Sound compensation method and device, earphone and storage medium

Abstract

The invention discloses a sound compensation method and device, an earphone and a storage medium, and the method comprises the steps: responding to the condition that the earphone is worn on a human ear, playing a test audio signal through a moving-iron loudspeaker and a moving-coil loudspeaker, and collecting an ear canal signal of the test audio signal through a feedback microphone; determining an amplitude response curve of a first frequency band based on the test audio signal and the ear canal signal, and determining a response error curve based on the amplitude response curve; determining time delay information of the ear canal signal based on a plurality of candidate crossover frequencies; and determining a plurality of loss values based on the plurality of candidate crossover frequencies, the response error curve and the time delay information, determining a digital filtering coefficient based on the plurality of loss values, and performing sound compensation based on the digital filtering coefficient. And the accuracy of sound restoration and the consistency of hearing feeling are obviously improved.

Description

Technical Field

[0001] This invention relates to the field of headphone technology, and more particularly to a sound compensation method, device, headphone, and storage medium. Background Technology

[0002] With the increasing demand for high-quality, personalized listening experiences in the consumer electronics sector, the acoustic performance of in-ear headphones has become a core factor determining user experience. Especially for hybrid driver headphones combining dynamic and balanced armature drivers, ensuring consistent frequency response across different ear canal structures and wearing conditions is a key challenge for improving sound fidelity and user satisfaction.

[0003] In existing technologies, fixed crossover networks or digital equalizers are commonly used to adjust the frequency response of hybrid driver earphones. However, fixed crossover networks can only guarantee power distribution at the electrical signal level and cannot compensate for acoustic phase misalignment and energy superposition distortion caused by differences in driver physical position and ear canal structure; while simple digital equalizer compensation can only adjust the amplitude response and cannot solve the comb filtering effect and group delay problems caused by phase mismatch near the crossover point. Therefore, existing technologies cannot fundamentally achieve personalized, high-fidelity sound reproduction for different users. Summary of the Invention

[0004] This invention provides a sound compensation method, device, earphone, and storage medium to solve the problem of inconsistent sound quality in hybrid driver earphones caused by differences in the ear canal.

[0005] According to one aspect of the present invention, a sound compensation method is provided, applied to a processor of an earphone, the earphone further comprising a feedback microphone, a balanced armature speaker, and a dynamic driver, the method comprising:

[0006] In response to the earphone being worn by a person, a test audio signal is played through the balanced armature speaker and the dynamic driver, and the ear canal signal of the test audio signal is collected through the feedback microphone;

[0007] Based on the test audio signal and ear canal signal, determine the amplitude response curve of the first frequency band, and determine the response error curve based on the amplitude response curve;

[0008] The time delay information of the ear canal signal is determined based on multiple candidate frequency divisions;

[0009] Multiple loss values ​​are determined based on the multiple candidate frequency division frequencies, response error curves, and time delay information. Digital filter coefficients are determined based on the multiple loss values. Sound compensation is performed based on the digital filter coefficients.

[0010] According to another aspect of the present invention, a sound compensation device is provided, applied to a processor of an earphone, the earphone further comprising a feedback microphone, a balanced armature speaker, and a dynamic driver, the device comprising:

[0011] The signal processing module is used to respond to the earphone being worn in the ear by playing a test audio signal through the balanced armature speaker and the dynamic driver, and to collect the ear canal signal of the test audio signal through the feedback microphone;

[0012] The curve determination module is used to determine the amplitude response curve of the first frequency band based on the test audio signal and the ear canal signal, and to determine the response error curve based on the amplitude response curve.

[0013] A delay determination module is used to determine the delay information of the ear canal signal based on multiple candidate frequency divisions;

[0014] The coefficient determination module is used to determine multiple loss values ​​based on the multiple candidate frequency division frequencies, response error curves and time delay information, determine digital filter coefficients based on the multiple loss values, and perform sound compensation based on the digital filter coefficients.

[0015] According to another aspect of the present invention, an earphone is provided, the earphone comprising:

[0016] A moving iron horn, used to play the first frequency band of test audio signals;

[0017] A dynamic speaker is used to play the second frequency band of the test audio signal, wherein the first frequency band signal and the second frequency band signal constitute an ear canal signal;

[0018] The feedback microphone is used to collect ear canal signals of the test audio signal;

[0019] One or more processors;

[0020] Storage device for storing one or more programs for testing audio signals and target amplitude response curves;

[0021] When the one or more programs are executed by the one or more processors, the one or more processors implement the sound compensation method described in any embodiment of the present invention.

[0022] According to another aspect of the present invention, a storage medium containing computer-executable instructions is also provided, which, when executed by a computer processor, are used to perform the sound compensation method described in any embodiment of the present invention.

[0023] The technical solution of this invention involves playing test audio signals from a balanced armature speaker and a dynamic driver when the earphone is worn in the ear, and collecting the ear canal signal of the test audio signal using a feedback microphone; determining the amplitude response curve of the first frequency band based on the obtained signal, and then generating a response error curve; analyzing the time delay information of the ear canal signal by combining multiple candidate crossover frequencies; and finally calculating the loss value based on the response error and time delay information corresponding to each candidate crossover frequency to determine the digital filter coefficient and achieve sound compensation. This solves the problem of frequency response deviation and phase mismatch caused by ear canal differences in traditional hybrid earphones, and achieves the beneficial effects of improving the consistency of hearing for different users and enhancing the accuracy of sound reproduction.

[0024] It should be understood that the description in this section is not intended to identify key or essential features of the embodiments of the present invention, nor is it intended to limit the scope of the invention. Other features of the invention will become readily apparent from the following description. Attached Figure Description

[0025] To more clearly illustrate the technical solutions in the embodiments of the present invention, the accompanying drawings used in the description of the embodiments will be briefly introduced below. Obviously, the accompanying drawings described below are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0026] Figure 1 This is a flowchart of a sound compensation method provided according to Embodiment 1 of the present invention;

[0027] Figure 2 This is a flowchart of another sound compensation method provided in Embodiment 2 of the present invention;

[0028] Figure 3 This is a flowchart of a sound compensation method applicable to a specific scenario in which this invention is implemented;

[0029] Figure 4 This is a schematic diagram of the structure of a sound compensation device according to Embodiment 3 of the present invention;

[0030] Figure 5 This is a schematic diagram of the structure of an earphone device that implements an embodiment of the present invention. Detailed Implementation

[0031] To enable those skilled in the art to better understand the present invention, the technical solutions of the present invention will be clearly and completely described below with reference to the accompanying drawings of the embodiments of the present invention. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort should fall within the scope of protection of the present invention.

[0032] It should be noted that the terms "first," "second," etc., in the specification, claims, and accompanying drawings of this invention are used to distinguish similar objects and are not necessarily used to describe a specific order or sequence. It should be understood that such data can be interchanged where appropriate so that the embodiments of the invention described herein can be implemented in orders other than those illustrated or described herein. Furthermore, the terms "comprising" and "having," and any variations thereof, are intended to cover non-exclusive inclusion; for example, a process, method, system, product, or apparatus that comprises a series of steps or units is not necessarily limited to those steps or units explicitly listed, but may include other steps or units not explicitly listed or inherent to such processes, methods, products, or apparatus.

[0033] Example 1

[0034] Figure 1 This is a flowchart of a sound compensation method provided in Embodiment 1 of the present invention. This embodiment is applicable to situations where headphones automatically adjust the crossover parameters according to the characteristics of the user's ear canal. The method can be executed by a sound compensation device, which can be implemented in hardware and / or software and is generally configured in the headphone processor.

[0035] Correspondingly, such as Figure 1 As shown, the method includes:

[0036] S110. In response to the earphone being worn in the ear, a test audio signal is played through the balanced armature speaker and the dynamic driver, and the ear canal signal of the test audio signal is collected through the feedback microphone.

[0037] A balanced armature speaker is an electrodynamic transducer responsible for reproducing high-frequency sounds. Internally, it is driven by a balanced armature to play the high-frequency portion of an audio signal. For example, a balanced armature speaker can play the high-frequency portion of a test audio signal. A dynamic coil speaker is an electrodynamic transducer responsible for reproducing mid-to-low-frequency sounds. Internally, it is driven by a voice coil to play the mid-to-low-frequency portion of an audio signal. For example, a dynamic coil speaker can play the mid-to-low-frequency portion of a test audio signal. The test audio signal can be understood as a pre-designed, known audio signal, typically covering all or a significant portion of the audible frequency range.

[0038] A feedback microphone is an acoustic sensor integrated inside headphones used to pick up sound signals. For example, a feedback microphone is used to collect ear canal signals for testing audio signals.

[0039] In this embodiment, the calibration process is triggered when the earphone is detected to be worn in the user's ear canal. Specifically, the integrated balanced armature and dynamic drivers inside the earphone play a specific test audio signal. This signal is reflected and modified by the ear canal space, and then received by the feedback microphone and synthesized into a complete ear canal signal. This process aims to capture the unique acoustic transmission characteristics of an individual's ear canal in real time, providing the raw data foundation for subsequent sound compensation.

[0040] Optionally, based on the above embodiments, the earphone is communicatively connected to a mobile device, and the step of playing a test audio signal through the balanced armature speaker and the dynamic driver in response to the earphone being worn on the ear includes:

[0041] In response to the earphone being worn on a person's ear, the device receives a calibration command sent by the mobile device and controls the balanced armature speaker and the dynamic driver to play the test audio signal.

[0042] Generally, after the headphones are worn, a mobile device, such as a smartphone or music player, that establishes a communication connection with the headphones actively sends a calibration start command to the headphones. This can be achieved by displaying calibration controls on the mobile device's interface, which, in response to being triggered, send a calibration start command to the headphones. Only after receiving this specific command from the external device will the headphones trigger their internal control logic, instructing their built-in balanced armature and dynamic drivers to play a preset test audio signal, thereby initiating the entire measurement and compensation process.

[0043] S120. Based on the test audio signal and ear canal signal, determine the amplitude response curve of the first frequency band, and determine the response error curve based on the amplitude response curve.

[0044] The amplitude response curve of the first frequency band refers to the relationship between the sound frequency and energy actually measured in the current user's ear canal, calculated by analyzing the test audio signal and the received ear canal signal. The response error curve is the difference curve obtained by comparing the measured amplitude response curve of the first frequency band with a pre-stored target curve representing an ideal flatness or a specific tuning goal.

[0045] In this embodiment, the acquired test audio signal and ear canal signal are transformed to calculate the actual amplitude response curve within the audible frequency range. Subsequently, the measured curve is compared with the target ideal frequency response curve pre-stored in the processor, and the difference between the two is calculated, thus obtaining the response error curve. This error curve quantifies the degree of deviation between the current ear canal characteristics and the ideal reproduction target.

[0046] Optionally, based on the above embodiments, determining the amplitude response curve of the first frequency band based on the test audio signal and the ear canal signal, and determining the response error curve based on the amplitude response curve, may include:

[0047] Perform a Fourier transform on the ear canal signal, and determine the amplitude response curve of the first frequency band based on the transform result;

[0048] The response error curve is obtained by calculating the amplitude response curve and the target amplitude response curve.

[0049] The test audio signal can be the time-domain signal of the test audio.

[0050] Generally, in order to convert a recorded sound signal that changes over time into a representation that reflects its frequency composition, a Fourier transform is required on the acquired ear canal signal. This transform process yields a result that includes the intensity information of each frequency component in the signal.

[0051] Generally, based on the above test audio signals and transformation results, a ratio characterizing the transmission characteristics of the ear canal can be obtained. Then, the logarithm of this ratio is calculated, and finally, the amplitude response curve of the ear canal's actual modification effect on sound energy within the audible frequency range is plotted.

[0052] Generally, the target amplitude response curve, Harman IE-2019Target(f), representing the ideal playback effect, is loaded first. Harman IE-2019Target(f) is a target frequency response curve for in-ear headphones, derived by Harman International through scientific research, representing the preferences of most listeners. The measured amplitude response curve is then subtracted from the target amplitude response curve at the same frequency point; the resulting difference sequence constitutes the response error curve. The calculation formula is: It quantifies the deviation values ​​between the actual acoustic characteristics of the ear canal and the ideal state at various frequencies.

[0053] S130. Determine the time delay information of the ear canal signal based on multiple candidate frequency divisions.

[0054] Candidate crossover frequencies refer to a series of alternative frequency values ​​preset to find the optimal crossover point. These frequencies are typically generated at certain intervals within the effective overlapping frequency bands of balanced armature and dynamic coil horns. Ear canal signal delay information specifically refers to the time delay that occurs from the transmission of the sound signal to its reception when a candidate crossover frequency is applied to a crossover filter.

[0055] In this embodiment, to optimize phase matching at the crossover point, the impact of different crossover frequencies on sound delay needs to be evaluated. Within a preset range of commonly used crossover frequencies, a series of candidate crossover frequencies are generated with a fixed step size. For each candidate frequency, its corresponding filter parameters are calculated, and based on these, the delay information of the ear canal signal after filtering is calculated. This delay information reflects the temporal consistency of the sound signal.

[0056] Optionally, based on the above embodiments, determining the time delay information of the ear canal signal based on multiple candidate frequency divisions may include:

[0057] By traversing the second frequency band according to a preset step size, multiple candidate frequency division frequencies are obtained.

[0058] For each candidate frequency division, the corresponding low-pass coefficient and high-pass coefficient are determined, and the expected frequency response curve is determined based on the low-pass coefficient, high-pass coefficient and the ear canal signal;

[0059] Based on the expected frequency response curve corresponding to each candidate frequency division frequency, the time delay information corresponding to each candidate frequency division frequency is determined.

[0060] Low-pass coefficients are a set of core parameters used to define digital low-pass filters, determining how the filter processes signals of different frequencies. High-pass coefficients are a set of core parameters used to define digital high-pass filters, and their function is the opposite of low-pass coefficients. The mathematical rules governing these coefficients enable the filter to retain high-frequency components of the input signal above a specific frequency division point, while effectively attenuating low-frequency components below that frequency division point.

[0061] Generally, to find the most suitable crossover point, a large number of candidate crossover frequencies need to be selected sequentially at fixed intervals within the common frequency range where the moving coil and balanced armature drivers are effectively operating. These candidate points form the basis for subsequent evaluation and comparison.

[0062] Generally, after determining a candidate division frequency, digital filter parameters are calculated based on that frequency. These parameters include low-pass and high-pass coefficients. Specifically, a set of low-pass coefficients is calculated mathematically based on the division frequency, the signal sampling rate, the selected filter type, and the filter order. To obtain the corresponding high-pass coefficients, a classic approach is to alternately flip the signs of the calculated low-pass coefficients. For example, if the low-pass coefficients are (a1, a2, a3, a4, a5), the high-pass coefficients can be quickly determined as (a1, -a2, a3, -a4, a5). This method ensures that the high-pass and low-pass filters have matching characteristics at the division point.

[0063] After obtaining the low-pass and high-pass coefficients for the candidate crossover frequency, these two sets of coefficients are used to construct low-pass and high-pass filters, respectively, to filter the actual acquired ear canal signal. The filtered low-frequency signal is then recombined with the high-frequency signal to calculate the possible sound frequency response characteristics, i.e., the expected frequency response curve, that might be formed in the ear canal if this crossover frequency is used.

[0064] Generally, based on the expected sound frequency response curves calculated for each candidate crossover frequency, the time delay of different frequency components after the sound signal passes through the crossover network can be further analyzed. This delay information specifically reflects the synchronicity of the high and low frequency units under this crossover point setting, and is one of the key indicators for evaluating the performance of the crossover point.

[0065] S140. Based on the multiple candidate frequency division, response error curve and time delay information, determine multiple loss values, determine digital filter coefficients based on the multiple loss values, and perform sound compensation based on the digital filter coefficients.

[0066] The loss value can be understood as a quantitative indicator used to comprehensively evaluate the performance of each candidate frequency division. Digital filter coefficients are a set of core parameters used to define the characteristics of a digital filter, determining how the filter processes the input signal, such as how it amplifies, attenuates, or delays components of different frequencies.

[0067] In this embodiment, considering both frequency response accuracy and phase consistency, a comprehensive evaluation index, namely the loss value (Cost), is defined for each candidate frequency division. The calculation of the loss value takes into account both the total response error within a specific frequency band centered on the candidate frequency and the corresponding time delay. By traversing all candidate frequencies and calculating their loss values, the candidate point that minimizes this index is finally selected as the optimal frequency division, and the final digital filter coefficients are determined accordingly, thereby achieving precise adaptive frequency division and compensation for the drive signals of the moving coil and balanced armature units.

[0068] The technical solution of this invention involves playing test audio signals from a balanced armature speaker and a dynamic driver when the earphone is worn in the ear, and collecting the ear canal signal of the test audio signal using a feedback microphone; determining the amplitude response curve of the first frequency band based on the obtained signal, and then generating a response error curve; analyzing the time delay information of the ear canal signal by combining multiple candidate crossover frequencies; and finally calculating the loss value based on the response error and time delay information corresponding to each candidate crossover frequency to determine the digital filter coefficient and achieve sound compensation. This solves the problem of frequency response deviation and phase mismatch caused by ear canal differences in traditional hybrid earphones, and achieves the beneficial effects of improving the consistency of hearing for different users and enhancing the accuracy of sound reproduction.

[0069] Example 2

[0070] Figure 2 This is a flowchart of another sound compensation method provided in Embodiment 2 of the present invention. This embodiment is based on the above embodiments and optimized. Specifically, the operation of "determining multiple loss values ​​based on the multiple candidate frequency division frequencies, response error curves, and time delay information" has been refined.

[0071] Correspondingly, such as Figure 2 As shown, the method includes:

[0072] S210. In response to the earphone being worn in the ear, a test audio signal is played through the balanced armature speaker and the dynamic driver, and the ear canal signal of the test audio signal is collected through the feedback microphone.

[0073] S220. Based on the test audio signal and ear canal signal, determine the amplitude response curve of the first frequency band, and determine the response error curve based on the amplitude response curve.

[0074] S230. Determine the time delay information of the ear canal signal based on multiple candidate frequency divisions.

[0075] S240. For each candidate frequency division, take the current candidate frequency division as the center, determine multiple frequencies within a preset range, and determine the response error based on the multiple frequencies and the response error curve.

[0076] In this embodiment, for each candidate frequency division being evaluated, it is necessary to examine the frequency response error within a specific frequency range centered on that frequency. Specifically, multiple discrete frequency points within a certain range above and below the center frequency are selected, and the error values ​​corresponding to these frequency points on the response error curve are accumulated to obtain the response error reflecting the comprehensive frequency response deviation in the vicinity of the candidate frequency division.

[0077] S250. Based on the response error and delay information corresponding to each candidate frequency division frequency, determine the loss value corresponding to each candidate frequency division frequency.

[0078] In this embodiment, the decision-making basis for determining the final frequency division point is a loss value that comprehensively considers both frequency response accuracy and phase consistency. This value is calculated by weighting the response error within a specific frequency band obtained in the above steps with the group delay information generated by the corresponding candidate frequency division, forming a single evaluation index to objectively measure the overall performance of each candidate frequency division. The formula for calculating the loss value is as follows:

[0079]

[0080] in, The candidate frequency division currently being evaluated, As a The frequency range centered on For time delay, The square of the absolute value. As a summation symbol, 1oct represents one octave.

[0081] S260. Determine digital filtering coefficients based on the multiple loss values, and perform sound compensation based on the digital filtering coefficients.

[0082] Optionally, based on the above embodiments, the step of determining digital filter coefficients based on the plurality of loss values ​​and performing sound compensation based on the digital filter coefficients may include:

[0083] Based on the sorting results of the multiple loss values, the target loss value is determined;

[0084] The digital filtering coefficients are determined based on the target loss value, and then sent to the frequency division control module. The frequency division control module performs sound compensation based on the digital filtering coefficients. The frequency division control module is used to determine the driving power of the balanced armature speaker and the moving coil speaker based on the digital filtering coefficients.

[0085] Generally, after calculating the loss values ​​corresponding to all candidate frequency divisions, these loss values ​​are sorted and compared, and the one with the smallest value is selected as the target loss value, because it represents the candidate point with the best overall performance.

[0086] Generally, once the target loss value is determined, the final digital filter coefficients can be calculated and determined based on the corresponding optimal candidate crossover frequency and the relevant time delay compensation requirements. These coefficients are then configured into a dedicated crossover control module responsible for signal distribution. This module processes the input digital audio signal in real time based on the applied digital filter coefficients, precisely controlling the frequency components, amplitude, and phase of the drive signals output to the balanced armature and dynamic coil speakers. In other words, it achieves adaptive sound compensation by adjusting the distribution of drive power.

[0087] Optionally, based on the above embodiments, determining the digital filter coefficients based on the target loss value may include:

[0088] The digital filter coefficients are determined based on the candidate frequency division frequency corresponding to the target loss value, the boundary frequencies corresponding to the plurality of frequencies, and the time delay information.

[0089] Here, the boundary frequency refers to the upper and lower limit frequencies set to evaluate the performance of the candidate frequency division. Optionally, the boundary frequency is determined based on the candidate frequency division and the length of the frequency selection window. It is the effective range for measuring frequency response error during algorithm optimization.

[0090] Generally, the final determination of digital filter coefficients is a comprehensive decision-making process. This process is based on the target loss value obtained through optimization, and takes its corresponding optimal candidate frequency division as the core parameter. At the same time, it combines the boundary frequencies GLF and GHF determined when calculating the loss value, as well as the time delay information obtained by measurement (for phase compensation). These key parameters are incorporated into a specific coefficient synthesis algorithm for calculation, thereby generating the final filter parameters that can simultaneously achieve accurate frequency division, amplitude correction and phase alignment.

[0091] The technical solution of this invention involves playing test audio signals from a balanced armature speaker and a dynamic driver when the earphone is worn in the ear, and collecting the ear canal signal of the test audio signal using a feedback microphone. Based on the obtained signal, the amplitude response curve of the first frequency band is determined, and a response error curve is generated. Then, the time delay information of the ear canal signal is analyzed in combination with multiple candidate crossover frequencies. Furthermore, for each candidate crossover frequency, the response error within a preset frequency range is determined with that frequency as the center. Finally, the loss value is calculated by comprehensively considering the response error and time delay information corresponding to each candidate crossover frequency to determine the digital filter coefficient and achieve sound compensation. This solves the problem of frequency response deviation and phase mismatch caused by ear canal differences in traditional hybrid earphones. By incorporating the amplitude error and phase delay of the frequency band adjacent to the crossover point into a unified evaluation system for comprehensive optimization, the scientificity and accuracy of the crossover point selection are significantly improved, resulting in beneficial effects such as making the earphone output frequency response more closely match the target curve, improving phase consistency, and significantly enhancing the consistency of listening experience for different users.

[0092] For ease of understanding, the specific application scenarios applicable to each embodiment of the invention are described. In this specific embodiment, in order to solve the problem of inconsistent hearing caused by differences in ear canal fit when wearing headphones, this embodiment of the invention designs a complete sound compensation scheme.

[0093] Specifically, in Figure 3 The figure shows a flowchart of a sound compensation method used in an embodiment of the present invention. As shown, wearing detection is performed first. When wearing of the headphones is detected, the ear canal signal is collected. Next, a Fast Fourier Transform is performed on the collected signal to calculate the actual ear canal frequency response. At the same time, a pre-stored Harman target curve is loaded, and the error curve between the two is calculated. Subsequently, the frequency is traversed starting from 1.5kHz and increasing in 100Hz increments. In each loop, the corresponding LR4 filter coefficients are generated for the current candidate frequency, and the expected frequency response curve is synthesized. Based on this, the sum of squared errors within an octave window around the frequency point is calculated, and the corresponding group delay data is extracted. Then, the comprehensive loss value of the current candidate frequency is calculated according to the formula. This loop process continues until the frequency traversal reaches 4kHz. After that, the frequency division point with the minimum loss value is selected from all candidate frequencies, and the corresponding new filter coefficients are finally issued, completing the entire adaptive compensation process and ending the process.

[0094] Furthermore, through the clever combination of the above steps, sound compensation can be achieved, resulting in the following effective effects:

[0095] 1) By collecting the acoustic characteristics of the ear canal in real time through a feedback microphone and comparing and analyzing them with the Harman target curve, a personalized and adaptive sound compensation effect can be achieved for different ear canal shapes of users.

[0096] 2) By incorporating the frequency response error of the frequency bands adjacent to the crossover point and the signal group delay into a unified loss function for joint optimization, the method can simultaneously correct amplitude distortion and phase mismatch, and significantly improve the accuracy of sound reproduction and the clarity of imaging.

[0097] 3) By using a preset candidate crossover frequency traversal and loss value rapid evaluation algorithm, combined with a dynamic filter coefficient distribution mechanism, the parameter self-calibration is completed the instant the user wears the headphones, achieving a seamless, real-time high-fidelity listening experience optimization effect.

[0098] Example 3

[0099] Figure 4 This is a schematic diagram of a sound compensation device provided in Embodiment 3 of the present invention. Figure 4 As shown, the device includes: a signal processing module 410, a curve determination module 420, a time delay determination module 430, and a coefficient determination module 440, wherein:

[0100] Signal processing module 410, the signal processing module, is used to respond to the earphone being worn in the ear, play a test audio signal through the balanced armature speaker and the dynamic coil speaker, and collect the ear canal signal of the test audio signal through the feedback microphone;

[0101] The curve determination module 420 is used to determine the amplitude response curve of the first frequency band based on the test audio signal and the ear canal signal, and to determine the response error curve based on the amplitude response curve.

[0102] The delay determination module 430 is used to determine the delay information of the ear canal signal based on multiple candidate frequency divisions;

[0103] The coefficient determination module 440 is used to determine multiple loss values ​​based on the multiple candidate frequency division frequencies, response error curves and time delay information, determine digital filter coefficients based on the multiple loss values, and perform sound compensation based on the digital filter coefficients.

[0104] The technical solution of this invention involves playing test audio signals from a balanced armature speaker and a dynamic driver when the earphone is worn in the ear, and collecting the ear canal signal of the test audio signal using a feedback microphone; determining the amplitude response curve of the first frequency band based on the obtained signal, and then generating a response error curve; analyzing the time delay information of the ear canal signal by combining multiple candidate crossover frequencies; and finally calculating the loss value based on the response error and time delay information corresponding to each candidate crossover frequency to determine the digital filter coefficient and achieve sound compensation. This solves the problem of frequency response deviation and phase mismatch caused by ear canal differences in traditional hybrid earphones, and achieves the beneficial effects of improving the consistency of hearing for different users and enhancing the accuracy of sound reproduction.

[0105] Based on the above embodiments, the curve determination module 420 is specifically used for:

[0106] Perform a Fourier transform on the ear canal signal, and determine the amplitude response curve of the first frequency band based on the transform result;

[0107] The response error curve is obtained by calculating the amplitude response curve and the target amplitude response curve.

[0108] Based on the above embodiments, the delay determination module 430 is specifically used for:

[0109] By traversing the second frequency band according to a preset step size, multiple candidate frequency division frequencies are obtained.

[0110] For each candidate frequency division, the corresponding low-pass coefficient and high-pass coefficient are determined, and the expected frequency response curve is determined based on the low-pass coefficient, high-pass coefficient and the ear canal signal;

[0111] Based on the expected frequency response curve corresponding to each candidate frequency division frequency, the time delay information corresponding to each candidate frequency division frequency is determined.

[0112] Based on the above embodiments, the coefficient determination module 440 is specifically used for:

[0113] For each candidate frequency division, multiple frequencies within a preset range are determined with the current candidate frequency division as the center, and the response error is determined based on the multiple frequencies and the response error curve.

[0114] Based on the response error and delay information corresponding to each candidate frequency division frequency, the loss value corresponding to each candidate frequency division frequency is determined.

[0115] Furthermore, based on the above embodiments, the coefficient determination module 440 may further include:

[0116] The loss value determination submodule is used to determine the target loss value based on the sorting result of the multiple loss values;

[0117] The compensation submodule is used to determine the digital filtering coefficients based on the target loss value, pass the digital filtering coefficients to the frequency division control module, and perform sound compensation based on the digital filtering coefficients through the frequency division control module. The frequency division control module is used to determine the driving power of the balanced armature speaker and the moving coil speaker based on the digital filtering coefficients.

[0118] Based on the above embodiments, the compensation submodule is specifically used for:

[0119] The digital filter coefficients are determined based on the candidate frequency division frequency corresponding to the target loss value, the boundary frequencies corresponding to the plurality of frequencies, and the time delay information.

[0120] Based on the above embodiments, the earphone is communicatively connected to the mobile device. Correspondingly, based on the above embodiments, the signal processing module 410 is specifically used for:

[0121] In response to the earphone being worn on a person's ear, the device receives a calibration command sent by the mobile device and controls the balanced armature speaker and the dynamic driver to play the test audio signal.

[0122] The sound compensation device provided in the embodiments of the present invention can execute the sound compensation method provided in any embodiment of the present invention, and has the corresponding functional modules and beneficial effects of the method.

[0123] The collection, storage, use, processing, transmission, provision, and disclosure of user personal information involved in the technical solution disclosed herein comply with the provisions of relevant laws and regulations and do not violate public order and good morals.

[0124] Example 4

[0125] Figure 5 A schematic diagram of a headphone device that can be used to implement embodiments of the present invention is shown. As shown, the in-ear headphone, from the outside in and from front to back, includes the following core components connected by wires: ear tips and ear tip supports for sealing and securing the earphone; the inner earphone cover forms the outer surface; the internal FB feedback microphone is responsible for picking up acoustic feedback within the ear canal; the dynamic driver and balanced armature driver are cooperative sound-producing units, responsible for reproducing mid-low and high frequencies respectively; a magnetic shielding sheet is provided behind the unit to suppress electromagnetic interference; the battery is fixed by a battery holder and supplies power to the system; the PCBA (Printed Circuit Board Assembly) board serves as the core circuit, integrating signal processing, driving, and control functions; the earphone shell provides main structural support; and the earphone rear shell completes the overall encapsulation. These components, through precision assembly and electrical connection, together constitute a complete adaptive acoustic system.

[0126] The device includes a balanced armature speaker for playing a first frequency band of the test audio signal; a dynamic coil speaker for playing a second frequency band of the test audio signal, the first and second frequency band signals constituting an ear canal signal; a feedback microphone for acquiring the ear canal signal of the test audio signal; one or more processors; a storage device for storing one or more programs for testing the audio signal and the target amplitude response curve; and a PCBA board containing the earphone's processor and storage medium.

[0127] In the context of this invention, a computer-readable storage medium can be a tangible medium that may contain or store a computer program for use by or in conjunction with an instruction execution system, apparatus, or device. A computer-readable storage medium may include, but is not limited to, electronic, magnetic, optical, electromagnetic, infrared, or semiconductor systems, apparatus, or devices, or any suitable combination thereof. Alternatively, a computer-readable storage medium may be a machine-readable signal medium. More specific examples of machine-readable storage media include electrical connections based on one or more wires, portable computer disks, hard disks, random access memory (RAM), read-only memory (ROM), erasable programmable read-only memory (EPROM or flash memory), optical fibers, portable compact disk read-only memory (CD-ROM), optical storage devices, magnetic storage devices, or any suitable combination thereof.

[0128] It should be understood that the various forms of processes shown above can be used, with steps reordered, added, or deleted. For example, the steps described in this invention can be executed in parallel, sequentially, or in different orders, as long as the desired result of the technical solution of this invention can be achieved, and this is not limited herein.

[0129] The specific embodiments described above do not constitute a limitation on the scope of protection of this invention. Those skilled in the art should understand that various modifications, combinations, sub-combinations, and substitutions can be made according to design requirements and other factors. Any modifications, equivalent substitutions, and improvements made within the spirit and principles of this invention should be included within the scope of protection of this invention.

Claims

1. A sound compensation method, characterized in that, A processor for use in headphones, the headphones further including a feedback microphone, a balanced armature speaker, and a dynamic driver, the method comprising: In response to the earphone being worn by a person, a test audio signal is played through the balanced armature speaker and the dynamic driver, and the ear canal signal of the test audio signal is collected through the feedback microphone; Based on the test audio signal and ear canal signal, determine the amplitude response curve of the first frequency band, and determine the response error curve based on the amplitude response curve; The time delay information of the ear canal signal is determined based on multiple candidate frequency divisions; Multiple loss values ​​are determined based on the multiple candidate frequency division frequencies, response error curves, and time delay information. Digital filter coefficients are determined based on the multiple loss values. Sound compensation is performed based on the digital filter coefficients.

2. The method according to claim 1, characterized in that, The step of determining the amplitude response curve of the first frequency band based on the test audio signal and ear canal signal, and determining the response error curve based on the amplitude response curve, includes: Perform a Fourier transform on the ear canal signal, and determine the amplitude response curve of the first frequency band based on the transform result; The response error curve is obtained by calculating the amplitude response curve and the target amplitude response curve.

3. The method according to claim 1, characterized in that, The determination of the time delay information of the ear canal signal based on multiple candidate frequency divisions includes: By traversing the second frequency band according to a preset step size, multiple candidate frequency division frequencies are obtained. For each candidate frequency division, the corresponding low-pass coefficient and high-pass coefficient are determined, and the expected frequency response curve is determined based on the low-pass coefficient, high-pass coefficient and the ear canal signal; Based on the expected frequency response curve corresponding to each candidate frequency division frequency, the time delay information corresponding to each candidate frequency division frequency is determined.

4. The method according to claim 1, characterized in that, The determination of multiple loss values ​​based on the multiple candidate frequency division frequencies, response error curves, and time delay information includes: For each candidate frequency division, multiple frequencies within a preset range are determined with the current candidate frequency division as the center, and the response error is determined based on the multiple frequencies and the response error curve. Based on the response error and delay information corresponding to each candidate frequency division frequency, the loss value corresponding to each candidate frequency division frequency is determined.

5. The method according to claim 1, characterized in that, The step of determining digital filter coefficients based on the plurality of loss values ​​and performing sound compensation based on the digital filter coefficients includes: Based on the sorting results of the multiple loss values, the target loss value is determined; The digital filtering coefficients are determined based on the target loss value, and then sent to the frequency division control module. The frequency division control module performs sound compensation based on the digital filtering coefficients. The frequency division control module is used to determine the driving power of the balanced armature speaker and the moving coil speaker based on the digital filtering coefficients.

6. The method according to claim 5, characterized in that, Determining the digital filter coefficients based on the target loss value includes: The digital filter coefficients are determined based on the candidate frequency division frequency corresponding to the target loss value, the boundary frequencies corresponding to the plurality of frequencies, and the time delay information.

7. The method according to claim 1, characterized in that, The earphones are communicatively connected to a mobile device. The step of responding to the earphones being worn by a person, and playing test audio signals through the balanced armature speaker and the dynamic driver, includes: In response to the earphone being worn on a person's ear, the device receives a calibration command sent by the mobile device and controls the balanced armature speaker and the dynamic driver to play the test audio signal.

8. A sound compensation device, characterized in that, A processor for use in headphones, the headphones also including a feedback microphone, a balanced armature speaker, and a dynamic driver, the device comprising: The signal processing module is used to respond to the earphone being worn in the ear by playing a test audio signal through the balanced armature speaker and the dynamic driver, and to collect the ear canal signal of the test audio signal through the feedback microphone; The curve determination module is used to determine the amplitude response curve of the first frequency band based on the test audio signal and the ear canal signal, and to determine the response error curve based on the amplitude response curve. A delay determination module is used to determine the delay information of the ear canal signal based on multiple candidate frequency divisions; The coefficient determination module is used to determine multiple loss values ​​based on the multiple candidate frequency division frequencies, response error curves and time delay information, determine digital filter coefficients based on the multiple loss values, and perform sound compensation based on the digital filter coefficients.

9. An earphone, characterized in that, The headphones include: A moving iron horn, used to play the first frequency band of test audio signals; A dynamic speaker is used to play the second frequency band of the test audio signal, wherein the first frequency band signal and the second frequency band signal constitute an ear canal signal; The feedback microphone is used to collect ear canal signals of the test audio signal; One or more processors; Storage device for storing one or more programs for testing audio signals and target amplitude response curves; When the one or more programs are executed by the one or more processors, the one or more processors implement the sound compensation method as described in any one of claims 1-7.

10. A storage medium containing computer-executable instructions, characterized in that, The computer-executable instructions, when executed by a computer processor, are used to perform the sound compensation method as described in any one of claims 1-7.

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