Audio processing method, electronic equipment and readable storage medium

Abstract

The invention relates to the technical field of audio processing, in particular to an audio processing method, electronic equipment and a readable storage medium. In the method, a device which can be worn on an ear of a user is used for receiving a sound signal, then a filter is used for carrying out frequency division on the sound signal to obtain a low-frequency sound signal and a high-frequency sound signal, and a steady-state sound signal and a transient sound signal in the low-frequency sound signal and the high-frequency sound signal are extracted. And the steady-state sound signal and the transient sound signal are processed based on the personalized calibration parameters corresponding to the user, so that the calibrated sound signal finally heard by the user conforms to the target psychological acoustic loudness of the user. Therefore, hearing protection of the user can be realized.

Description

Technical Field

[0001] This application relates to the technical field of audio processing, and particularly relates to an audio processing method, an electronic device, and a readable storage medium. Background Art

[0002] According to current research results, the brains of children process sounds differently from those of adults. The hearing systems of children are not fully developed, and their auditory nerves and auditory cortices have relatively weak sound processing capabilities and are easily disturbed and stimulated by sounds. Therefore, for the hearing systems of children, too high an environmental sound pressure level (for example, exceeding 70 decibels) and sudden sounds can damage the hearing systems of children. However, in some places, the environmental noise is relatively large, and there are sounds that can affect the hearing systems of children. For example, there is relatively large environmental noise in places such as cinemas, KTVs, and game arcades. Staying in such environments for a long time will affect the hearing of children. Summary of the Invention

[0003] To solve the above problems, embodiments of this application provide an audio processing method, an electronic device, and a readable storage medium.

[0004] In a first aspect, an embodiment of this application provides an audio processing method applied to a headset. The method includes: receiving a first sound signal and obtaining a first sound pressure level of the first sound signal; based on the first sound pressure level, obtaining a first psychoacoustic loudness of a first target user for the first sound signal; based on the first psychoacoustic loudness being inconsistent with the target psychoacoustic loudness of the first target user, obtaining a first calibration parameter corresponding to the first target user to calibrate the first sound signal to obtain a second sound signal, where the second psychoacoustic loudness of the first target user for the second sound signal is consistent with the target psychoacoustic loudness.

[0005] It can be understood that the first sound signal may refer to the sound signal received by the headset microphone mentioned in the embodiments of this application, and the second sound signal may refer to the target pass-through signal that finally reaches the eardrum of the human ear mentioned in the embodiments of this application.

[0006] Based on the above solution, the user can receive a sound signal by wearing the wireless headset 10, and then the wireless headset 10 calibrates the sound signal based on the user's personalized calibration parameter to obtain a target pass-through signal and transmit it to the human ear. Moreover, the psychoacoustic loudness corresponding to the target pass-through signal is consistent with the target psychoacoustic loudness of the user, thereby protecting the user's hearing.

[0007] In a possible implementation of the above first aspect, obtaining the first psychoacoustic loudness of the first target user for the first sound signal based on the first sound pressure level includes: obtaining the first passive isolation degree of the earphone, and performing Bark spectral analysis on the first sound signal based on the first passive isolation degree and the first sound pressure level to obtain the first psychoacoustic loudness of the first target user for the first sound signal.

[0008] It can be understood that the first passive isolation degree refers to the physical isolation of the earphone for the sound signal. The passive isolation degree can be preset in the development and design stage of the earphone, or obtained through real-time detection. For earphones that do not support detecting the passive isolation degree, the passive isolation degree preset in the development and design stage can be directly obtained.

[0009] It can be understood that the determination method of the first target user can be: the electronic device that establishes a communication connection with the earphone can determine that the first target user is currently using the earphone in response to the user's selection operation based on the display interface of the electronic device.

[0010] It can be understood that the input of the Bark spectral analysis is the sound signal. The first passive isolation degree can be filtered out during the Bark spectral analysis, so that the result of the Bark spectral analysis can be more accurate. The output (result) of the Bark spectral analysis is the Bark curve corresponding to the sound signal.

[0011] It can be understood that the psychoacoustic loudness corresponding to the sound signal can be directly obtained from the Bark curve of the sound signal obtained based on the Bark spectral analysis. The psychoacoustic loudness refers to the subjective perception of the sound signal by the human ear.

[0012] In a possible implementation of the above first aspect, obtaining the first calibration parameter corresponding to the first target user is determined by the following method: the earphone plays a first test tone to the first target user; in response to the operation of the first target user adjusting the first test tone to a second test tone; obtaining the parameter values of the time-domain calibration parameter and the frequency-domain calibration parameter to obtain the first calibration parameter, where the first calibration parameter includes the first time-domain calibration parameter and the first frequency-domain calibration parameter.

[0013] It can be understood that in response to the operation of the first target user adjusting the first test tone to a second test tone, it can refer to the operation of clicking, sliding or dragging on the personalized calibration interface based on the personalized calibration interface mentioned in the embodiments of the present application to obtain the time-domain calibration parameter and the frequency-domain calibration parameter.

[0014] It can be understood that the first calibration parameter may refer to the time-domain calibration parameter and the frequency-domain calibration parameter mentioned in the embodiments of the present application; the first time-domain calibration parameter may refer to "Attack", "Decay", "Sustain", "Release", and "peak amplitude" mentioned in the embodiments of the present application. The first frequency-domain calibration parameter may refer to the frequency response mentioned in the embodiments of the present application.

[0015] In a possible implementation of the first aspect above, obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal includes: performing frequency division processing on the first sound signal to obtain the steady-state amplitude of the first sound signal; determining whether the steady-state amplitude of the first sound signal is greater than the first amplitude threshold; corresponding to the steady-state amplitude of the first sound signal being greater than the first amplitude threshold, calibrating the first sound signal based on the first calibration parameter.

[0016] In a possible implementation of the first aspect above, obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal further includes: performing frequency division processing on the first sound signal to obtain the transient amplitude of the first sound signal; determining whether the transient amplitude of the first sound signal is greater than the second amplitude threshold; corresponding to the transient amplitude of the first sound signal being greater than the second amplitude threshold, calibrating the first sound signal based on the first calibration parameter.

[0017] In a possible implementation of the first aspect above, performing frequency division processing on the first sound signal to obtain the steady-state amplitude of the first sound signal includes: using a filter to perform frequency division on the first sound signal to obtain a first high-frequency sound signal and a first low-frequency sound signal; performing two-dimensional median filtering on the first high-frequency sound signal to obtain a first high-frequency steady-state sound signal of the first high-frequency sound signal; performing two-dimensional median filtering on the first low-frequency sound signal to obtain a first low-frequency steady-state sound signal of the first low-frequency sound signal; performing equal-loudness weighting calculation and weighted amplitude calculation on the amplitude of the first high-frequency steady-state sound signal and the amplitude of the first low-frequency steady-state sound signal to obtain the steady-state amplitude of the first sound signal.

[0018] In a possible implementation of the first aspect above, performing frequency division processing on the first sound signal to obtain the transient amplitude of the first sound signal includes: using a filter to perform frequency division on the first sound signal to obtain a first high-frequency sound signal and a first low-frequency sound signal; performing two-dimensional median filtering on the first high-frequency sound signal to obtain a first high-frequency transient sound signal of the first high-frequency sound signal; performing two-dimensional median filtering on the first low-frequency sound signal to obtain a first low-frequency transient sound signal of the first low-frequency sound signal; performing equal-loudness weighting calculation and weighted amplitude calculation on the amplitude of the first high-frequency transient sound signal and the amplitude of the first low-frequency transient sound signal to obtain the transient amplitude of the first sound signal.

[0019] In a possible implementation of the above first aspect, corresponding to the steady-state amplitude of the first sound signal being greater than the first amplitude threshold, calibrating the first sound signal based on the first calibration parameter includes: using the first frequency-domain calibration parameter as the filtering coefficient of the filter, and calibrating the steady-state sound signal of the first sound signal based on the filter to obtain the calibrated steady-state sound signal of the first sound signal.

[0020] In a possible implementation of the above first aspect, corresponding to the transient amplitude of the first sound signal being greater than the second amplitude threshold, calibrating the first sound signal based on the first calibration parameter includes: using the first time-domain calibration parameter as the time-domain compression parameter of the pass-through compressor, and calibrating the transient sound signal of the first sound signal based on the pass-through compressor to obtain the calibrated transient sound signal of the first sound signal.

[0021] In a possible implementation of the above first aspect, obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal to obtain a second sound signal includes: using a synthesis filter to synthesize the steady-state sound signal of the calibrated first sound signal and the transient sound signal of the calibrated first sound signal to obtain the second sound signal.

[0022] In a possible implementation of the above first aspect, the first calibration parameter is the calibration parameter corresponding to children.

[0023] In a second aspect, an embodiment of the present application provides a headset, which includes: a memory for storing instructions; and a processor for executing the instructions to implement the audio processing method provided by the above first aspect and any possible implementation of the first aspect.

[0024] In a third aspect, an embodiment of the present application provides a readable storage medium, on which instructions are stored, and when the instructions are executed on an electronic device, the headset is caused to execute the audio processing method provided by the above first aspect and any possible implementation of the first aspect. Description of the Drawings

[0025] Figure 1 According to some embodiments of the present application, a schematic diagram of a scenario where a user watches a movie in a cinema is shown;

[0026] Figure 2 According to some embodiments of the present application, a schematic diagram of a scenario where a child wears wireless headphones to watch a movie in a cinema is shown;

[0027] Figure 3 According to some embodiments of the present application, a schematic diagram of the working principle of a wireless headset 10 for transmitting sound signals is shown;

[0028] Figure 4According to some embodiments of the present application, a specific flowchart of an audio processing method is shown;

[0029] Figure 5 According to some embodiments of the present application, a schematic diagram of frequency division of a sound signal is shown;

[0030] Figure 6 According to some embodiments of the present application, a schematic diagram of a personalized calibration interface 300 is shown;

[0031] Figure 7 According to some embodiments of the present application, a schematic diagram of a speaker playing an in-phase pass-through signal SPK(t) is shown;

[0032] Figure 8 According to some embodiments of the present application, a schematic diagram of a speaker transmitting a calibration signal to the eardrum of a human ear is shown. Detailed implementation manners

[0033] Exemplary embodiments of the present application include, but are not limited to, an audio processing method, an electronic device, and a readable storage medium.

[0034] It can be understood that the electronic device provided in the embodiments of the present application can be various earphones, such as wireless earphones, wired earphones, and augmented reality (AR) devices, virtual reality (VR) devices, etc. that support the pass-through function, or an electronic device including the pass-through mode, which is not limited herein.

[0035] It can be understood that the pass-through function means that when the electronic device is in the pass-through mode, it does not completely isolate the external sound, so that the user can also hear the external environmental sound when wearing the electronic device.

[0036] It can be understood that the electronic device provided in the embodiments of the present application can be a noise-canceling earphone or an ordinary earphone, which is not limited herein. For the convenience of understanding the audio processing method provided in the embodiments of the present application, the wireless earphone is taken as an example in the embodiments of the present application for illustration.

[0037] For the convenience of understanding the audio processing method provided in the embodiments of the present application, the relevant terms involved in the embodiments of the present application are first introduced.

[0038] (1) Sound pressure level

[0039] The sound pressure level (SPL), SPL can also be called the sound pressure level, and the unit of SPL is Pa.

[0040] (2) Bark spectrum analysis

[0041] Bark spectral analysis is a commonly used analysis method in signal processing and psychoacoustic research. It can convert the frequency-domain signal corresponding to sound into a concept that is more in line with human auditory perception, that is, frequency bands based on human auditory perception.

[0042] Bark spectral analysis means dividing the frequency-domain signal of a sound signal into a series of uniformly distributed frequency bands, and this series of frequency bands can map the frequency of the signal to 24 critical frequency bands in psychoacoustics. The human ear structure can resonate with the frequency points in the above 24 critical frequency bands, and the width of one critical frequency band is equal to one Bark. Simply put, Bark spectral analysis is to convert (map) physical frequencies to psychoacoustic frequencies to obtain the human ear perception curve of the sound signal perceived by different users.

[0043] (3) Psychoacoustic loudness

[0044] Psychoacoustic loudness is a measure that converts the intensity of sound into the intensity perceived by the human ear. It is designed according to the human ear perception curve and can accurately reflect the subjective perception of sound by the human ear. For example, the psychoacoustic loudness of children is 15 sone, and that of adults is 20 sone.

[0045] The technical solution of this application will be introduced below in conjunction with the accompanying drawings.

[0046] As mentioned above, the environmental noise in some places can damage the hearing systems of children. For example, as Figure 1 shown, in the movie-watching scenario in a cinema, child A is watching a movie with adults B and C. It can be understood that in order to provide users with a good movie-watching experience, the movie sound in the cinema is generally amplified, and the movie sound is relatively loud (for example, 100 decibels) so that users in the entire screening hall can get an immersive movie-watching experience. However, the hearing systems of children are not fully developed, so their ability to process sound is relatively weak and they are easily disturbed and stimulated by sound. If the external environmental sound pressure level is too high (for example, exceeding 70 decibels), or there are sudden and harsh sounds in the external environment, such as suddenly increasing the decibel level of the movie sound during the movie screening, then these sudden and high-decibel sounds will affect the hearing systems of children and cause damage to their hearing systems.

[0047] To solve the above problems, the embodiments of this application provide an audio processing method. This method uses a device that users can wear on their ears, such as headphones, to process the sound signal according to the target psychoacoustic loudness of different wearing users, based on the calibration parameters corresponding to the target psychoacoustic loudness of different users, so that the sound signal finally heard by the user conforms to the user's target psychoacoustic loudness and protects the user's hearing.

[0048] Specifically, for example, taking headphones as an example, in places such as the previously mentioned cinemas, KTVs, game arcades, etc., children can wear headphones. The headphones can receive sound signals, and then use a filter to perform frequency division processing on the sound signals to obtain low-frequency sound signals and high-frequency sound signals. Then, the steady-state sound signals and transient sound signals in the low-frequency sound signals and high-frequency sound signals are extracted. Among them, the steady-state sound signal represents a relatively stable and continuous sound signal, and the transient sound signal represents a short and impulsive sound signal. In some embodiments, for the steady-state sound signal, conventional processing can be performed, while for the transient sound signal, calibration is performed based on some of the personalized calibration parameters. In other embodiments, calibration can also be performed on both the steady-state sound signal and the transient sound signal based on the personalized calibration parameters, which is not limited here.

[0049] It can be understood that since the psychoacoustic loudness of different users is different. For example, the psychoacoustic loudness of children is 15 sone, and the psychoacoustic loudness of adults is 20 sone. Therefore, the personalized calibration parameters of different users can be determined based on the target psychoacoustic loudness of different users, so as to calibrate the frequency-divided sound signals based on the personalized calibration parameters. Then, the processed low-frequency sound signal and high-frequency sound signal after frequency division are synthesized to obtain the target pass-through signal, and the target pass-through signal is transmitted to the human ear through the headphones. The psychoacoustic loudness of the target pass-through signal is consistent with the target psychoacoustic loudness of the user. In this way, the hearing system of the user can be protected.

[0050] For example Figure 2 As shown, in the movie-watching scenario in the cinema, child A can watch movies by wearing wireless headphones 10 / 11. The wireless headphones 10 and the wireless headphones 11 can process the sounds in the received movie sounds that are not conducive to the hearing protection of children, so that the movie sounds heard by child A conform to the psychoacoustic loudness of children, thereby protecting the hearing of children.

[0051] It can be understood that both adults and children can protect their hearing by wearing wireless headphones in scenarios such as cinemas and noisy places.

[0052] It can be understood that the headphones worn by the user can include a child mode and an adult mode, and the audio processing method provided by the embodiments of the present application can be applied in both the child mode and the adult mode to achieve hearing protection.

[0053] To better understand the method provided by the embodiments of the present application, Figure 3 The working principle diagram of a wireless headphone 10 transmitting sound signals is shown.

[0054] As Figure 3As shown in the figure, the wireless earphone 10 may include a microphone 101, an amplifier (Proportional Amplifier, PA) 102, an analog-to-digital converter (ADC) 103, a digital signal processor (DSP) 104, and a speaker 105.

[0055] The working principle of the wireless earphone 10 for transmitting a sound signal to the human ear is as follows: After the microphone 101 receives a sound signal from the outside world, it converts the sound signal into an analog electrical signal and then sends the analog electrical signal to the amplifier 102; the amplifier 102 receives the analog electrical signal, amplifies it, and then sends it to the analog-to-digital converter 103; after receiving the amplified analog electrical signal, the analog-to-digital converter 103 converts it into a digital electrical signal and then sends the digital electrical signal to the digital signal processor 104. The digital signal processor 104 processes the received digital electrical signal and sends the processed digital electrical signal to the speaker 105; after receiving the processed digital electrical signal, the speaker 105 converts the processed digital electrical signal into a corresponding sound signal and transmits it to the human ear, enabling the user to hear the processed sound signal.

[0056] Further, referring to Figure 4 , Figure 4 shows a specific flowchart of an audio processing method. This method can be executed by an electronic device. For ease of understanding, in the embodiments of the present application, the electronic device is taken as the wireless earphone 10 as an example for description. The specific process of the audio processing method includes:

[0057] S201: Obtain a sound signal.

[0058] In some embodiments, the user wears the wireless earphone 10 in a noisy environment. As described above Figure 3 in, the microphone 101 of the wireless earphone 10 receives the sound signal, converts the sound signal into an analog electrical signal, and then sends it to the amplifier 102; the amplifier 102 receives the analog electrical signal, amplifies it, and then sends it to the analog-to-digital converter 103; after receiving the amplified analog electrical signal, the analog-to-digital converter 103 converts it into a digital electrical signal and then sends the digital electrical signal to the digital signal processor 104, and then the digital signal processor 104 obtains the digital electrical signal corresponding to the sound signal.

[0059] S202: Obtain the sound pressure level (SPL) of the sound signal.

[0060] In some embodiments, after the wireless earphone 10 obtains the sound signal, that is, after the digital signal processor 104 of the wireless earphone 10 obtains the digital electrical signal corresponding to the sound signal, the digital signal processor 104 can obtain the sound pressure level SPL of the sound signal based on it.

[0061] Specifically, since the relationship between the sound pressure level SPL of a sound signal and the sound pressure Pa is: Pa = Pr * 10^(SPL / 20), where Pr = 20 μPa; therefore, to obtain the SPL, the value of the sound pressure Pa needs to be calculated first. It can be understood that Pr can represent a reference sound pressure, which is the sound pressure of the lowest sound that the human ear can feel for a 1 kHz air sound.

[0062] Assume that the digital electrical signal corresponding to the sound signal is dBFS. Since dBFS is obtained after the digital signal processor 104 processes the digital electrical signal dBu, and 0 dBFS = x dBu. For example, if input_dBFS = 3 dBFS, then 3 dBFS = (x - 3) dBu; therefore, the digital electrical signal dBu can be obtained based on dBFS. Since the digital electrical signal dBu is obtained by the analog-to-digital converter 103 converting the analog electrical signal dBv, and dBu = dBv + 2.21, therefore, the analog electrical signal dBv can be obtained based on the digital electrical signal dBu, and the corresponding analog voltage value can be obtained. Assume the analog voltage value is (x) mv. The analog electrical signal dBv is obtained by the microphone 101 converting the sound signal into an analog electrical signal and then amplifying it through the amplifier 102. The gain of the amplifier is 20 * LOG10(output / input) dB. For example, the input of the amplifier 102 is 4 mv / pa, and the output is 361 mv / pa, then the gain is 20 * LOG10(316 / 4) dB; since the relationship between the sound pressure Pa and the analog electrical signal is Pa * (mv / Pa) = (x) mv = (x) * 10 -3 v, therefore, the sound pressure Pa can be obtained based on the analog voltage value corresponding to the analog electrical signal dBv; where, mv / Pa refers to the sensitivity of the microphone 101, and mv / Pa can be 12.87; therefore, the sound pressure Pa can be obtained based on the sensitivity mv / Pa and the analog voltage value corresponding to the analog electrical signal dBv. Finally, based on the relationship between the sound pressure Pa and the sound pressure level SPL, Pa = Pr * 10^(SPL / 20), the sound pressure level SPL of the sound signal can be obtained. Based on the above processing, the corresponding relationship between the SPL of the sound signal received by the wireless earphone 10 and its corresponding digital electrical signal can be obtained; for example, when the sound signal is x dBSPL, the corresponding digital electrical signal is y dBFS.

[0063] It can be understood that the sensitivity mv / Pa of the above microphone 101 being 12.87 is only an exemplary illustration, and the sensitivities of the microphones 101 of different electronic devices can be different, which are not limited herein.

[0064] It can be understood that the input when calculating the gain of the above amplifier 103 is 4 mv / Pa and the output is 316 mv / Pa is only an exemplary illustration. In some other embodiments, the gain of the amplifier 103 will change with different input and output of the amplifier, which is not limited herein.

[0065] S203: Perform Bark spectral analysis on the sound signal based on the SPL to obtain a Bark curve, and obtain the psychoacoustic loudness of the first target user for the sound signal based on the Bark curve.

[0066] It can be understood that the sound pressure level SPL can represent the strength of the sound signal, with the unit of decibel (dB). Therefore, the size of the sound signal can be described by the sound pressure level SPL. The Bark spectral analysis is actually based on the sound pressure level of the sound signal to analyze the frequency-domain signal corresponding to the sound signal to obtain a Bark curve. The psychoacoustic loudness of the user for the sound signal can be obtained based on the Bark curve.

[0067] In some embodiments, since the human ear's perception of sound is not linear, in order to obtain the psychological perception of the human ear for sound signals in different frequency ranges, the Bark spectral analysis can be performed on the sound signal based on the obtained sound pressure level SPL of the sound signal. The Bark spectral analysis can better reflect the perception sensitivity of the human ear's hearing to the frequencies of different sound signals and more accurately understand the perception effect of the human ear on sound signals in different frequency ranges. For example, if the current user of the earphone is the first target user, then by performing Bark spectral analysis on the sound signal to obtain a Bark curve, the psychoacoustic loudness of the first target user for the sound signal can be obtained.

[0068] The input of the Bark spectral analysis is the sound signal. When performing the Bark spectral analysis, the passive isolation degree can be filtered out. Filtering out the passive isolation degree is to make the result of the Bark spectral analysis more accurate. The output (result) of the Bark spectral analysis is the Bark curve corresponding to the sound signal. It can be understood that the psychoacoustic loudness corresponding to the sound signal can be directly obtained based on the Bark curve of the sound signal obtained by the Bark spectral analysis, where the psychoacoustic loudness refers to the subjective perception of the human ear for the sound signal.

[0069] It can be understood that the passive isolation degree refers to the physical isolation of the wireless earphone 10 from the sound signal. The passive isolation degree can be preset in the development and design stage of the wireless earphone 10, or obtained through real-time detection. It can be understood that for electronic devices that do not support detecting the passive isolation degree, the passive isolation degree preset in the development and design stage can be directly obtained without detection, which is not limited here. For example, if the passive isolation degree preset in the development and design stage of the wireless earphone 10 is (m)dB, then the preset passive isolation degree of the electronic device can be directly filtered out when performing Bark spectrum analysis; for another example, assuming that the sound pressure level of the sound signal obtained through the microphone 101 of the wireless earphone 10 is xdBSPL, and the sound pressure level of the processed sound signal heard at the eardrum of the human ear is ydBSPL, then the passive isolation degree corresponding to the wireless earphone 10 can be determined as (x - y)dB, and then the passive isolation degree (x - y) of the electronic device can be filtered out when performing Bark spectrum analysis.

[0070] It can be understood that Bark spectrum analysis is an analysis method (algorithm) commonly used in signal processing and psychoacoustic research. Bark spectrum analysis can convert the frequency-domain signal corresponding to the sound signal into a concept that is more in line with the auditory psychological perception of the human ear, that is, map the frequency-domain signal corresponding to the sound signal to the frequency band corresponding to the auditory psychological perception of the human ear to obtain the Bark curve corresponding to the sound signal that conforms to the user's psychological perception.

[0071] It can be understood that the psychoacoustic loudness can be obtained through the ISO 532-1 physiological acoustics model or the ISO 532-2 physiological acoustics model, or the psychoacoustic loudness can be obtained based on other physiological acoustics models, which is not limited here.

[0072] It can be understood that the process of obtaining the psychoacoustic loudness based on the ISO 532-1 physiological acoustics model or the ISO 532-2 physiological acoustics model can include: first, input the sound signal into the physiological acoustics model; then the physiological acoustics model performs filtering processing on the sound signal to obtain the sound energy close to the human ear perception curve, that is, the Bark curve. Further, the frequency-domain signal of the filtered sound signal is divided into a series of different frequency bands, and then the sound energy of each frequency band is weighted to obtain the loudness value corresponding to each frequency band. Furthermore, according to the weighted frequency band loudness values, the loudness values obtained from each frequency band can be synthesized into the final psychoacoustic loudness. Finally, the physiological acoustics model outputs a psychoacoustic loudness, which can be used to measure the subjective perception intensity of the user for the sound signal.

[0073] S204: When the psychoacoustic loudness of the sound signal for the first target user is inconsistent with the target psychoacoustic loudness of the first target user, perform frequency division on the sound signal to obtain the steady-state amplitude and transient amplitude of the sound signal.

[0074] It can be understood that the personalized calibration parameter of the first target user and the target psychoacoustic loudness of the first target user that can be obtained by adjusting the calibration parameter will be described below in the process of obtaining the personalized calibration parameter.

[0075] When the psychoacoustic loudness of the first target user for the sound signal is inconsistent with the target psychoacoustic loudness of the first target user, for example, the psychoacoustic loudness of the first target user for the sound signal is 50 sone and the target psychoacoustic loudness of the first target user is 30 sone, then the sound signal needs to be frequency-divided.

[0076] In some embodiments, refer to Figure 5 the schematic diagram of frequency-dividing the sound signal shown. As Figure 5 shown, the Linkwitz-Riley 4th-order (LR-4) filter can be used to frequency-divide the sound signal (i.e., the sound signal) to obtain a high-frequency sound signal hs(t) and a low-frequency sound signal Ls(t). Then, two-dimensional median filtering is performed on hs(t) and Ls(t) respectively. When performing two-dimensional median filtering on the low-frequency sound signal Ls(t), the low-frequency steady-state sound signal Ls_s(t) and the low-frequency transient sound signal Ls_i(t) in the low-frequency sound signal Ls(t) can be extracted; when performing two-dimensional median filtering on the high-frequency sound signal hs(t), the high-frequency steady-state sound signal hs_s(t) and the high-frequency transient sound signal hs_i(t) in the high-frequency sound signal hs(t) can be extracted.

[0077] Furthermore, equal-loudness weighting calculations are performed on the low-frequency transient sound signal Ls_i(t) and the high-frequency transient sound signal hs_i(t) to obtain the transient amplitude of the low-frequency transient sound signal Ls_i(t) and the transient amplitude of the high-frequency transient sound signal hs_i(t); equal-loudness weighting calculations and weighted amplitude calculations are performed on the low-frequency steady-state sound signal Ls_s(t) and the high-frequency steady-state sound signal hs_s(t) to obtain the transient amplitude of the low-frequency steady-state sound signal Ls_s(t) and the transient amplitude of the high-frequency steady-state sound signal hs_s(t). For example, the weight of the low-frequency transient sound signal Ls_i(t) is W1, the weight of the high-frequency transient sound signal hs_i(t) is W2, the sum of W1 and W2 is 1, and then the weighted amplitude calculation is performed based on W1 and W2 to obtain the transient amplitude corresponding to the transient sound signal; the weight of the low-frequency steady-state sound signal Ls_s(t) is W3, the weight of the high-frequency steady-state sound signal hs_s(t) is W4, the sum of W3 and W4 is 1. Then the weighted amplitude calculation is performed based on W3 and W4 to obtain the steady-state amplitude of the steady-state sound signal.

[0078] It can be understood that the cut-off frequency when using the LR-4 filter for frequency division can be the cut-off frequency fc of the speaker, which is not limited herein.

[0079] It can be understood that since users of different ages have different target psychoacoustic loudness for sound signals, users can be divided based on age, and the personalized calibration parameters and the target psychoacoustic loudness of users in different age groups are different.

[0080] It can be understood that when a pair of headphones has the personalized calibration parameters of two users (such as users of different ages or different age groups) tested and entered, the electronic device communicating with the headphones can, in response to the selection operation of the user based on the display interface of the electronic device, determine that the first target user is the one currently using the headphones, and send the personalized calibration parameters corresponding to the first target user to the headphones; and then the headphones can calibrate the sound signals received by the headphones based on the personalized calibration parameters corresponding to the first target user, so that the psychoacoustic loudness of the calibrated sound signals finally heard by the first target user is consistent with the target psychoacoustic loudness of the first target user.

[0081] The process of obtaining the personalized calibration parameters of the user will be described below. It can be understood that the user can be the first target user or other target users.

[0082] In some embodiments, in order to calibrate the sound signal so that the psychoacoustic loudness corresponding to the calibrated sound signal conforms to the target psychoacoustic loudness of the user, it is first necessary to obtain the personalized calibration parameters that can calibrate the sound signal, so that after calibrating the sound signal based on the personalized calibration parameters, the psychoacoustic loudness corresponding to the calibrated sound signal is consistent with the target psychoacoustic loudness of the user.

[0083] Since the sound signals in different environments are different, in order to obtain the personalized calibration parameters, the wireless headphones 10 can be used to play a preset calibration signal (the calibration signal is also a kind of sound signal) that can simulate various sound signals for the user. When the user wears the wireless headphones 10, the user can start the calibration based on the personalized calibration interface of the electronic device that can set the wireless headphones 10.

[0084] For example Figure 6 , Figure 6 shows a schematic diagram of a personalized calibration interface 300. It can be seen that the personalized calibration interface includes time-domain adjustment and frequency-domain adjustment. When the user clicks "Calibration Start" in the personalized calibration interface 300, the wireless headphones 10 will play a calibration signal that can simulate the sound signal, and the user will perform time-domain adjustment and / or frequency-domain adjustment on the calibration signal based on the heard calibration signal, so as to obtain the corresponding time-domain calibration parameters and / or frequency-domain calibration parameters.

[0085] Performing time-domain adjustment on the calibration signal can adjust the impact feeling and psychoacoustic loudness of the calibration signal, so that the adjusted calibration signal meets the user's target psychoacoustic loudness.

[0086] Reference Figure 6 , the time-domain calibration parameters for time-domain adjustment can include "Attack", "Decay", "Sustain", "Release", and "peak amplitude". Among them, "Attack" can be abbreviated as "A", which describes the impulse-like increase in the amplitude of the calibration signal at 0 - t1; "Decay" can be abbreviated as "D", which describes the decrease in the amplitude of the calibration signal at t1 - t2; "Sustain" can be abbreviated as "S", which describes the stable state of the amplitude of the calibration signal at t2 - t3; "Release" can be abbreviated as "R", which describes the state where the amplitude of the calibration signal gradually becomes zero at t3 - t4, that is, the calibration signal played by the wireless earphone 10 gradually disappears; the "peak amplitude" is also the psychoacoustic loudness. After the user hears the calibration signal, based on the user's psychological perception of the calibration signal, the user can slide to adjust the above time-domain parameters "A", "D", "S", "R" to adjust the impact feeling of the calibration signal and adjust the "peak amplitude". For example, if the user adjusts the peak amplitude of the calibration signal to 65 dB, it means that the maximum psychoacoustic loudness acceptable to the user is 65 dB. Furthermore, it can make the calibration signal after time-domain adjustment meet the user's target psychoacoustic loudness and can reduce the impact feeling of the calibration signal. Based on this, the time-domain calibration parameters in the personalized calibration parameters can be obtained.

[0087] It can be understood that the time-domain calibration parameters can be used as the time-domain compression parameters of the pass-through compressor in step S208 above, so that the pass-through compressor can calibrate the transient sound signal.

[0088] Performing frequency-domain adjustment on the calibration signal can adjust the loudness and timbre of the calibration signal, so that the adjusted calibration signal meets the loudness and timbre of the sound signal that the user psychologically perceives.

[0089] Continue to refer to Figure 6 , the frequency-domain calibration parameters for frequency-domain adjustment can include the frequency response (Equaliser, EQ), such as Figure 6 the frequency response curve under frequency-domain adjustment. The frequency response can reflect the amplitude of the calibration signal at different frequencies. After the user hears the calibration signal, based on the user's psychological perception of the calibration signal, drag and adjust each frequency point in the frequency response curve to adjust the amplitude of different frequency points in the calibration signal. The frequency-domain adjustment can change the loudness and timbre of the calibration signal to meet the loudness and timbre of the calibration signal required by the user. Based on this, the frequency-domain calibration parameters in the personalized calibration parameters can be obtained.

[0090] After obtaining the personalized calibration parameters, the user can end the adjustment of the calibration signal by clicking "Calibration Stop" in the personalized calibration interface 300, and then the wireless earphone 10 stops playing the calibration signal.

[0091] The wireless earphone 10 plays a calibration signal that can simulate various sound signals, allowing the user to adjust the calibration signal in the time domain and / or frequency domain to obtain corresponding time-domain calibration parameters and / or frequency-domain calibration parameters. The time-domain calibration parameters and frequency-domain calibration parameters are the personalized adjustment parameters. Furthermore, when receiving a sound signal, the wireless earphone 10 can calibrate the sound signal based on the personalized adjustment parameters, so that after the calibrated sound signal is heard by the user, it conforms to the psychoacoustic loudness acceptable to the user, achieving hearing protection for the user.

[0092] It can be understood that the user can choose to perform left-ear calibration, right-ear calibration, or simultaneous calibration of both ears based on the personalized calibration interface 300, which is not limited here.

[0093] It can be understood that the calibration signal is the acquisition of real signals that are likely to have a psychological impact on children or adults in various living environments and some movie sound clips, which is not limited here.

[0094] It can be understood that the personalized calibration is a pre-calibration process. The purpose of the personalized calibration is to obtain personalized calibration parameters that can calibrate the sound signal to conform to the psychoacoustic loudness of the user. The personalized calibration can be performed once or multiple times, which is not limited here.

[0095] S205: Determine whether the steady-state amplitude is greater than the first amplitude value Th_s.

[0096] In some embodiments, it is determined whether the steady-state amplitude corresponding to the obtained steady-state sound signal is greater than the first amplitude value Th_s. If the steady-state amplitude corresponding to the steady-state sound signal is greater than the first amplitude value Th_s, go to step S206, that is, calibrate the steady-state sound signal based on the personalized calibration parameters. Otherwise, go to step S209, that is, use the signal synthesis filter to synthesize the steady-state sound signal and the transient sound signal to obtain a target pass-through signal that conforms to the user's target psychoacoustic loudness, that is, use the signal synthesis filter to synthesize the steady-state sound signal with a steady-state amplitude less than or equal to the first amplitude value Th_s and the transient sound signal.

[0097] S206: Obtain the personalized calibration parameters corresponding to the first target user, and calibrate the steady-state sound signal based on the personalized calibration parameters corresponding to the first target user.

[0098] Based on the above Figure 6The method can obtain the personalized calibration parameters corresponding to the first target user, and the personalized calibration parameters corresponding to the first target user include time-domain calibration parameters and frequency-domain calibration parameters.

[0099] When it is determined that the steady-state amplitude corresponding to the steady-state sound signal is greater than the first amplitude value Th_s, the steady-state sound signal can be processed by using a filter based on the personalized calibration parameters corresponding to the first target user. That is to say, the frequency-domain calibration parameter in the personalized calibration parameters corresponding to the first target user can be used as the filtering coefficient of the filter to calibrate the steady-state sound signal. For example, the loudness of the steady-state sound signal can be suppressed based on the frequency response (Equaliser, EQ) of the frequency-domain calibration parameter.

[0100] It can be understood that the first amplitude value Th_s is a preset steady-state amplitude threshold.

[0101] In some other embodiments, the frequency-domain calibration parameter in the personalized calibration parameters corresponding to the first target user can also be used as the filtering coefficient of the filter to calibrate the transient sound signal, which is not limited here.

[0102] S207: Determine whether the transient amplitude is greater than the second amplitude value Th_i.

[0103] In some embodiments, it is determined whether the transient amplitude corresponding to the obtained transient sound signal is greater than the second amplitude value Th_i. If the transient amplitude corresponding to the transient sound signal is greater than the second amplitude value Th_i, go to step S208, that is, calibrate the transient sound signal based on the personalized calibration parameters. Otherwise, go to step S209, that is, use a signal synthesis filter to synthesize the steady-state sound signal and the transient sound signal to obtain a target passthrough signal that meets the user's target psychoacoustic loudness. That is, use a signal synthesis filter to synthesize the transient sound signal with a transient amplitude less than or equal to the second amplitude value Th_i and the steady-state sound signal.

[0104] S208: Obtain the personalized calibration parameters corresponding to the first target user, and calibrate the transient sound signal based on the personalized calibration parameters corresponding to the first target user.

[0105] When it is determined that the transient amplitude corresponding to the transient sound signal is greater than the second amplitude value Th_i, the transient sound signal can be processed by using a passthrough compressor based on the personalized calibration parameters corresponding to the first target user. That is to say, the time-domain calibration parameter in the personalized calibration parameters corresponding to the first target user can be used as the time-domain compression coefficient of the passthrough compressor to suppress (compress) the transient sound signal.

[0106] It can be understood that the second amplitude value Th_i is a preset transient amplitude threshold.

[0107] In some other embodiments, the time-domain calibration parameter in the personalized calibration parameter corresponding to the first target user may also be used as the time-domain compression coefficient of the transparent compressor to calibrate the steady-state sound signal, which is not limited herein.

[0108] S209: Synthesize the steady-state sound signal and the transient sound signal by using a signal synthesis filter to obtain a target transparent signal, and the psychoacoustic loudness corresponding to the target transparent signal is consistent with the target psychoacoustic loudness corresponding to the first target user.

[0109] In some embodiments, the processed steady-state sound signal and transient sound signal are synthesized by using a synthesis filter to generate a target transparent signal HT_target(t). The target transparent signal refers to the processed sound signal that is finally transmitted to the eardrum of the human ear through the speaker 105, and the psychoacoustic loudness corresponding to the target transparent signal is consistent with the target psychoacoustic loudness corresponding to the first target user, or the difference between the psychoacoustic loudness corresponding to the target transparent signal and the target psychoacoustic loudness corresponding to the first target user is within a preset range.

[0110] It can be understood that in some embodiments, an inverse transparent signal played by the speaker 105 can be obtained based on the sound signal and the target transparent signal HT_target(t). Based on this, the sound signal received by the wireless earphone 10 can be calibrated by using the inverse transparent signal, that is, part of the noise in the sound signal is cancelled, so that the psychoacoustic loudness corresponding to the target transparent signal finally reaching the eardrum of the human ear is consistent with the target psychoacoustic loudness of the user, thereby realizing the protection of the user's hearing.

[0111] For ease of understanding, Figure 7 A schematic diagram showing a speaker playing an inverse transparent signal SPK(t) is shown.

[0112] As Figure 7 shown in, assume that the target transparent signal is HT_target(t), the sound signal is S(t), and the inverse transparent signal is SPK(t). Among them, the target transparent signal is the signal that is finally heard by the human ear and conforms to the target psychoacoustic loudness of the user; the inverse transparent signal SPK(t) refers to the signal required to cancel part of the noise in the sound signal S(t) in order to obtain the target transparent signal HT_target(t). Then the relationship between the target transparent signal HT_target(t), the sound signal S(t), and the inverse transparent signal SPK(t) can refer to formula (1).

[0113] S(s) ·H1(s) ·SPK(s) ·H2(s)= HT_target(s) (1)

[0114] In Equation (1), S(s)·H1(s)·SPK(s)·H2(s) describes the target pass-through signal HT_target(s) that reaches the eardrum of the human ear after being calibrated by the transfer function H1(s) of the sound signal passing through the microphone 101, the inverted pass-through signal SPK(s) played by the speaker 105, and the transfer function H2(s) of the speaker 105.

[0115] Furthermore, the inverted pass-through signal SPK(t) can be derived from Equation (1), and the inverted pass-through signal SPK(t) can be obtained based on Equation (2).

[0116]

[0117] Next, based on Figure 8 the schematic diagram showing the transfer of the calibration signal by the speaker to the eardrum of the human ear, the principle of obtaining personalized calibration parameters based on the calibration signal that can simulate the sound signal will be described.

[0118] As Figure 8 shown, the wireless earphone 10 plays the calibration signal through the speaker 105. In order to make the calibration signal played through the speaker 105 close to the sound signal and better simulate the sound signal, the calibration signal needs to be convolved with a transfer function Hinv(s). Therefore, the transfer function Hinv(s) needs to be obtained so that the calibration signal can better simulate the sound signal by convolving with the transfer function Hinv(s).

[0119] Assume that the sound signal is s(t), the sound signal directly reaching the eardrum of the human ear is input(t); assume that the calibration signal that can simulate the sound signal is spk(t). After the user wears the wireless earphone 10, the calibration signal spk(t) is played through the speaker 105, and the microphone 106 receives the calibration signal spk(t) to simulate the sound signal s(t) received by the eardrum of the human ear. It can be understood that the microphone 106 is used to simulate the eardrum of the human ear receiving the sound.

[0120] In order to make the calibration signal spk(t) simulate the sound signal s(t), it is necessary to achieve: the transfer function H1 estimated from the sound signal s(t) and the input signal directly reaching the eardrum of the human ear is equal to the transfer function H2 of the calibration signal spk(t) reaching the microphone 106 (i.e., reaching the eardrum of the human ear), that is, referring to Equation (3) and Equation (4).

[0121] S(s) ·H(s)=SPK(s) ·H2(s) ·Hinv(s) (3)

[0122] S(s) ·H1(s) ·H3(s)= SPK(s) ·H2(s) ·Hinv(s) (4)

[0123] In Equation (3), S(s) represents the sound signal, H(s) is the transfer function of S(s), and H(s) is used to convert S(s) into the sound signal that directly reaches the eardrum of the human ear. S(s)·H(s) describes the sound signal heard by the eardrum of the human ear. SPK(s) represents the calibration signal played by the speaker 105, H2(s) is the transfer function of SPK(s), and H2(s) is used to convert the calibration signal SPK(s) into the sound signal heard by the eardrum of the human ear; Hinv(s) is the transfer function Hinv(s) that needs to be convolved to enable the calibration signal to simulate the sound signal.

[0124] In Equation (4), H1(s)·H3(s) is equal to H(s), and both H3(s) and H1(s) are transfer functions of S(s). Among them, H3(s) is used to convert S(s) into the sound signal received by the microphone 101, and H1(s) is used to convert S(s) into the sound signal received by the microphone 106 (i.e., the sound signal received by the eardrum of the human ear).

[0125] It can be understood that the transfer function H1(s) can be obtained based on Equation (5); H2(s) can be obtained based on Equation (6); H3(s) can be obtained based on Equation (7).

[0126] It can be understood that the tfestimate_() function in Equations (5)-(7) is a tool in MATLAB for estimating the transfer function.

[0127] H3(s) = tfestimate(s,mic1) (5)

[0128] H2(s) = tfestimate(spk,mic2) (6)

[0129] H1(s) = tfestimate(mic1,mic2) (7)

[0130] Furthermore, assume that the sound signal S(s) is the same as the calibration signal SPK(s) played by the speaker 105, that is, SPK(s) = S(s). Then the above Equation (4) can be transformed into Equation (8). That is

[0131] S(s)·H1(s)·H3(s) = S(s)·H2(s)·Hinv(s) (8)

[0132] Based on this, the transfer function Hinv(s) can be calculated, referring to Equation (9).

[0133]

[0134] Based on the above derivation and calculation, the transfer function Hinv(s) of the convolution required to enable the calibration signal to simulate various sound signals can be obtained. Therefore, when simulating various sound signals based on the calibration signal played by the speaker 105, it can be directly achieved by convolving the calibration signal with the transfer function Hinv(s).

[0135] Based on the above, when it is necessary to perform calibration processing on the sound signal, the user can receive the sound signal by wearing the wireless earphone 10, and then play the inverted passthrough signal through the speaker of the wireless earphone 10 to cancel part of the noise of the sound signal, so that the sound signal finally heard by the user (i.e., the target passthrough signal) conforms to the psychoacoustic loudness acceptable to the user, realizing the protection of the user's hearing.

[0136] Through the method provided by the embodiments of the present application, the user can receive the sound signal by wearing the wireless earphone 10, and then process the sound signal based on the user's personalized calibration parameters through the wireless earphone 10 to obtain the target passthrough signal and transmit it to the human ear. Moreover, the psychoacoustic loudness corresponding to the target passthrough signal is consistent with the user's target psychoacoustic loudness, thereby realizing the protection of the user's hearing.

[0137] It can be understood that the audio processing method provided in the embodiments of the present application can be applied to an electronic device with a sound passthrough function. The electronic device can customize and control the activation of the passthrough function according to a specific scenario, and the closing time of the passthrough function can depend on the duration of the received external sound signal, which is not limited herein.

[0138] The embodiments of the mechanism disclosed in the present application can be implemented in hardware, software, firmware, or a combination of these implementation methods. The embodiments of the present application can be implemented as a computer program or program code executed on a programmable system, which includes at least one processor, a storage system (including volatile and non-volatile memories and / or storage elements), at least one input device, and at least one output device.

[0139] The program code can be applied to the input instructions to execute the various functions described in the present application and generate output information. The output information can be applied to one or more output devices in a known manner. For the purpose of the present application, the processing system includes any system having a processor such as, for example, a digital signal processor (DSP), a microcontroller, an application specific integrated circuit (ASIC), or a microprocessor.

[0140] The program code can be implemented in a high-level procedural language or an object-oriented programming language to communicate with the processing system. When necessary, the program code can also be implemented in assembly language or machine language. In fact, the mechanism described in the present application is not limited to the scope of any specific programming language. In any case, the language can be a compiled language or an interpreted language.

[0141] In some cases, the disclosed embodiments may be implemented in hardware, firmware, software, or any combination thereof. The disclosed embodiments may also be implemented as instructions carried or stored on one or more transitory or non-transitory machine-readable (e.g., computer-readable) storage media, which may be read and executed by one or more processors. For example, the instructions may be distributed via a network or via other computer-readable media. Accordingly, machine-readable media may include any mechanism for storing or transmitting information in a form readable by a machine (e.g., a computer), including but not limited to, floppy disks, optical disks, optical discs, compact disc read-only memories (CD-ROMs), magneto-optical discs, read-only memories (ROMs), random access memories (RAMs), erasable programmable read-only memories (EPROMs), electrically erasable programmable read-only memories (EEPROMs), magnetic or optical cards, flash memory, or tangible machine-readable memories for transmitting information using propagated signals in electrical, optical, acoustic, or other forms via the Internet. Accordingly, machine-readable media include any type of machine-readable media suitable for storing or transmitting electronic instructions or information in a form readable by a machine (e.g., a computer).

[0142] In the accompanying drawings, some structural or method features may be shown in a particular arrangement and / or order. However, it should be understood that such a particular arrangement and / or ordering may not be required. Rather, in some embodiments, these features may be arranged in a manner and / or order different from that shown in the illustrative drawings. Additionally, the inclusion of a structural or method feature in a particular figure does not imply that such a feature is required in all embodiments, and in some embodiments, such a feature may not be included or may be combined with other features.

[0143] It should be noted that each unit / module mentioned in the device embodiments of the present application is a logical unit / module. Physically, a logical unit / module may be a physical unit / module, a part of a physical unit / module, or may be implemented as a combination of multiple physical units / module. The physical implementation manner of these logical units / module themselves is not the most important. The combination of the functions implemented by these logical units / module is the key to solving the technical problems proposed by the present application. In addition, in order to highlight the innovative part of the present application, the above device embodiments of the present application do not introduce units / modules that are not closely related to solving the technical problems proposed by the present application. This does not mean that there are no other units / modules in the above device embodiments.

[0144] It should be noted that in the examples and description of this patent, relational terms such as first and second are only used to distinguish one entity or operation from another entity or operation, and do not necessarily require or imply any actual relationship or order between these entities or operations. Moreover, the term "comprising" or any other variant thereof is intended to cover non-exclusive inclusion, such that a process, method, article, or device comprising a series of elements includes not only those elements but also other elements not expressly listed, or elements inherent to such process, method, article, or device. Without further limitation, an element defined by the statement "comprising one" does not exclude the presence of additional identical elements in the process, method, article, or device comprising the element.

[0145] Although this application has been illustrated and described by reference to certain preferred embodiments thereof, those of ordinary skill in the art should understand that various changes in form and detail may be made therein without departing from the spirit and scope of this application.

Claims

1. An audio processing method, characterized in that, applied to headphones, the method includes; Receiving a first sound signal and obtaining a first sound pressure level of the first sound signal; Based on the first sound pressure level, obtaining a first psychoacoustic loudness of a first target user for the first sound signal; Based on the inconsistency between the first psychoacoustic loudness and the target psychoacoustic loudness of the first target user, obtaining a first calibration parameter corresponding to the first target user to calibrate the first sound signal to obtain a second sound signal, wherein, the second psychoacoustic loudness of the first target user for the second sound signal is consistent with the target psychoacoustic loudness.

2. The method according to claim 1, characterized in that, The obtaining the first psychoacoustic loudness of the first target user for the first sound signal based on the first sound pressure level includes: Obtaining a first passive isolation degree of the headphones, and performing Bark spectrum analysis on the first sound signal based on the first passive isolation degree and the first sound pressure level to obtain the first psychoacoustic loudness of the first target user for the first sound signal.

3. The method according to claim 1, characterized in that, The obtaining of the first calibration parameter corresponding to the first target user is determined by the following method: The headphones play a first test tone to the first target user; In response to the operation of the first target user adjusting the first test tone to a second test tone; Obtaining parameter values of a time-domain calibration parameter and parameter values of a frequency-domain calibration parameter to obtain the first calibration parameter, wherein the first calibration parameter includes a first time-domain calibration parameter and a first frequency-domain calibration parameter.

4. The method according to claim 1, characterized in that, The obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal includes: Performing frequency division processing on the first sound signal to obtain a steady-state amplitude of the first sound signal; Judging whether the steady-state amplitude of the first sound signal is greater than a first amplitude threshold; Corresponding to the steady-state amplitude of the first sound signal being greater than the first amplitude threshold, calibrating the first sound signal based on the first calibration parameter.

5. The method according to claim 4, characterized in that, The obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal further includes: Performing frequency division processing on the first sound signal to obtain a transient amplitude of the first sound signal; Judging whether the transient amplitude of the first sound signal is greater than a second amplitude threshold; Corresponding to the transient amplitude of the first sound signal being greater than the second amplitude threshold, calibrating the first sound signal based on the first calibration parameter.

6. The method according to claim 4, characterized in that, The performing frequency division processing on the first sound signal to obtain a steady-state amplitude of the first sound signal includes: Using a filter to perform frequency division on the first sound signal to obtain a first high-frequency sound signal and a first low-frequency sound signal; Perform two-dimensional median filtering on the first high-frequency sound signal to obtain the first high-frequency steady-state sound signal of the first high-frequency sound signal; Perform two-dimensional median filtering on the first low-frequency sound signal to obtain the first low-frequency steady-state sound signal of the first low-frequency sound signal; Perform equal-loudness weighting calculation and weighted amplitude calculation on the amplitude of the first high-frequency steady-state sound signal and the amplitude of the first low-frequency steady-state sound signal to obtain the steady-state amplitude of the first sound signal.

7. The method according to claim 5, wherein, The performing frequency division processing on the first sound signal to obtain the transient amplitude of the first sound signal includes: Using a filter to perform frequency division on the first sound signal to obtain a first high-frequency sound signal and a first low-frequency sound signal; Perform two-dimensional median filtering on the first high-frequency sound signal to obtain the first high-frequency transient sound signal of the first high-frequency sound signal; Perform two-dimensional median filtering on the first low-frequency sound signal to obtain the first low-frequency transient sound signal of the first low-frequency sound signal; Perform equal-loudness weighting calculation and weighted amplitude calculation on the amplitude of the first high-frequency transient sound signal and the amplitude of the first low-frequency transient sound signal to obtain the transient amplitude of the first sound signal.

8. The method according to claim 4, wherein, The corresponding steady-state amplitude of the first sound signal is greater than the first amplitude threshold, and calibrating the first sound signal based on the first calibration parameter includes: Taking the first frequency-domain calibration parameter as the filtering coefficient of the filter, and calibrating the steady-state sound signal of the first sound signal based on the filter to obtain the calibrated steady-state sound signal of the first sound signal.

9. The method according to claim 5, wherein, The corresponding transient amplitude of the first sound signal is greater than the second amplitude threshold, and calibrating the first sound signal based on the first calibration parameter includes: Taking the first time-domain calibration parameter as the time-domain compression parameter of the pass-through compressor, and calibrating the transient sound signal of the first sound signal based on the pass-through compressor to obtain the calibrated transient sound signal of the first sound signal.

10. The method according to claim 1, wherein, The obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal to obtain a second sound signal includes: Using a synthesis filter to synthesize the steady-state sound signal of the calibrated first sound signal and the transient sound signal of the calibrated first sound signal to obtain the second sound signal.

11. The method according to claim 1, wherein, The first calibration parameter is a calibration parameter corresponding to children.

12. A headset, wherein, The headset includes: a memory for storing instructions; a processor for executing the instructions to implement the audio processing method according to any one of claims 1-11.

13. A readable storage medium, wherein, Instructions are stored on the readable medium, and when the instructions are executed on the earphone, the audio processing method described in any one of claims 1-11 is implemented.

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