An audio processing method, electronic device and readable storage medium

Abstract

The application relates to the technical field of audio processing, in particular to an audio processing method, an electronic device and a readable storage medium. In the method, a sound signal is received by a device that can be worn on the ear of a user, then the sound signal is divided into a low-frequency sound signal and a high-frequency sound signal by using a filter, 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. The steady-state sound signal and the transient sound signal are processed based on the individual calibration parameters corresponding to the user, so that the calibrated sound signal finally heard by the user conforms to the target psychoacoustic loudness of the user. In this way, the hearing protection of the user can be realized.

Description

Technical Field

[0001] This application relates to the field of audio processing technology, and in particular to an audio processing method, an electronic device, and a readable storage medium. Background Technology

[0002] Current research indicates that children's brains process sound differently than adults'. Children's auditory systems are not fully developed, and their auditory nerves and auditory cortex have weaker sound processing capabilities, making them more susceptible to sound interference and stimulation. Therefore, excessively high ambient sound pressure levels (e.g., exceeding 70 decibels) and sudden sounds can damage children's hearing. Furthermore, certain environments with high levels of ambient noise can also negatively impact children's hearing. For example, cinemas, karaoke bars, and arcades often have significant ambient noise levels, and prolonged exposure to such environments can affect children's hearing. Summary of the Invention

[0003] To address the aforementioned issues, embodiments of this application provide an audio processing method, an electronic device, and a readable storage medium.

[0004] In a first aspect, embodiments of this application provide an audio processing method applied to headphones, the method comprising: receiving a first sound signal and obtaining a first sound pressure level of the first sound signal; obtaining a first psychoacoustic loudness of the first sound signal for a first target user based on the first sound pressure level; and calibrating the first sound signal by obtaining a first calibration parameter corresponding to the first target user based on the inconsistency between the first psychoacoustic loudness and the target psychoacoustic loudness of the first target user, thereby obtaining a second sound signal, wherein the second psychoacoustic loudness of the second sound signal for the first target user is consistent with the target psychoacoustic loudness.

[0005] It is understood that the first sound signal may refer to the sound signal received by the headphone microphone mentioned in the embodiments of this application, and the second sound signal may refer to the target transparent signal that ultimately reaches the eardrum mentioned in the embodiments of this application.

[0006] Based on the above scheme, users can receive sound signals by wearing wireless earphones 10. The earphones 10 then calibrate the sound signals based on the user's personalized calibration parameters to obtain a target transparent signal that is transmitted to the user's ear. Furthermore, the psychoacoustic loudness corresponding to the target transparent signal is consistent with the user's target psychoacoustic loudness, thereby protecting the user's hearing.

[0007] In one possible implementation of the first aspect above, 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 of the headphones, performing Bark spectrum analysis on the first sound signal based on the first passive isolation and the first sound pressure level, and obtaining the first psychoacoustic loudness of the first target user for the first sound signal.

[0008] It is understandable that the first passive isolation refers to the physical isolation that the headphones provide to the sound signal. The passive isolation can be preset during the development and design stage of the headphones, or it can be obtained through real-time detection. For headphones that do not support the detection of passive isolation, the passive isolation preset during the development and design stage can be obtained directly.

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

[0010] It is understandable that the input to Bark spectral analysis is a sound signal. During Bark spectral analysis, the first passive isolation can be filtered out, thereby making the results of Bark spectral analysis more accurate. The output (result) of Bark spectral analysis is the Bark curve corresponding to the sound signal.

[0011] It is understandable that the Bark curve of the sound signal obtained based on Bark spectrum analysis can directly yield the psychoacoustic loudness of the sound signal, which refers to the subjective perception of the sound signal by the human ear.

[0012] In one possible implementation of the first aspect described above, 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 first target user adjusting the first test tone to a second test tone; the parameter values ​​of the time-domain calibration parameter and the parameter values ​​of the frequency-domain calibration parameter are obtained 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.

[0013] It can be understood that responding to the first target user's operation of adjusting the first test tone to the second test tone can refer to the personalized calibration interface mentioned in the embodiments of this application, where the personalized calibration interface is clicked, slid, or dragged to obtain time-domain calibration parameters and frequency-domain calibration parameters.

[0014] It is 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 this application; the first time-domain calibration parameter may refer to "Attack", "Decay", "Sustain", "Release" and "Peak Amplitude" mentioned in the embodiments of this application. The first frequency-domain calibration parameter may refer to the frequency response mentioned in the embodiments of this application.

[0015] In one possible implementation of the first aspect above, calibrating the first sound signal by obtaining the first calibration parameters corresponding to the first target user 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 a first amplitude threshold; and calibrating the first sound signal based on the first calibration parameters corresponding to the steady-state amplitude of the first sound signal being greater than the first amplitude threshold.

[0016] In one possible implementation of the first aspect described 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 a second amplitude threshold; and calibrating the first sound signal based on the first calibration parameter corresponding to the transient amplitude of the first sound signal being greater than the second amplitude threshold.

[0017] In one possible implementation of the first aspect described above, frequency division processing of the first sound signal to obtain the steady-state amplitude of the first sound signal includes: frequency division of the first sound signal using a filter to obtain a first high-frequency sound signal and a first low-frequency sound signal; two-dimensional median filtering of the first high-frequency sound signal to obtain a first high-frequency steady-state sound signal of the first high-frequency sound signal; two-dimensional median filtering of the first low-frequency sound signal to obtain a first low-frequency steady-state sound signal of the first low-frequency sound signal; and equal-loudness weighting calculation and weighted amplitude calculation of 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 one possible implementation of the first aspect described above, frequency division processing of the first sound signal to obtain the transient amplitude of the first sound signal includes: frequency division of the first sound signal using a filter to obtain a first high-frequency sound signal and a first low-frequency sound signal; two-dimensional median filtering of the first high-frequency sound signal to obtain a first high-frequency transient sound signal of the first high-frequency sound signal; two-dimensional median filtering of the first low-frequency sound signal to obtain a first low-frequency transient sound signal of the first low-frequency sound signal; and equal-loudness weighting calculation and weighted amplitude calculation of 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 one possible implementation of the first aspect above, corresponding to the steady-state amplitude of the first sound signal being greater than a first amplitude threshold, the first sound signal is calibrated based on the first calibration parameter, which includes: using the first frequency domain calibration parameter as the filter coefficient of the filter, calibrating the steady-state sound signal of the first sound signal based on the filter, and obtaining the calibrated steady-state sound signal of the first sound signal.

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

[0021] In one possible implementation of the first aspect above, obtaining the first calibration parameters corresponding to the first target user and calibrating the first sound signal to obtain the 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 one possible implementation of the first aspect described above, the first calibration parameter is the calibration parameter corresponding to the child.

[0023] Secondly, embodiments of this application provide an earphone, which includes: a memory for storing instructions; and a processor for executing the instructions to implement the first aspect and any possible implementation of the audio processing method provided by the first aspect.

[0024] Thirdly, embodiments of this application provide a readable storage medium storing instructions that, when executed on an electronic device, cause the headphones to perform the first aspect and any possible implementation of the provided audio processing method. Attached Figure Description

[0025] Figure 1 According to some embodiments of this application, a schematic diagram of a user watching a movie in a cinema is shown;

[0026] Figure 2 According to some embodiments of this application, a schematic diagram of a scenario in which a child wears wireless headphones while watching a movie in a cinema is shown;

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

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

[0029] Figure 5 According to some embodiments of this application, a schematic diagram of frequency division of an audio signal is shown;

[0030] Figure 6 A schematic diagram of a personalized calibration interface 300 is shown according to some embodiments of this application;

[0031] Figure 7 According to some embodiments of this application, a schematic diagram of a loudspeaker playing an inverted transparent signal SPK(t) is shown;

[0032] Figure 8 According to some embodiments of this application, a schematic diagram is shown of a loudspeaker transmitting a calibration signal to the eardrum. Detailed Implementation

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

[0034] It is understood that the electronic devices provided in the embodiments of this application may be various headphones, such as wireless headphones, wired headphones, and augmented reality (AR) devices, virtual reality (VR) devices, etc. that support pass-through function, or electronic devices including pass-through mode, which are not limited here.

[0035] It is understandable that the pass-through function means that when the electronic device is in pass-through mode, it will not completely block out external sounds, so that users can hear the sounds of the outside environment while wearing the electronic device.

[0036] It is understood that the electronic device provided in this application embodiment can be either a noise-canceling headphone or a regular headphone, and is not limited thereto. To facilitate understanding of the audio processing method provided in this application embodiment, a wireless headphone is used as an example for illustration.

[0037] To facilitate understanding of the audio processing methods provided in the embodiments of this application, the relevant terms involved in the embodiments of this application will be introduced first.

[0038] (1) Sound pressure level

[0039] Sound pressure level (SPL), also known as sound pressure level, is measured in Pa.

[0040] (2) Bark spectral analysis

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

[0042] Bark spectral analysis involves dividing a sound signal into a series of uniformly distributed frequency bands. These bands map the signal's frequencies onto 24 psychoacoustic critical bands. The human ear can resonate with frequencies within these 24 critical bands, and the width of a critical band is equal to one bark. Simply put, Bark spectral analysis converts (maps) physical frequencies to psychoacoustic frequencies to obtain the human ear's perception curves of sound signals for 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 based on the human ear's perception curve and accurately reflects the subjective perception of sound. For example, the psychoacoustic loudness of a child is 15 sones, and that of an adult is 20 sones.

[0045] The technical solution of this application is described below with reference to the accompanying drawings.

[0046] As mentioned earlier, environmental noise in some places can damage children's hearing systems. For example, Figure 1 As shown, in a movie theater setting, child A watches a film with adults B and C. It's understandable that to provide a good viewing experience, movie theater sound is generally amplified, making the sound louder (e.g., 100 decibels) to ensure an immersive experience for all viewers. However, children's hearing systems are not fully developed, resulting in weaker sound processing capabilities and making them more susceptible to sound interference and stimulation. If the ambient sound pressure level is too high (e.g., exceeding 70 decibels), or if there are sudden, piercing sounds in the environment, such as a sudden increase in the film's volume during the movie, these sudden, high-decibel sounds can affect and damage a child's hearing system.

[0047] To address the aforementioned problems, embodiments of this application provide an audio processing method. This method utilizes a device worn on the user's ear, such as headphones, to process the sound signal based on calibration parameters corresponding to the target psychoacoustic loudness of different users. This ensures that the sound signal ultimately heard by the user matches the user's target psychoacoustic loudness, thereby protecting the user's hearing.

[0048] Specifically, for example, taking headphones as an example, children can wear headphones in places like cinemas, KTVs, and arcades mentioned earlier. These headphones can receive sound signals, and then use filters to perform frequency division processing on the sound signals to obtain low-frequency and high-frequency sound signals. Then, steady-state and transient sound signals are extracted from the low-frequency and high-frequency sound signals. The steady-state sound signal represents a relatively smooth and continuous sound signal, while the transient sound signal represents a brief, impulsive sound signal. In some embodiments, the steady-state sound signal can be processed conventionally, while the transient sound signal is calibrated based on some parameters from a personalized calibration parameter set. In other embodiments, both the steady-state and transient sound signals can be calibrated based on personalized calibration parameters; this is not limited to this approach.

[0049] It is understandable that different users have different psychoacoustic loudness levels; for example, a child's psychoacoustic loudness is 15 sones, while an adult's is 20 sones. Therefore, personalized calibration parameters can be determined based on each user's target psychoacoustic loudness, and the frequency-splitting sound signal can be calibrated based on these personalized parameters. Then, the processed low-frequency and high-frequency sound signals are combined to obtain the target pass-through signal, which is then transmitted to the user's ear through headphones. The psychoacoustic loudness of the target pass-through signal matches the user's target psychoacoustic loudness. In this way, the user's hearing system can be protected.

[0050] For example Figure 2 As shown, in a movie theater setting, child A can watch a movie by wearing wireless headphones 10 / 11. Wireless headphones 10 and 11 can process sounds in the received movie sound that are detrimental to the protection of children's hearing, so that the movie sound heard by child A is in line with the psychoacoustic loudness of children, thereby protecting children's hearing.

[0051] It's understandable that both adults and children can protect their hearing by wearing wireless headphones in situations such as movie theaters and noisy places.

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

[0053] To better understand the methods provided in the embodiments of this application, Figure 3 A schematic diagram illustrating the working principle of a wireless earphone 10 transmitting sound signals is shown.

[0054] like Figure 3As shown, the wireless headset 10 may include a microphone 101, an 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 in transmitting sound signals to the human ear is as follows: the microphone 101 receives sound signals from the outside world and converts them into analog electrical signals, and then sends the analog electrical signals to the amplifier 102; the amplifier 102 receives the analog electrical signals, amplifies them, and sends them to the analog-to-digital converter 103; the analog-to-digital converter 103 receives the amplified analog electrical signals, converts them into digital electrical signals, and then sends the digital electrical signals to the digital signal processor 104; the digital signal processor 104 receives the digital electrical signals, processes them, and sends the processed digital electrical signals to the speaker 105; the speaker 105 receives the processed digital electrical signals, converts them into corresponding sound signals, and transmits them to the human ear, so that the user can hear the processed sound signals.

[0056] Further, refer to Figure 4 , Figure 4 A schematic diagram illustrating a specific flow of an audio processing method is shown. This method can be executed by an electronic device. For ease of understanding, this embodiment uses a wireless headset 10 as an example for illustration. The specific flow of the audio processing method includes:

[0057] S201: Acquire sound signal.

[0058] In some embodiments, the user wears wireless headphones 10 in noisy environments. As described above. Figure 3 In the process, the microphone 101 of the wireless earphone 10 receives the sound signal, converts the sound signal into an analog electrical signal, and sends it to the amplifier 102; the amplifier 102 receives the analog electrical signal, amplifies it, and sends it to the analog-to-digital converter 103; the analog-to-digital converter 103 receives the amplified analog electrical signal, converts it into a digital electrical signal, and then sends the digital electrical signal to the digital signal processor 104, thereby obtaining 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 acquires the sound signal, that is, after the digital signal processor 104 of the wireless earphone 10 acquires the digital electrical signal corresponding to the sound signal, the sound pressure level (SPL) of the sound signal can be obtained based on the digital signal processor 104.

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

[0062] Assuming the digital electrical signal corresponding to the sound signal is dBFS, since dBFS is obtained by processing the digital electrical signal dBu by the digital signal processor 104, and 0 dBFS = x dBu, for example, 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 converting the analog electrical signal dBv by the analog-to-digital converter 103, and dBu = dBv + 2.21, the analog electrical signal dBv can be obtained based on the digital electrical signal dBu, and the corresponding analog voltage value can be obtained, assuming the analog voltage value is (x) mV. The analog electrical signal dBv is obtained by converting the sound signal into an analog electrical signal through microphone 101 and then amplifying it through amplifier 102. The gain of the amplifier is 20*LOG10(output / input) dB. For example, if the input of amplifier 102 is 4mV / Pa and the output is 361mV / Pa, then the gain is 20*LOG10(316 / 4) dB. Since the correspondence between sound pressure Pa and analog electrical signal is Pa*(mV / Pa)=(x)mV=(x)*10 -3 Therefore, the sound pressure level 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 microphone 101, and mv / Pa can be 12.87; thus, the sound pressure level Pa can be obtained based on the sensitivity mv / Pa and the analog voltage value corresponding to the analog electrical signal dBv. Finally, the sound pressure level SPL of the sound signal can be obtained based on the correspondence between sound pressure Pa and sound pressure level SPL Pa = Pr * 10^(SPL / 20). Based on the above processing, the correspondence 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 xdBSPL, the corresponding digital electrical signal is ydBFS.

[0063] It is understood that the sensitivity of the microphone 101 of 12.87 mv / Pa mentioned above is only an example. The sensitivity of the microphone 101 in different electronic devices may be different, and no limitation is made here.

[0064] It is understood that the above calculation of the gain of amplifier 102 with an input of 4mV / Pa and an output of 316mV / Pa is merely an illustrative example. In other embodiments, the gain of amplifier 102 may vary depending on the input and output of the amplifier, which is not limited here.

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

[0066] Sound pressure level (SPL) can be understood as representing the intensity of a sound signal, measured in decibels (dB). Therefore, the magnitude of a sound signal can be described using SPL. Bark spectral analysis is essentially based on the SPL of a sound signal, analyzing the corresponding frequency domain signal to obtain a Bark curve. Based on the Bark curve, the psychoacoustic loudness of the sound signal to the user can be obtained.

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

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

[0069] Passive isolation refers to the physical isolation provided by the wireless earphone 10 to the sound signal. Passive isolation can be preset during the development and design phase of the wireless earphone 10, or it can be obtained through real-time detection. For electronic devices that do not support passive isolation detection, the preset passive isolation during the development and design phase can be directly obtained without detection, and this is not limited here. For example, if the preset passive isolation during the development and design phase of the wireless earphone 10 is (m) dB, then the preset passive isolation of the electronic device can be directly filtered out during Bark spectrum analysis. As another example, assuming the sound pressure level of the sound signal obtained through the microphone 101 of the wireless earphone 10 is x dB SPL, and the sound pressure level of the processed sound signal heard at the eardrum is y dB SPL, then the passive isolation of the wireless earphone 10 can be determined to be (xy) dB, and the passive isolation (xy) of the electronic device can be filtered out during Bark spectrum analysis.

[0070] It is understandable that Bark spectral analysis is a commonly used analysis method (algorithm) in signal processing and psychoacoustics research. Bark spectral analysis can convert the frequency domain signal corresponding to the sound signal into a concept that is more in line with the human ear's auditory psychological perception. That is, it maps the frequency domain signal corresponding to the sound signal to the frequency band corresponding to the human ear's auditory psychological perception, so as to obtain the Bark curve corresponding to the sound signal that conforms to the user's psychological perception.

[0071] It is understandable that psychoacoustic loudness can be obtained using the ISO 532-1 or ISO 532-2 physiological acoustic model, or it can be obtained based on other physiological acoustic models, without any limitation here.

[0072] The process of obtaining psychoacoustic loudness based on the ISO 532-1 or ISO 532-2 physiological acoustic model can be understood as follows: First, a sound signal is input into the physiological acoustic model; then, the physiological acoustic model filters the sound signal to obtain sound energy that approximates the human ear's perception curve, i.e., the Bark curve. Further, the frequency domain signal of the filtered sound signal is divided into a series of different frequency bands, and the sound energy of each frequency band is weighted to obtain the loudness value corresponding to each frequency band. Then, based on the weighted frequency band loudness values, the loudness values ​​obtained from each frequency band are synthesized into the final psychoacoustic loudness. Finally, the physiological acoustic model outputs a psychoacoustic loudness, which can be used to measure the user's subjective perception of the intensity of a sound signal.

[0073] S204: When the psychoacoustic loudness of the sound signal to the first target user is inconsistent with the target psychoacoustic loudness of the first target user, the sound signal is divided into frequencies to obtain the steady-state amplitude and transient amplitude of the sound signal.

[0074] It is understandable that the personalized calibration parameters and the target psychoacoustic loudness of the first target user can be obtained by adjusting the calibration parameters. The process of obtaining personalized calibration parameters will be explained below.

[0075] When the psychoacoustic loudness of the sound signal to the first target user is inconsistent with the target psychoacoustic loudness of the first target user, for example, the psychoacoustic loudness of the sound signal to the first target user 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, reference Figure 5 The diagram illustrates frequency division of an audio signal. (For example...) Figure 5 As shown, a Linkwithz-Riley 4th-order (LR-4) filter can be used to divide the audio signal (i.e., the sound signal) into high-frequency signals hs(t) and low-frequency signals 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 audio signal Ls(t), the low-frequency steady-state audio signal Ls_s(t) and the low-frequency transient audio signal Ls_i(t) can be extracted from the low-frequency audio signal Ls(t); when performing two-dimensional median filtering on the high-frequency audio signal hs(t), the high-frequency steady-state audio signal hs_s(t) and the high-frequency transient audio signal hs_i(t) can be extracted from the high-frequency audio signal hs(t).

[0077] Furthermore, by performing equal-loudness weighting calculations on the low-frequency transient sound signal Ls_i(t) and the high-frequency transient sound signal hs_i(t), the transient amplitudes of the low-frequency transient sound signal Ls_i(t) and the high-frequency transient sound signal hs_i(t) can be obtained; by performing equal-loudness weighting calculations and weighted amplitude calculations on the low-frequency steady-state sound signal Ls_s(t) and the high-frequency steady-state sound signal hs_s(t), the transient amplitudes of the low-frequency steady-state sound signal Ls_s(t) and the high-frequency steady-state sound signal hs_s(t) can be obtained. For example, the weight of the low-frequency transient sound signal Ls_i(t) is W1, and the weight of the high-frequency transient sound signal hs_i(t) is W2. The sum of W1 and W2 is 1. Then, the transient amplitude of the transient sound signal is calculated based on the weighted amplitude of W1 and W2. Similarly, the weight of the low-frequency steady-state sound signal Ls_s(t) is W3, and 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 steady-state amplitude of the steady-state sound signal is calculated based on the weighted amplitude of W3 and W4.

[0078] It is understandable that the cutoff frequency when using an LR-4 filter for crossover can be the cutoff frequency fc of the speaker, and no limitation is made here.

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

[0080] It is understandable that when a headset is tested and personalized calibration parameters are entered for two users (e.g., users of different ages or age groups), the electronic device that establishes a communication connection with the headset can respond to the user's selection operation based on the electronic device's display interface, determine that the current user of the headset is the first target user, and send the personalized calibration parameters corresponding to the first target user to the headset; then the headset can calibrate the sound signal received by the headset based on the personalized calibration parameters corresponding to the first target user, so that the psychoacoustic loudness of the calibrated sound signal finally heard by the first target user is consistent with the target psychoacoustic loudness of the first target user.

[0081] The process of obtaining a user's personalized calibration parameters is explained below. It can be understood that the user can be the first target user to obtain other target users.

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

[0083] Since sound signals differ in different environments, in order to obtain personalized calibration parameters, the wireless earphone 10 can play preset calibration signals that can simulate various sound signals (calibration signals are also a type of sound signal) for the user. When the user wears the wireless earphone 10, the user can start calibration through the personalized calibration interface of the electronic device that can configure the wireless earphone 10.

[0084] For example Figure 6 , Figure 6 A schematic diagram of a personalized calibration interface 300 is shown. It can be seen that the personalized calibration interface includes time-domain adjustment and frequency-domain adjustment. When the user clicks "Start Calibration" in the personalized calibration interface 300, the wireless earphone 10 plays a calibration signal that can simulate a sound signal. Based on the calibration signal heard, the user performs time-domain and / or frequency-domain adjustment on the calibration signal to obtain the corresponding time-domain calibration parameters and / or frequency-domain calibration parameters.

[0085] Time-domain adjustment of the calibration signal can modify its impact and psychoacoustic loudness, thereby making the adjusted calibration signal conform to the user's target psychoacoustic loudness.

[0086] refer to Figure 6 The time-domain calibration parameters for time-domain adjustment can include "Attack," "Decay," "Sustain," "Release," and "Peak Amplitude." "Attack," abbreviated as "A," describes the amplitude of the calibration signal as an impulsive increase from time 0 to t1; "Decay," abbreviated as "D," describes the amplitude of the calibration signal as a decreasing decrease from time t1 to t2; "Sustain," abbreviated as "S," describes the amplitude of the calibration signal as a stable state from time t2 to t3; "Release," abbreviated as "R," describes the amplitude of the calibration signal as gradually decreasing to zero from time t3 to t4, i.e., the calibration signal played by the wireless headphones 10 gradually disappears; "Peak Amplitude" is also known as psychoacoustic loudness. After hearing 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," and "R" to adjust the impact of the calibration signal and adjust the "Peak Amplitude." For example, if a user can adjust the peak amplitude of the calibration signal to 65dB, it means that the maximum acceptable psychoacoustic loudness for the user is 65dB. This allows the time-domain adjusted calibration signal to match the user's target psychoacoustic loudness and reduces the perceived impact of the calibration signal. Based on this, the time-domain calibration parameters in the personalized calibration parameters can be obtained.

[0087] It is 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 transient sound signals.

[0088] Frequency domain adjustment of the calibration signal can adjust its loudness and timbre, thereby making the adjusted calibration signal conform to the loudness and timbre of the sound signal perceived by the user.

[0089] Continue to refer to Figure 6 Frequency domain calibration parameters for frequency domain adjustment can include frequency response (Equaliser, EQ), such as... Figure 6 The frequency response curve under mid-frequency domain adjustment reflects the amplitude of the calibration signal at different frequencies. After hearing the calibration signal, the user, based on their psychological perception of the signal, drags and adjusts various frequency points on the frequency response curve to modify the amplitude at different frequencies. Frequency domain adjustment can alter the loudness and timbre of the calibration signal to meet the user's desired requirements. Based on this, the frequency domain calibration parameters in 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 the wireless headphones 10 will stop playing the calibration signal.

[0091] The wireless earphone 10 plays calibration signals that can simulate various sound signals, allowing the user to adjust these signals in the time and / or frequency domains to obtain corresponding time-domain and / or frequency-domain calibration parameters. These time-domain and frequency-domain calibration parameters are the personalized adjustment parameters. Furthermore, when receiving sound signals, the wireless earphone 10 can calibrate the sound signals based on these personalized adjustment parameters, ensuring that the calibrated sound signal, when heard by the user, meets the user's acceptable psychoacoustic loudness, thus protecting the user's hearing.

[0092] It is understandable that users can choose to perform left ear calibration, right ear calibration, or simultaneous calibration of both ears based on the personalized calibration interface 300, without any restrictions.

[0093] It is understandable that the calibration signal is a collection of real signals that simulate various living environments and are likely to have a psychological impact on children or adults, as well as some movie sound clips, and is not limited here.

[0094] Personalized calibration is understood to be a pre-calibration process. Its purpose is to obtain personalized calibration parameters that adjust the sound signal to match the user's psychoacoustic loudness. Personalized calibration can be performed once or multiple times; this 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, proceed to step S206, that is, calibrate the steady-state sound signal based on personalized calibration parameters. Otherwise, proceed 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 transparent signal that meets the user's target psychoacoustic loudness, that is, use a 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, which include time-domain calibration parameters and frequency-domain calibration parameters.

[0099] When the steady-state amplitude of the steady-state sound signal is determined to be greater than the first amplitude value Th_s, the steady-state sound signal can be processed using a filter based on the personalized calibration parameters corresponding to the first target user. In other words, the frequency domain calibration parameters in the personalized calibration parameters corresponding to the first target user can be used as filter coefficients 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) parameter.

[0100] It is understandable that the first amplitude value Th_s is the preset steady-state amplitude threshold.

[0101] In other embodiments, the frequency domain calibration parameters in the personalized calibration parameters corresponding to the first target user can also be used as the filter coefficients 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 transient sound signal obtained above 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, proceed to step S208, that is, calibrate the transient sound signal based on personalized calibration parameters. Otherwise, proceed 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 transparent signal that meets the user's target psychoacoustic loudness. That is, use a signal synthesis filter to synthesize transient sound signals with transient amplitudes less than or equal to the second amplitude value Th_i with steady-state sound signals.

[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 of the transient sound signal is greater than the second amplitude value Th_i, the transient sound signal can be processed by the pass-through compressor based on the personalized calibration parameters corresponding to the first target user. In other words, the time-domain calibration parameters in the personalized calibration parameters corresponding to the first target user can be used as the time-domain compression coefficient of the pass-through compressor to suppress (compress) the transient sound signal.

[0106] It is understandable that the second amplitude value Th_i is the preset transient amplitude threshold.

[0107] In other embodiments, the time-domain calibration parameters in the personalized calibration parameters corresponding to the first target user can also be used as the time-domain compression coefficient of the pass-through compressor to calibrate the steady-state sound signal, which is not limited here.

[0108] S209: The steady-state sound signal and the transient sound signal are synthesized using a signal synthesis filter to obtain the target transparent signal. 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 using a synthesis filter to generate a target pass-through signal HT_target(t). The target pass-through signal refers to the processed sound signal that is ultimately transmitted to the eardrum through the speaker 105, and the psychoacoustic loudness corresponding to the target pass-through 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 pass-through signal and the target psychoacoustic loudness corresponding to the first target user is within a preset range.

[0110] It is understood that, in some embodiments, the inverted 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 using the inverted transparent signal, that is, some noise in the sound signal can be canceled, so that the psychoacoustic loudness corresponding to the target transparent signal that finally reaches the eardrum is consistent with the user's target psychoacoustic loudness, thereby achieving the protection of the user's hearing.

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

[0112] like Figure 7 As shown in the figure, assume the target pass-through signal is HT_target(t), the sound signal is S(t), and the inverted pass-through signal is SPK(t). The target pass-through signal is the signal ultimately heard by the human ear, conforming to the user's target psychoacoustic loudness; the inverted pass-through signal SPK(t) is the signal required to cancel out some noise in the sound signal S(t) in order to obtain the target pass-through signal HT_target(t). The relationship between the target pass-through signal HT_target(t), the sound signal S(t), and the inverted pass-through signal SPK(t) can be found in formula (1).

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

[0114] In formula (1), S(s)·H1(s)·SPK(s)·H2(s) describes the target transparent signal HT_target(s) that is transmitted to the eardrum after the sound signal is calibrated by the transfer function H1(s) of the microphone 101, the inverted transparent signal SPK(s) played by the speaker 105 and the transfer function H2(s) of the speaker 105.

[0115] Furthermore, the inverted transparent signal SPK(t) can be derived from formula (1), and the inverted transparent signal SPK(t) can be obtained based on formula (2).

[0116]

[0117] The following is based on Figure 8 The diagram shown illustrates how a loudspeaker transmits a calibration signal to the eardrum, explaining the principle of obtaining personalized calibration parameters based on a calibration signal that can simulate sound signals.

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

[0119] Assume the sound signal is s(t), and the sound signal reaching the eardrum is input(t); assume the calibration signal that can simulate the sound signal is spk(t). After the user wears the wireless headphones 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 eardrum receiving the sound signal s(t). It can be understood that the microphone 106 is used to simulate the eardrum receiving sound.

[0120] In order for the calibration signal spk(t) to simulate the sound signal s(t), it is necessary to achieve that the transfer function H1 estimated by the sound signal s(t) and the input signal that reaches the eardrum is equal to the transfer function H2 that reaches the microphone 106 (i.e., the eardrum) through the calibration signal spk(t), i.e., refer to formulas (3) and (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 formula (3), S(s) refers to the sound signal, H(s) is the transfer function of S(s), and H(s) is used to convert S(s) into a sound signal that directly reaches the eardrum. S(s)·H(s) describes the sound signal heard by the eardrum. SPK(s) refers to the calibration signal played by the loudspeaker 105, H2(s) is the transfer function of SPK(s), and H2(s) is used to convert the calibration signal SPK(s) into a sound signal heard by the eardrum. Hinv(s) is the transfer function Hinv(s) required for the calibration signal to simulate the sound signal.

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

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

[0126] It is understandable that the tfestimate_() function in formulas (5)-(7) is a tool in MATLAB used to estimate 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, assuming the sound signal S(s) is the same as the calibration signal SPK(s) played by speaker 105, i.e., SPK(s) = S(s), then the above formula (4) can be transformed into formula (8).

[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 formula (9).

[0133]

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

[0135] Based on the above, when sound signal calibration is required, the user can receive the sound signal by wearing wireless headphones 10, and then the speaker of wireless headphones 10 plays an inverted pass-through signal to cancel out some of the noise in the sound signal, so that the sound signal finally heard by the user (i.e., the target pass-through signal) meets the user's acceptable psychoacoustic loudness. This achieves the protection of the user's hearing.

[0136] The method provided in this application embodiment allows a user to receive sound signals by wearing a wireless earphone 10. The wireless earphone 10 then processes the sound signals based on the user's personalized calibration parameters to obtain a target transparent signal that is transmitted to the human ear. Furthermore, the psychoacoustic loudness corresponding to the target transparent signal is consistent with the user's target psychoacoustic loudness, thereby achieving hearing protection for the user.

[0137] It is understood that the audio processing method provided in this application embodiment can be applied to electronic devices with sound pass-through function. The electronic device can customize the control to enable the pass-through function according to specific scenarios, and the time for the pass-through function to be turned off can depend on the duration of the received external sound signal, which is not limited here.

[0138] The various embodiments of the mechanisms disclosed in this application can be implemented in hardware, software, firmware, or a combination of these implementation methods. Embodiments of this application can be implemented as computer programs or program code executable on a programmable system, the programmable system including at least one processor, a storage system (including volatile and non-volatile memory and / or storage elements), at least one input device, and at least one output device.

[0139] Program code can be applied to input instructions to execute the functions described in this application and generate output information. The output information can be applied to one or more output devices in a known manner. For the purposes of this 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 using a high-level procedural language or an object-oriented programming language to communicate with the processing system. Assembly language or machine language can also be used when needed. In fact, the mechanisms described in this application are not limited to any particular programming language. In either 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 thereon on one or more temporary or non-temporary 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. Therefore, machine-readable media may include any mechanism for storing or transmitting information in a machine-readable (e.g., computer-readable) form, including but not limited to floppy disks, optical disks, CD-ROMs, magneto-optical disks, read-only memory (ROM), random access memory (RAM), erasable programmable read-only memory (EPROM), electrically erasable programmable read-only memory (EEPROM), magnetic cards or optical cards, flash memory, or tangible machine-readable storage for transmitting information using electrical, optical, acoustic, or other forms of propagation signals via the Internet. Therefore, machine-readable media include any type of machine-readable medium suitable for storing or transmitting electronic instructions or information in a machine-readable (e.g., computer-readable) form.

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

[0143] It should be noted that all units / modules mentioned in the device embodiments of this application are logical units / modules. Physically, a logical unit / module can be a physical unit / module, a part of a physical unit / module, or a combination of multiple physical units / modules. The physical implementation of these logical units / modules themselves is not the most important factor; the combination of functions implemented by these logical units / modules is the key to solving the technical problems proposed in this application. Furthermore, to highlight the innovative aspects of this application, the above-described device embodiments of this application have not introduced units / modules that are not closely related to solving the technical problems proposed in this application. This does not mean that the above-described device embodiments do not contain other units / modules.

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

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

Claims

1. An audio processing method, characterized by, The method is applied to earphones, and the method comprises the following steps: 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 a 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 a second psychoacoustic loudness of the first target user for the second sound signal is consistent with the target psychoacoustic loudness; the step of obtaining the first psychoacoustic loudness of the first target user for the first sound signal based on the first sound pressure level comprises the following steps: obtaining a first passive isolation degree of the earphones, 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.

2. The method of claim 1, wherein, the first calibration parameter corresponding to the first target user is determined by the following method: the earphones play a first test sound to the first target user; in response to an operation of the first target user adjusting the first test sound to a second test sound; obtaining a parameter value of a time domain calibration parameter and a parameter value of a frequency domain calibration parameter, to obtain the first calibration parameter, wherein the first calibration parameter comprises a first time domain calibration parameter and a first frequency domain calibration parameter.

3. The method of claim 1, wherein, the step of obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal comprises the following steps: 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 value; corresponding to the steady-state amplitude of the first sound signal being greater than the first amplitude threshold value, calibrating the first sound signal based on the first calibration parameter.

4. The method of claim 3, wherein, the step of obtaining the first calibration parameter corresponding to the first target user to calibrate the first sound signal further comprises the following steps: 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 value; corresponding to the transient amplitude of the first sound signal being greater than the second amplitude threshold value, calibrating the first sound signal based on the first calibration parameter.

5. The method of claim 3, wherein, the step of performing frequency division processing on the first sound signal to obtain the steady-state amplitude of the first sound signal comprises the following steps: performing frequency division on the first sound signal by using a filter, 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 amplitudes of the first high-frequency steady-state sound signal and the first low-frequency steady-state sound signal, to obtain the steady-state amplitude of the first sound signal.

6. The method of claim 4, wherein, The frequency division processing on the first sound signal to obtain the transient amplitude of the first sound signal comprises: performing frequency division on the first sound signal by using a filter 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 amplitudes of the first high-frequency transient sound signal and the first low-frequency transient sound signal to obtain the transient amplitude of the first sound signal.

7. The method of claim 3, wherein, The first sound signal corresponds to the steady-state amplitude greater than the first amplitude threshold, and the first sound signal is calibrated based on the first calibration parameter, comprising: taking the first frequency domain calibration parameter as a filter coefficient of a filter, and calibrating the steady-state sound signal of the first sound signal based on the filter to obtain the steady-state sound signal of the calibrated first sound signal.

8. The method of claim 4, wherein, The first sound signal corresponds to the transient amplitude greater than the second amplitude threshold, and the first sound signal is calibrated based on the first calibration parameter, comprising: taking the first time domain calibration parameter as a time domain compression parameter of a transparent compressor, and calibrating the transient sound signal of the first sound signal based on the transparent compressor to obtain the transient sound signal of the calibrated first sound signal.

9. The method of claim 1, wherein, The first calibration parameter corresponding to the first target user is obtained, the first sound signal is calibrated based on the first calibration parameter, and a second sound signal is obtained, comprising: synthesizing the steady-state sound signal of the calibrated first sound signal and the transient sound signal of the calibrated first sound signal by using a synthesis filter to obtain the second sound signal.

10. The method of claim 1, wherein, The first calibration parameter is a calibration parameter corresponding to a child.

11. An earphone, characterized by The earphone comprises a memory for storing instructions, and a processor for executing the instructions to implement the audio processing method of any one of claims 1-10.

12. A readable storage medium, characterized by, The readable storage medium has instructions stored thereon, and the instructions implement the audio processing method of any one of claims 1-10 when executed on the earphone.

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