the hearing aid to its most suitable state.

[0060] The audio processing module performs feedback suppression, noise suppression, low-frequency removal, and millisecond frequency shifting respectively;

[0061] The frequency response setting module intelligently adjusts the output sound pressure to the 90dB-105dB range across the entire frequency band from 500Hz to 20kHz.

[0062] The hearing aid adjustment module includes a hearing test submodule and a voice enhancement submodule;

[0063] The hearing test submodule detects the hearing level of the left and right ears respectively and obtains hearing curves;

[0064] The human voice enhancement submodule uses DSP algorithms to implement dynamic graded compensation in the key human voice frequency range of 500-3500Hz, and customizes parameters according to the compensation hearing curve.

[0065] The DSP algorithm convolution formula, used to process linear time-invariant systems of signal sequences, is calculated as follows:

[0066]

[0067] Where x[n] is the input signal sequence, h[n] is the filter coefficient, and y[n] is the output signal sequence; the formula for calculating the sequence length after convolution is:

[0068] M+N-1,

[0069] M is the length of the input sequence, N is the filter length, and the DSP algorithm calculation steps are as follows:

[0070] S1, Reverse Filter Coefficients: Reverse the filter coefficient sequence;

[0071] S2, Shift Input Sequence: Align the input sequence with the filter coefficients and multiply the corresponding positions;

[0072] S3. Accumulation result: The products are added in the order of displacement to obtain the output sequence.

[0073] The sound processing module includes a loud noise protection submodule, a feedback suppression submodule, a noise suppression submodule, a low-frequency removal submodule, and a millisecond frequency shift submodule;

[0074] The loud noise protection submodule can control the output volume to protect the user's ears when a sudden impact sound occurs in the environment, or when the sound source is at its maximum volume at the moment of impact.

[0075] The howling suppression submodule reduces the occurrence of howling.

[0076] The noise suppression submodule reduces noise that interferes with speech, providing better sound processing in noisy environments, thereby helping to obtain a clearer speech signal for comfortable and better speech comprehension: the higher the segment number, the better the noise reduction effect;

[0077] The low-frequency removal submodule prevents low-frequency noise from being excessively amplified. When low-frequency noise is detected, it suppresses the volume of the sound output to avoid discomfort caused by low-frequency noise when used in the external environment.

[0078] The millisecond frequency shifting submodule detects high-frequency subtones in real time and adjusts them to the low-frequency range in real time to improve semantic misunderstandings caused by unclear high-frequency sounds.

[0079] The frequency response setting module includes a frequency response focusing submodule, a frequency response output submodule, and a frequency response range submodule;

[0080] The frequency response focusing submodule focuses on the human voice frequency band of 100Hz-20kHz;

[0081] The frequency response output submodule adjusts the output sound pressure level to the 90dB-105dB range;

[0082] The frequency response range submodule receives frequencies ranging from 200Hz to 8kHz.

[0083] In this invention, the user carries the speaker housing 1 by hooking the sling 4 with a shoulder strap with a hook. The amplifier assembly 2 is fastened to the housing 6 by the locking pin 22, completing the assembly of the amplifier assembly 2 and the speaker housing 1. The concave-convex structure formed by the hexagonal groove 24 can effectively disperse impact energy and reduce vibration and noise transmission through buffering. The microphone 8 receives ambient sound waves and converts them into electrical signals. The circuit board 9 performs frequency division, noise reduction, and personalized compensation on the electrical signals. The amplifier body 23 adjusts the gain according to the degree of hearing loss, using nonlinear amplification technology to significantly enhance weak sounds and compress strong sounds to avoid distortion. The hearing detection submodule detects the hearing level of the left and right ears to obtain hearing curves. The human voice enhancement submodule implements dynamic graded compensation in the key human voice frequency range of 500-3500Hz through DSP algorithms, and customizes parameters according to the compensation hearing curve. The loud noise protection submodule responds to sudden impact sounds in the environment. When a source inputs an audio source at its maximum instantaneous volume, the output volume can be controlled to protect the user's ears; the feedback suppression submodule reduces feedback; the noise suppression submodule reduces noise that interferes with speech, providing better sound processing in noisy environments, thus helping to obtain clearer speech signals for comfort and better speech comprehension; the higher the number of segments, the better the noise reduction effect; the low-frequency removal submodule prevents low-frequency noise from being excessively amplified, and when low-frequency noise is detected, it suppresses the sound output volume to avoid discomfort caused by low-frequency noise in external environments; the millisecond frequency shift submodule instantly detects high-frequency consonants and instantly tunes them to the low-frequency range to improve semantic misunderstandings caused by unclear high-frequency sounds; the frequency response focusing submodule focuses on the human voice frequency band of 100Hz-20kHz; the frequency response output submodule adjusts the output sound pressure to the 90dB-105dB range; the frequency response range submodule receives a frequency response range of 200Hz-8kHz, refer to the instruction manual. Figure 9 The red line represents the sound of this device up to 500Hz-20kHz, with the dB value compensated from 90dB to 105dB; the green line represents ordinary speakers with a dB value of only 80-90, especially at the high frequency of 10kHz, where it is only 70dB and does not have a compensation function.

[0084] Although the present invention has been described in detail with reference to the foregoing embodiments, those skilled in the art can still modify the technical solutions described in the foregoing embodiments or make equivalent substitutions for some of the technical features. Any modifications, equivalent substitutions, improvements, etc., made within the spirit and principles of the present invention should be included within the protection scope of the present invention.

Claims

1. A desktop hearing aid loudspeaker, comprising a loudspeaker housing (1), characterized in that: An amplifier assembly (2) is snapped onto one end of the speaker housing (1). A microphone (8) is fixedly installed inside the speaker housing (1). A circuit board (9) is fixedly installed on one side of the bottom of the inner wall of the speaker housing (1). A battery (10) is fixedly installed on the other side of the bottom of the inner wall of the speaker housing (1). The amplifier assembly (2) includes an amplifier housing (21) and an amplifier body (23). The middle part of one side of the amplifier housing (21) is fixedly connected to one side of the amplifier body (23). A tuning knob (3) is rotatably connected to the top of the speaker housing (1).

2. A desktop hearing aid speaker according to claim 1, characterized in that: Several clips (6) are fixedly installed on one side of the inner wall of the speaker housing (1), and several clips (22) are fixedly installed on the surface of the amplifier housing (21). The several clips (6) are respectively set to correspond to the several clips (22).

3. A desktop hearing aid speaker according to claim 1, characterized in that: The amplifier housing (21) has several grooves (24) inside, and all of the grooves (24) are fixedly set as hexagons.

4. A desktop hearing aid speaker according to claim 1, characterized in that: A reinforcing plate (7) is fixedly installed inside the speaker housing (1).

5. A desktop hearing aid speaker according to claim 1, characterized in that: The microphone (8), circuit board (9) and amplifier body (23) are all electrically connected to the battery (10).

6. A desktop hearing aid speaker according to claim 1, characterized in that: Mounting bases (5) are fixedly installed on both sides of the speaker housing (1), and the top of each mounting base (5) is rotatably connected to a lifting ring (4).

7. A desktop hearing aid loudspeaker intelligent human voice frequency band compensation system according to any one of claims 1-6, comprising a human voice frequency band compensation system, characterized in that: The human voice frequency band compensation system includes a hearing aid adjustment module, a sound processing module, and a frequency response setting module; The hearing aid adjustment module performs certain adjustments to bring the hearing aid to the most suitable state. The sound processing module performs howling suppression, noise suppression, low-frequency removal, and millisecond frequency shifting respectively; The frequency response setting module intelligently adjusts the output sound pressure to the 90dB-105dB range across the entire frequency band from 500Hz to 20kHz.

8. The intelligent human voice frequency band compensation system for a desktop hearing aid speaker according to claim 7, characterized in that: The hearing aid adjustment module includes a hearing detection submodule and a voice enhancement submodule; The hearing detection submodule detects the hearing level of the left and right ears respectively to obtain hearing curves; The human voice enhancement submodule uses a DSP algorithm to implement dynamic graded compensation in the key human voice frequency range of 500-3500Hz, and customizes parameters according to the compensation hearing curve. The DSP algorithm convolution formula, used to process linear time-invariant systems of signal sequences, is calculated as follows: Where x[n] is the input signal sequence, h[n] is the filter coefficient, and y[n] is the output signal sequence; the formula for calculating the sequence length after convolution is: M+N-1, M is the length of the input sequence, N is the filter length, and the DSP algorithm calculation steps are as follows: S1, Reverse Filter Coefficients: Reverse the filter coefficient sequence; S2, Shift Input Sequence: Align the input sequence with the filter coefficients and multiply the corresponding positions; S3. Accumulation result: The products are added in the order of displacement to obtain the output sequence.

9. The intelligent human voice frequency band compensation system for a desktop hearing aid speaker according to claim 7, characterized in that: The sound processing module includes a loud noise protection submodule, a howling suppression submodule, a noise suppression submodule, a low-frequency removal submodule, and a millisecond frequency shift submodule; The loud noise protection submodule can control the output volume to protect the user's ears when a sudden impact sound occurs in the environment and the sound source is at its maximum volume at that moment. The whistling suppression submodule reduces the occurrence of whistling. The noise suppression submodule reduces noise that interferes with speech and provides better sound processing in noisy environments, thereby helping to obtain clearer speech signals for comfortable and better speech comprehension: the higher the number of segments, the better the noise reduction effect. The low-frequency removal submodule prevents low-frequency noise from being excessively amplified. When low-frequency noise is detected, it suppresses the volume of the sound output to avoid discomfort caused by low-frequency noise when used in the external environment. The millisecond frequency shifting submodule detects high-frequency subtones in real time and adjusts them to the low-frequency range in real time to improve semantic misunderstandings caused by unclear high-frequency sounds.

10. The intelligent human voice frequency band compensation system for a desktop hearing aid loudspeaker according to claim 7, characterized in that: The frequency response setting module includes a frequency response focusing submodule, a frequency response output submodule, and a frequency response range submodule; The frequency response focusing submodule focuses on the human voice frequency band of 100Hz-20kHz; The frequency response output submodule adjusts the output sound pressure to the 90dB-105dB range; The frequency response range submodule receives frequencies ranging from 200Hz to 8kHz.

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