Patent Publication Number: US-2022225026-A1

Title: Method and Apparatus for Improving Sound Quality of Speaker

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
     This is a continuation of International Patent Application No. PCT/CN2020/110632 filed on Aug. 21, 2020, which claims priority to Chinese Patent Application No. 201910883760.4 filed on Sep. 18, 2019. The disclosures of the aforementioned applications are hereby incorporated by reference in their entireties. 
    
    
     TECHNICAL FIELD 
     Embodiments of this application relate to the field of media technologies, and in particular, to a method and an apparatus for improving sound quality of a speaker. 
     BACKGROUND 
     A speaker is increasingly widely used in a portable terminal device. For example, the speaker is used to play music or a video, make a hands-free call, play a ringtone of a mobile phone, or the like. Sound quality of the speaker is an important indicator of performance of the speaker, and the sound quality of the speaker directly affects a subjective experience. 
     For some widely used micro speakers, nonlinearity of the speaker becomes more obvious. Consequently, obvious distortion is generated in an output result, sound quality of the speaker becomes poor, and an auditory experience is affected. Currently, a nonlinear compensation technology may be used to perform nonlinear compensation on an input signal of the speaker, to reduce signal distortion. Specifically, a nonlinear parameter (for example, a force factor, mechanical stiffness, inductance, or damping) of the speaker may be obtained in a related nonlinear parameter identification method, and then nonlinear compensation is performed on the input signal based on the nonlinear parameter of the speaker. 
     However, because a status of the speaker may change, the nonlinear parameter obtained in the foregoing method differs from an actual nonlinear parameter. Consequently, there is a poor effect of performing nonlinear compensation on the speaker, and therefore, the sound quality of the speaker may be poor. 
     SUMMARY 
     Embodiments of this application provide a speaker improvement method and apparatus, to effectively improve a sound effect of the speaker. 
     To achieve the foregoing objective, the following technical solutions are used in embodiments of this application. 
     According to a first aspect, an embodiment of this application provides a method for improving sound quality of a speaker. The method includes: performing interpolation on a second nonlinear parameter of the speaker based on direct current resistance of the speaker, to obtain a third nonlinear parameter of the speaker, where the second nonlinear parameter is a nonlinear parameter preconfigured in the speaker; performing signal compensation on a first input signal of the speaker based on the third nonlinear parameter, to obtain a compensated first input signal; and performing filtering on the compensated first input signal, to obtain an output signal of the speaker. 
     Optionally, the second nonlinear parameter may be obtained by adjusting a first nonlinear parameter of the speaker. 
     A nonlinear parameter of the speaker is a nonlinear characteristic of the speaker, and is an inherent nonlinear characteristic brought by a hardware structure of the speaker (for example, a structural feature such as a small size and large displacement of the speaker). The nonlinear parameter of the speaker includes a force factor, mechanical stiffness, inductance, damping, or the like of the speaker. In this embodiment of this application, the nonlinear parameter of the speaker (for example, the first nonlinear parameter, the second nonlinear parameter, and the third nonlinear parameter) includes at least one of the force factor, the mechanical stiffness, the inductance, and the damping of the speaker. 
     According to the method for improving sound quality of a speaker in this embodiment of this application, the third nonlinear parameter is obtained by performing interpolation on the second nonlinear parameter based on the direct current resistance of the speaker. The nonlinear parameter (that is, the third nonlinear parameter) is a nonlinear parameter corresponding to a current working state of the speaker, in other words, is a real-time nonlinear parameter. The nonlinear parameter has high accuracy. Therefore, signal compensation can be more effectively performed on the first input signal of the speaker based on the third nonlinear parameter, and filtering is performed on the compensated first input signal, to further reduce signal distortion. In this way, the sound quality of the speaker can be effectively improved. 
     In a possible implementation, the method for performing interpolation on a second nonlinear parameter based on direct current resistance of the speaker, to obtain a third nonlinear parameter of the speaker may specifically include: determining a temperature of a coil of the speaker based on the direct current resistance of the speaker; and performing interpolation on the second nonlinear parameter based on the temperature of the coil, to obtain the third nonlinear parameter. 
     In this embodiment of this application, the output signal of the speaker, the second nonlinear parameter, and a linear parameter of the speaker are input into a speaker model, to obtain the direct current resistance of the speaker and a current, displacement, and a velocity that exist at a current moment. The current, the displacement, and the velocity that exist at the current moment are used to perform signal compensation on a next signal in the input signal. 
     In this embodiment of this application, a relationship between the temperature of the coil (which may also be referred to as a temperature of a voice coil) of the speaker and the direct current resistance of the coil of the speaker is as follows: 
     
       
         
           
             T 
             = 
             
               
                 
                   1 
                   η 
                 
                 ⁢ 
                 
                   ( 
                   
                     
                       R 
                       
                         R 
                         0 
                       
                     
                     - 
                     1 
                   
                   ) 
                 
               
               + 
               
                 2 
                 ⁢ 
                 5 
               
             
           
         
       
     
     Herein, T is the temperature of the coil of the speaker, R is the direct current resistance of the coil of the speaker, η is a temperature rise coefficient, R o  is direct current resistance of the coil that corresponds to a calibration temperature, and the temperature of the voice coil is usually calibrated at 25 degrees Celsius. 
     After the direct current resistance of the speaker is obtained, the temperature of the coil of the speaker may be obtained based on the foregoing formula. 
     In this embodiment of this application, characteristic curves of the nonlinear parameter that exist when the temperature of the coil of the speaker has different temperature values may be obtained, and then linear interpolation is performed on the characteristic curve of the nonlinear parameter based on the temperature of the coil of the speaker, a temperature threshold 1, and a temperature threshold 2, to obtain a target characteristic curve (the target characteristic curve may be understood as an estimation result of a characteristic curve of the third nonlinear parameter). The temperature threshold 2 is greater than the temperature threshold 1. 
     For example, the temperature of the coil of the speaker is denoted as T, the temperature threshold 1 is denoted as T min , and the temperature threshold 2 is denoted as T max . 
     If T&lt;T min , a characteristic curve corresponding to T min  is used as the target characteristic curve. 
     If T&gt;T max , a characteristic curve corresponding to T max  is used as the target characteristic curve. 
     If T min ≤T≤T max , linear interpolation is performed on a characteristic curve corresponding to T min  and a characteristic curve corresponding to T max  based on the temperature of the coil of the speaker, to generate the target characteristic curve. 
     Finally, polynomial fitting is performed on the target characteristic curve, to obtain each coefficient of a polynomial corresponding to the target characteristic curve. Each coefficient is in a one-to-one correspondence with a nonlinear parameter. Therefore, the third nonlinear parameter may be determined based on each coefficient of the polynomial. 
     In a possible implementation, in this embodiment of this application, the compensated first input signal is recorded as a first signal, and the process of performing filtering on the compensated first input signal includes: performing filtering on the first signal by using a wavetrap, to obtain a second signal; calculating a difference between the first signal and the second signal, to obtain a third signal; multiplying a filtering gain by the third signal, to obtain a fourth signal; calculating a difference between the first signal and the fourth signal, to obtain a fifth signal; and using the fifth signal as an output signal of the speaker. 
     In a possible implementation, the method for improving sound quality of a speaker in this embodiment of this application may further include: generating a filtering gain. The filtering gain is used to perform filtering on the compensated first input signal. 
     In this embodiment of this application, the filtering gain may be generated based on the input signal. Specifically, the generating a filtering gain may include S 1  and S 2 . 
     S 1 : Determine a maximum value of an absolute voltage value in a current frame. 
     It should be understood that the first input signal includes a plurality of signal frames, each signal frame includes a plurality of input voltages, and a voltage with a largest absolute value in the plurality of input voltages is the maximum value of the absolute voltage value in the current frame. 
     S 2 : Determine the filtering gain based on the maximum value of the absolute voltage value. 
     In this embodiment of this application, the maximum value of the absolute voltage value is denoted as U max , the filtering gain is denoted as α, and the determining the filtering gain based on the maximum value of the absolute voltage value includes: 
     
       
         
           
             
               
                 When 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 
                   U 
                   max 
                 
               
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                 U 
                 lowlimit 
               
             
             , 
             
               
 
             
             ⁢ 
             
               α 
               = 
               
                 
                   α 
                   buffer 
                 
                 * 
                 
                   α 
                   smooth 
                 
               
             
             , 
           
         
       
     
     where 
     U lowlimit  is a voltage control lower limit, α buffer  is a filtering gain corresponding to a previous frame, α smooth  is a smoothing coefficient of the filtering gain, and * represents convolution; 
     
       
         
           
             
               
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                 ⁢ 
                 
                     
                 
                 ⁢ 
                 
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               α 
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                         U 
                         max 
                       
                       - 
                       
                         U 
                         lowlimit 
                       
                     
                     
                       
                         U 
                         uplimit 
                       
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     where 
     U uplimit  is a voltage control upper limit, α uplimit  is a control upper limit of the filtering gain, and α lowlimit  is a control lower limit of the filtering gain; or 
     
       
         
           
             
               
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                 ⁢ 
                 
                     
                 
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                   max 
                 
               
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     In a possible implementation, the first nonlinear parameter of the speaker may be adjusted based on an acoustic signal of the speaker or a displacement signal of the speaker, to obtain the second nonlinear parameter of the speaker. 
     In this embodiment of this application, the first nonlinear parameter is a nonlinear parameter of the speaker in an original state (which can be understood as a state in which the speaker is manufactured and is not put into use). Before delivery of the speaker, the first nonlinear parameter of the speaker is first adjusted, a parameter of the speaker is adjusted from the first nonlinear parameter to the second nonlinear parameter, and then the speaker is delivered and is put into use. 
     In a possible implementation, the method for adjusting the first nonlinear parameter of the speaker based on the acoustic signal of the speaker or the displacement signal of the speaker, to obtain the second nonlinear parameter of the speaker may specifically include: determining a target to-be-adjusted parameter from the first nonlinear parameter of the speaker based on the acoustic signal of the speaker or the displacement signal of the speaker; and calibrating the target to-be-adjusted parameter in the first nonlinear parameter of the speaker based on a target direction and the target step, to obtain the second nonlinear parameter. 
     In this embodiment of this application, the target direction may include a forward direction and a reverse direction. The forward direction may be defined as a direction in which the nonlinear parameter is increased, and the reverse direction may be defined as a direction in which the nonlinear parameter is decreased. The target direction may be specifically defined based on an actual requirement. This is not limited in this embodiment of this application. 
     The target step represents an adjustment (increase or decrease) amplitude of the nonlinear parameter, and the target step may include a target step corresponding to the forward direction and a target step corresponding to the reverse direction. 
     Optionally, the target step corresponding to the forward direction may be the same as or different from the target step corresponding to the reverse direction. This is not specifically limited in this embodiment of this application. 
     In a possible implementation, the method for determining a target to-be-adjusted parameter from the first nonlinear parameter of the speaker based on the acoustic signal of the speaker or the displacement signal of the speaker may include: performing Fourier transform on the acoustic signal of the speaker or the displacement signal of the speaker, to obtain harmonic distortion; determining a candidate to-be-adjusted parameter from the first nonlinear parameter of the speaker based on the harmonic distortion; and determining the target to-be-adjusted parameter from the candidate to-be-adjusted parameter. 
     In this embodiment of this application, harmonic distortion in a Fourier transform result corresponding to an acoustic signal or a displacement signal collected after signal compensation is performed on the input signal is referred to as first harmonic distortion, and harmonic distortion in a Fourier transform result corresponding to an acoustic signal or a displacement signal collected before compensation is performed on the input signal is referred to as second harmonic distortion. Therefore, the determining a candidate to-be-adjusted parameter from the first nonlinear parameter of the speaker may include: determining the candidate to-be-adjusted parameter based on the first harmonic distortion and the second harmonic distortion. The determining the candidate to-be-adjusted parameter based on the first harmonic distortion and the second harmonic distortion specifically includes: determining a ratio of each order of harmonic distortion in the second harmonic distortion to each order of harmonic distortion corresponding to the first harmonic distortion; and determining, as the candidate to-be-adjusted parameter, a nonlinear parameter corresponding to each ratio that is of harmonic distortion and that is greater than a preset threshold. 
     After the candidate to-be-adjusted parameter is determined, a convergence error of each nonlinear parameter in the candidate to-be-adjusted parameter is obtained, the convergence error of each nonlinear parameter in the candidate to-be-adjusted parameter is compared with a preset error threshold corresponding to each nonlinear parameter, and each nonlinear parameter whose convergence error is greater than the error threshold in the candidate to-be-adjusted parameter is determined as the target to-be-adjusted parameter. 
     In a possible implementation, the method for improving sound quality of a speaker in this embodiment of this application may further include: obtaining the acoustic signal of the speaker; or obtaining the displacement signal of the speaker. 
     In a possible implementation, the method for improving sound quality of the speaker in this embodiment of this application may further include: determining displacement of the speaker; determining a signal control gain of the speaker based on the displacement of the speaker and a preset displacement threshold; and performing gain control on a second input signal of the speaker based on the signal control gain of the speaker, to obtain the first input signal. 
     In this embodiment of this application, displacement protection may be performed on the speaker by determining the displacement of the speaker; determining the signal control gain of the speaker based on the displacement of the speaker and the preset displacement threshold; and performing gain control on the second input signal of the speaker based on the signal control gain of the speaker, to reduce a gain of the input signal, avoid an abrupt change in a speaker volume of the speaker, ensure that the displacement of the speaker does not exceed an upper safety limit, and improve a sound effect of the speaker. 
     Further, a signal obtained after gain control is performed on the second input signal of the speaker is used as the first input signal, and signal compensation is performed based on this, so that a signal compensation effect can be improved, and a sound effect of the speaker can be further improved. 
     In a possible implementation, the method for determining displacement of the speaker may include: performing first displacement conversion on the second input signal, to obtain a maximum value of first predicted displacement and an effective value of the first predicted displacement; determining a displacement correction gain; and determining the displacement of the speaker based on the maximum value of the first predicted displacement and the displacement correction gain. 
     In this embodiment of this application, a displacement transfer function of the speaker is used when the first displacement conversion is performed, and the displacement transfer function may be updated based on a nonlinear parameter obtained in real time. Specifically, a feedback signal (a feedback voltage) that is of the speaker and that exists at a previous moment is input into a linear parameter identification model, to obtain a linear parameter of the speaker, and further, the displacement transfer function is updated based on the linear parameter. 
     In a possible implementation, the method for improving sound quality of a speaker in this embodiment of this application may further include: performing second displacement conversion on the feedback signal of the speaker, to obtain an effective value of second predicted displacement. 
     In this embodiment of this application, an induced electromotive force model of the speaker is used when the second displacement conversion is performed, and the induced electromotive force model may be updated based on a nonlinear parameter obtained in real time. Specifically, a feedback signal (the feedback voltage) that is of the speaker and that exists at a previous moment is input into a linear parameter identification model, to obtain a linear parameter of the speaker, and further, the induced electromotive force model is updated based on the linear parameter. 
     In a possible implementation, the method for determining a displacement correction gain may specifically include: determining the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the second predicted displacement. 
     In a possible implementation, the method for determining the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the second predicted displacement may specifically include: determining an effective value of third predicted displacement based on the effective value of the first predicted displacement and the effective value of the second predicted displacement; and determining the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the third predicted displacement. 
     In this embodiment of this application, the displacement correction gain G c (t n ) may be as follows: 
     
       
         
           
             
               
                 G 
                 c 
               
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                     ⁢ 
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             , 
             
               
                 
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                     ⁢ 
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                 KalmanFilter 
                 ⁡ 
                 
                   [ 
                   
                     
                       
                         X 
                         
                           mean 
                           ⁢ 
                           _ 
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                       ⁡ 
                       
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     X mean_ts (t n ) represents the effective value of the first predicted displacement, X mean_enf (t n ) represents the effective value of the second predicted displacement, X mean_est (t n ) represents calculating the effective value of the third predicted displacement, and KalmanFilter represents a Kalman filter. 
     According to a second aspect, an embodiment of this application provides a method for improving sound quality of a speaker. The method may include: performing first displacement conversion on an input signal of the speaker, to obtain a maximum value of first predicted displacement and an effective value of the first predicted displacement; performing second displacement conversion on a feedback signal of the speaker, to obtain an effective value of second predicted displacement; determining a displacement correction gain based on the effective value of the first predicted displacement and the effective value of the second predicted displacement; determining displacement of the speaker based on the maximum value of the first predicted displacement and the displacement correction gain; determining a signal control gain of the speaker based on the displacement of the speaker and a preset displacement threshold; and performing gain control on the input signal of the speaker based on the signal control gain of the speaker, to obtain an output signal of the speaker. 
     Compared with the conventional technology, in the foregoing method, the displacement of the speaker can be determined in real time, and displacement protection is performed on the speaker based on the determined displacement of the speaker (which indicates to perform gain control on the input signal of the speaker based on the signal control gain determined based on the displacement of the speaker), so that a gain of the input signal is reduced, an abrupt change in a speaker volume of the speaker is avoided, it can be ensured that the displacement of the speaker does not exceed an upper safety limit, and a sound effect of the speaker can be improved. 
     In a possible implementation, the method for determining a displacement correction gain based on the effective value of the first predicted displacement and the effective value of the second predicted displacement specifically includes: determining an effective value of third predicted displacement based on the effective value of the first predicted displacement and the effective value of the second predicted displacement; and determining the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the third predicted displacement. 
     According to a third aspect, an embodiment of this application provides a sound quality improvement apparatus, including an interpolation module, a signal compensation module, and a filtering module. The interpolation module is configured to perform interpolation on a second nonlinear parameter of a speaker based on direct current resistance of the speaker, to obtain a third nonlinear parameter of the speaker. The second nonlinear parameter is a nonlinear parameter preconfigured in the speaker. The signal compensation module is configured to perform signal compensation on a first input signal of the speaker based on the third nonlinear parameter, to obtain a compensated first input signal. The filtering module is configured to perform filtering on the compensated first input signal, to obtain an output signal of the speaker. 
     In a possible implementation, the interpolation module is specifically configured to: determine a temperature of a coil of the speaker based on the direct current resistance of the speaker, and perform interpolation on the second nonlinear parameter based on the temperature of the coil, to obtain the third nonlinear parameter. 
     In a possible implementation, the sound quality improvement apparatus provided in this embodiment of this application further includes a generation module. The generation module is configured to generate a filtering gain. The filtering gain is used to perform filtering on the compensated first input signal. 
     In a possible implementation, the sound quality improvement apparatus provided in this embodiment of this application further includes a parameter adjustment module. The parameter adjustment module is configured to adjust a first nonlinear parameter of the speaker based on an acoustic signal of the speaker or a displacement signal of the speaker, to obtain the second nonlinear parameter of the speaker. 
     In a possible implementation, the parameter adjustment module is specifically configured to: determine a target to-be-adjusted parameter from the first nonlinear parameter of the speaker based on the acoustic signal of the speaker or the displacement signal of the speaker; and calibrate the target to-be-adjusted parameter in the first nonlinear parameter of the speaker based on a target direction and a target step, to obtain the second nonlinear parameter. 
     In a possible implementation, the parameter adjustment module is specifically configured to: perform Fourier transform on the acoustic signal of the speaker or the displacement signal of the speaker, to obtain harmonic distortion; determine a candidate to-be-adjusted parameter from the first nonlinear parameter of the speaker based on the harmonic distortion; and determine the target to-be-adjusted parameter from the candidate to-be-adjusted parameter. 
     In a possible implementation, the sound quality improvement apparatus provided in this embodiment of this application may further include an obtaining module. The obtaining module is configured to: obtain the acoustic signal of the speaker; or obtain the displacement signal of the speaker. 
     In a possible implementation, the sound quality improvement apparatus provided in this embodiment of this application further includes a displacement determining module, a control gain determining module, and a gain control module. The displacement determining module is configured to determine displacement of the speaker. The control gain determining module is configured to determine a signal control gain of the speaker based on the displacement of the speaker and a preset displacement threshold. The gain control module is configured to perform gain control on a second input signal of the speaker based on the signal control gain of the speaker, to obtain the first input signal. 
     In a possible implementation, the displacement determining module is specifically configured to: perform first displacement conversion on the second input signal, to obtain a maximum value of first predicted displacement and an effective value of the first predicted displacement; determine a displacement correction gain; and determine the displacement of the speaker based on the maximum value of the first predicted displacement and the displacement correction gain. 
     In a possible implementation, the displacement determining module is further configured to perform second displacement conversion on a feedback signal of the speaker, to obtain an effective value of second predicted displacement. 
     In a possible implementation, the displacement determining module is specifically configured to determine the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the second predicted displacement. 
     In a possible implementation, the displacement determining module is specifically configured to: determine an effective value of third predicted displacement based on the effective value of the first predicted displacement and the effective value of the second predicted displacement; and determine the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the third predicted displacement. 
     According to a fourth aspect, an embodiment of this application provides a sound quality improvement apparatus, including a displacement determining module, a control gain determining module, and a gain control module. The displacement determining module is configured to: perform first displacement conversion on an input signal of a speaker, to obtain a maximum value of first predicted displacement and an effective value of the first predicted displacement; perform second displacement conversion on a feedback signal of the speaker, to obtain an effective value of second predicted displacement; determine a displacement correction gain based on the effective value of the first predicted displacement and the effective value of the second predicted displacement; and determine displacement of the speaker based on the maximum value of the first predicted displacement and the displacement correction gain. The control gain determining module is configured to determine a signal control gain of the speaker based on the displacement of the speaker and a preset displacement threshold. The gain control module is configured to perform gain control on the input signal of the speaker based on the signal control gain of the speaker, to obtain an output signal of the speaker. 
     In a possible implementation, the displacement determining module is specifically configured to: determine an effective value of third predicted displacement based on the effective value of the first predicted displacement and the effective value of the second predicted displacement; and determine the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the third predicted displacement. 
     According to a fifth aspect, an embodiment of this application provides a sound quality improvement apparatus, including a processor and a memory coupled to the processor. The memory is configured to store computer instructions. When the apparatus runs, the processor executes the computer instructions stored in the memory, so that the apparatus performs the method for improving sound quality of a speaker according to any one of the first aspect and the possible implementations of the first aspect. 
     According to a sixth aspect, an embodiment of this application provides a sound quality improvement apparatus. The apparatus exists in a product form of a chip. A structure of the apparatus includes a processor and a memory. The memory is configured to be coupled to the processor, the memory is configured to store computer instructions, and the processor is configured to execute the computer instructions stored in the memory, so that the apparatus performs the method for improving sound quality of a speaker according to any one of the first aspect and the possible implementations of the first aspect. 
     According to a seventh aspect, an embodiment of this application provides a computer-readable storage medium. The computer-readable storage medium may include computer instructions, and when the computer instructions run on a computer, a sound quality improvement apparatus is enabled to perform the method for improving sound quality of a speaker according to any one of the first aspect and the possible implementations of the first aspect. 
     According to an eighth aspect, an embodiment of this application provides a sound quality improvement apparatus, including a processor and a memory coupled to the processor. The memory is configured to store computer instructions. When the apparatus runs, the processor executes the computer instructions stored in the memory, so that the apparatus performs the method for improving sound quality of a speaker according to any one of the second aspect and the possible implementations of the second aspect. 
     According to a ninth aspect, an embodiment of this application provides a sound quality improvement apparatus. The apparatus exists in a product form of a chip. A structure of the apparatus includes a processor and a memory. The memory is configured to be coupled to the processor, the memory is configured to store computer instructions, and the processor is configured to execute the computer instructions stored in the memory, so that the apparatus performs the method for improving sound quality of a speaker according to any one of the second aspect and the possible implementations of the second aspect. 
     According to a tenth aspect, an embodiment of this application provides a computer-readable storage medium. The computer-readable storage medium may include computer instructions, and when the computer instructions run on a computer, a sound quality improvement apparatus is enabled to perform the method for improving sound quality of a speaker according to any one of the second aspect and the possible implementations of the second aspect. 
     It should be understood that, for beneficial effects achieved by the technical solutions in the third aspect to the tenth aspect of embodiments of this application and the corresponding possible implementations, refer to the foregoing technical effects of the first aspect and the corresponding possible implementations of the first aspect or the second aspect and the corresponding possible implementations of the second aspect. Details are not described herein again. 
    
    
     
       BRIEF DESCRIPTION OF DRAWINGS 
         FIG. 1  is a schematic diagram of a structure of a mobile phone according to an embodiment of this application; 
         FIG. 2  is a schematic diagram 1 of a speaker improvement method according to an embodiment of this application; 
         FIG. 3  is a schematic diagram 2 of a speaker improvement method according to an embodiment of this application; 
         FIG. 4  is a flow block diagram 1 of a method for improving sound quality of a speaker according to an embodiment of this application; 
         FIG. 5  is a schematic diagram of a method for adjusting a nonlinear parameter of a speaker according to an embodiment of this application; 
         FIG. 6  is a flow block diagram of a method for adjusting a nonlinear parameter of a speaker according to an embodiment of this application; 
         FIG. 7  is a schematic diagram 3 of a speaker improvement method according to an embodiment of this application; 
         FIG. 8  is a flow block diagram 2 of a method for improving sound quality of a speaker according to an embodiment of this application; 
         FIG. 9  is a flow block diagram 3 of a method for improving sound quality of a speaker according to an embodiment of this application; 
         FIG. 10  is a flow block diagram 4 of a method for improving sound quality of a speaker according to an embodiment of this application; 
         FIG. 11  is a schematic diagram 1 of a structure of a sound quality improvement apparatus according to an embodiment of this application; 
         FIG. 12  is a schematic diagram 2 of a structure of a sound quality improvement apparatus according to an embodiment of this application; 
         FIG. 13  is a schematic diagram 3 of a structure of a sound quality improvement apparatus according to an embodiment of this application; and 
         FIG. 14  is a schematic diagram 4 of a structure of a sound quality improvement apparatus according to an embodiment of this application. 
     
    
    
     DESCRIPTION OF EMBODIMENTS 
     The term “and/or” in this specification describes only an association relationship for describing associated objects and represents that three relationships may exist. For example, A and/or B may represent the following three cases: Only A exists, both A and B exist, and only B exists. 
     In the specification and claims in embodiments of this application, the terms “first”, “second”, and the like are intended to distinguish between different objects but do not indicate a particular order of the objects. For example, a first nonlinear parameter, a second nonlinear parameter, and the like are used to distinguish between different nonlinear parameters, but are not used to describe specific orders of the nonlinear parameters. A first input signal and a second input signal are used to distinguish between different input signals, but are not used to describe particular orders of the input signals. 
     In embodiments of this application, the word such as “example” or “for example” is used to represent giving an example, an illustration, or a description. Any embodiment or design scheme described as “example” or “for example” in embodiments of this application should not be explained as being more preferred or having more advantages than another embodiment or design scheme. To be exact, use of the expressions like “example” and “for example” is intended to give a specific presentation of a related concept. 
     In the description of embodiments of this application, unless otherwise stated, “a plurality of” means two or more than two. For example, a plurality of processing units refer to two or more processing units, and a plurality of systems refer to two or more systems. 
     First, some basic knowledge and concepts in a method and an apparatus for improving sound quality of a speaker in the embodiments of this application are explained and described. 
     Currently, factors that affect the sound quality of the speaker may include a nonlinear factor of the speaker and displacement of the speaker (the following displacement of the speaker is displacement of a diaphragm of the speaker). 
     Impact of the nonlinear factor of speaker on the sound quality: Nonlinearity of the speaker is a phenomenon that output sound quality of the speaker is distorted due to a hardware structure of the speaker (for example, a structural feature such as a small size and large displacement of the speaker), and may be referred to as nonlinear distortion. Especially, when a signal of high amplitude is input into the speaker, the nonlinearity of the speaker is more obvious, and there may be too much distortion in an output signal, affecting an auditory experience. 
     Impact of the displacement of the speaker on the sound quality: When the displacement of the diaphragm of the speaker is too large, the diaphragm of the speaker may be in a physical collision, causing noise and even mechanical damage to the speaker. 
     In the embodiments of this application, a nonlinear parameter of the speaker may be used to compensate for the nonlinear distortion caused by hardware of the speaker, to improve the sound quality of the speaker. In addition, the displacement of the speaker can be controlled, to improve the sound quality of the speaker. 
     It should be understood that the nonlinear parameter of the speaker may include but is not limited to the following parameters: 
     A force factor BL(x) is a force factor of a magnetic circuit system of the speaker. 
     Mechanical stiffness Kms(x) is stiffness of a suspension system of the speaker, and Kms(x) may include different coefficients such as a first order coefficient, a second order coefficient, and a third order coefficient. 
     Inductance Le(x) is inductance of a coil of the speaker. 
     Damping Rm(v) is a damping coefficient of the speaker, and Rm(v) may include different coefficients such as a first order coefficient, a second order coefficient, and a third order coefficient. 
     Herein, x is the displacement of the diaphragm of the speaker, and v is a moving velocity of the diaphragm of the speaker. 
     It should be noted that, the nonlinear parameter of the speaker may change in different working states of the speaker. For example, when the coil of the speaker is at different temperatures, a value of Kms(x) may change, and a value of Rm(v) may also change. In other words, there are different values of Kms(x) at different temperatures, and there are different values of Rm(v) at different temperatures. 
     Input signals of the speaker include M (M is a positive integer greater than or equal to 1) digital signals and n corresponding voltage values (which may also be referred to as n points). For example, the input signals U in =[U in (1), U in (2), . . . , U in (n), . . . , and U in (M)]. In the embodiments of this application, processing the input signals is to sequentially process all digital signals in the input signals. For ease of description, a moment at which the nth digital signal is input is denoted as t n , and an input signal corresponding to the moment t n  is denoted as U in (n) or U in (t n ). 
     Based on the problem existing in the background, the embodiments of this application provide a method and an apparatus for improving sound quality of a speaker. A nonlinear parameter (that is, a third nonlinear parameter in the following embodiments) of the speaker may be obtained in an interpolation method based on direct current resistance of the speaker, nonlinear compensation is performed on an input signal of the speaker based on the nonlinear parameter, and filtering is performed on the compensated input signal, to obtain an output signal of the speaker. Therefore, nonlinear compensation is performed on the speaker, and sound quality of the speaker can be improved. 
     The method and the apparatus for improving sound quality of a speaker in the embodiments of this application may be applied to a terminal device having a speaker function, for example, an electronic device provided with a speaker such as a mobile phone, a tablet computer, a notebook computer, a smart speaker, or a television. For example, the technical solutions provided in the embodiments of this application may be used in a scenario such as playing music and a movie in a speaker mode (including monaural, binaural, and quadraphonic playing), a hands-free call (including an operator call, an Internet call, or the like), a ringtone of a mobile phone (including a speaker mode and a headphone mode), or playing a game in a speaker mode, to improve the sound quality of the speaker, and improve a subjective experience of a user. 
     For example, the terminal device is a mobile phone.  FIG. 1  is a schematic diagram of a structure of a mobile phone  100 . The mobile phone  100  may include a processor  110 , an external memory interface  120 , an internal memory  121 , a universal serial bus (USB) interface  130 , a charging management module  140 , a power management module  141 , a battery  142 , an antenna 1, an antenna 2, a mobile communications module  150 , a wireless communications module  160 , an audio module  170 , a speaker  170 A, a receiver  170 B, a microphone  170 C, a headset jack  170 D, a sensor module  180 , a button  190 , a motor  191 , an indicator  192 , a camera  193 , a display  194 , a subscriber identification module (SIM) card interface  195 , and the like. The sensor module  180  may include a pressure sensor  180 A, a gyro sensor  180 B, a barometric pressure sensor  180 C, a magnetic sensor  180 D, an acceleration sensor  180 E, a distance sensor  180 F, an optical proximity sensor  180 G, a fingerprint sensor  180 H, a temperature sensor  180 J, a touch sensor  180 K, an ambient light sensor  180 L, a bone conduction sensor  180 M, and the like. 
     It may be understood that the structure shown in this embodiment of this application does not constitute a specific limitation on the mobile phone  100 . In some other embodiments of this application, the mobile phone  100  may include more or fewer components than those shown in the figure, or some components may be combined, or some components may be split, or there may be a different component layout. The components shown in the figure may be implemented by hardware, software, or a combination of software and hardware. 
     The processor  110  may include one or more processing units. For example, the processor  110  may include an application processor (AP), a modem processor, a graphics processing unit (GPU), an image signal processor (ISP), a controller, a memory, a video codec, a digital signal processor (DSP), a baseband processor, and/or a neural-network processing unit (NPU). Different processing units may be independent devices, or may be integrated into one or more processors. 
     The controller may be a nerve center and a command center of the mobile phone  100 . The controller may generate an operation control signal based on an instruction operation code and a time sequence signal, to control to fetch instructions and execute instructions. 
     A memory may be further disposed in the processor  110 , and is configured to store instructions and data. In some embodiments, the memory in the processor  110  is a cache. The memory may store instructions or data that has just been used or is cyclically used by the processor  110 . If the processor  110  needs to use the instructions or the data again, the processor may directly invoke the instructions or the data from the memory. This avoids repeated access and reduces waiting time of the processor  110 . Therefore, system efficiency is improved. 
     In some embodiments, the processor  110  may include one or more interfaces. The interface may include an inter-integrated circuit (I2C) interface, an inter-integrated circuit sound (I2S) interface, a pulse code modulation (PCM) interface, a universal asynchronous receiver/transmitter (UART) interface, a mobile industry processor interface (MIPI), a general-purpose input/output (GPIO) interface, a SIM interface, a USB interface, and/or the like. 
     The I2C interface is a two-way synchronous serial bus, including a serial data line (SDA) and a serial clock line (SCL). In some embodiments, the processor  110  may include a plurality of groups of I2C buses. The processor  110  may be separately coupled to the touch sensor  180 K, a charger, a flash, the camera  193 , and the like through different I2C bus interfaces. For example, the processor  110  may be coupled to the touch sensor  180 K through the I2C interface, so that the processor  110  communicates with the touch sensor  180 K through the I2C bus interface, to implement a touch function of the mobile phone  100 . 
     The I2S interface may be configured to perform audio communication. In some embodiments, the processor  110  may include a plurality of groups of I2S buses. The processor  110  may be coupled to the audio module  170  through the I2S bus, to implement communication between the processor  110  and the audio module  170 . In some embodiments, the audio module  170  may transfer an audio signal to the wireless communications module  160  through the I2S interface, to implement a function of answering a call by using a Bluetooth headset. 
     The PCM interface may also be configured to perform audio communication, and sample, quantize, and code an analog signal. In some embodiments, the audio module  170  may be coupled to the wireless communications module  160  through a PCM bus interface. In some embodiments, the audio module  170  may alternatively transfer an audio signal to the wireless communications module  160  through the PCM interface, to implement a function of answering a call by using a Bluetooth headset. Both the I2S interface and the PCM interface may be configured to perform audio communication. 
     The UART interface is a universal serial data bus, and is configured to perform asynchronous communication. The bus may be a two-way communications bus. The bus switches to-be-transmitted data between serial communication and parallel communication. In some embodiments, the UART interface is usually configured to connect the processor  110  to the wireless communications module  160 . For example, the processor  110  communicates with a Bluetooth module in the wireless communications module  160  through the UART interface, to implement a Bluetooth function. In some embodiments, the audio module  170  may transfer an audio signal to the wireless communications module  160  through the UART interface, to implement a function of playing music by using a Bluetooth headset. 
     The MIPI interface may be configured to connect the processor  110  to a peripheral component such as the display  194  or the camera  193 . The MIPI interface includes a camera serial interface (CSI), a display serial interface (DSI), or the like. In some embodiments, the processor  110  communicates with the camera  193  through the CSI interface, to implement a photographing function of the mobile phone  100 . The processor  110  communicates with the display  194  through the DSI interface, to implement a display function of the mobile phone  100 . 
     The GPIO interface may be configured by using software. The GPIO interface may be configured as a control signal or a data signal. In some embodiments, the GPIO interface may be configured to connect the processor  110  to the camera  193 , the display  194 , the wireless communications module  160 , the audio module  170 , the sensor module  180 , and the like. The GPIO interface may alternatively be configured as an I2C interface, an I2S interface, a UART interface, an MIPI interface, or the like. 
     The USB interface  130  is an interface that complies with a USB standard specification, and may be specifically a mini USB interface, a micro USB interface, a USB Type-C interface, or the like. The USB interface  130  may be configured to connect to the charger to charge the mobile phone  100 , or may be configured to transmit data between the mobile phone  100  and a peripheral device, or may be configured to connect to a headset, to play audio by using the headset. The interface may be further configured to connect to another electronic device such as an AR device. 
     It may be understood that an interface connection relationship between the modules shown in this embodiment of this application is merely an example for description, and does not constitute a limitation on the structure of the mobile phone  100 . In some other embodiments of this application, the mobile phone  100  may alternatively use an interface connection manner different from that in the foregoing embodiment, or use a combination of a plurality of interface connection manners. 
     The charging management module  140  is configured to receive a charging input from the charger. The charger may be a wireless charger or a wired charger. In some embodiments in which wired charging is used, the charging management module  140  may receive a charging input from the wired charger through the USB interface  130 . In some embodiments of wireless charging, the charging management module  140  may receive a wireless charging input by using a wireless charging coil of the mobile phone  100 . The charging management module  140  may further supply power to the electronic device by using the power management module  141  when the battery  142  is charged. 
     The power management module  141  is configured to connect to the battery  142 , the charging management module  140 , and the processor  110 . The power management module  141  receives an input of the battery  142  and/or the charging management module  140 , and supplies power to the processor  110 , the internal memory  121 , an external memory, the display  194 , the camera  193 , the wireless communications module  160 , and the like. The power management module  141  may be further configured to monitor parameters such as a battery capacity, a battery cycle count, and a battery health status (electric leakage or impedance). In some other embodiments, the power management module  141  may alternatively be disposed in the processor  110 . In some other embodiments, the power management module  141  and the charging management module  140  may alternatively be disposed in a same device. 
     A wireless communication function of the mobile phone  100  may be implemented by using the antenna 1, the antenna 2, the mobile communications module  150 , the wireless communications module  160 , the modem processor, the baseband processor, and the like. 
     The antenna 1 and the antenna 2 are configured to transmit and receive electromagnetic wave signals. Each antenna in the mobile phone  100  may be configured to cover one or more communication frequency bands. Different antennas may be multiplexed to improve antenna utilization. For example, the antenna 1 may be multiplexed as a diversity antenna in a wireless local area network. In some other embodiments, an antenna may be used in combination with a tuning switch. 
     The mobile communications module  150  may provide a solution, applied to the mobile phone  100 , for wireless communication including 2G, 3G, 4G, 5G, or the like. The mobile communications module  150  may include at least one filter, a switch, a power amplifier, a low-noise amplifier (LNA), and the like. The mobile communications module  150  may receive an electromagnetic wave through the antenna 1, perform processing such as filtering and amplification on the received electromagnetic wave, and transmit a processed electromagnetic wave to the modem processor for demodulation. The mobile communications module  150  may further amplify a signal modulated by the modem processor, and convert the signal to an electromagnetic wave for radiation through the antenna 1. In some embodiments, at least some function modules of the mobile communications module  150  may be disposed in the processor  110 . In some embodiments, at least some function modules in the mobile communications module  150  may be disposed in a same device as at least some modules in the processor  110 . 
     The modem processor may include a modulator and a demodulator. The modulator is configured to modulate a to-be-sent low-frequency baseband signal into a medium/high-frequency signal. The demodulator is configured to demodulate a received electromagnetic wave signal into a low-frequency baseband signal. Then, the demodulator transmits the low-frequency baseband signal obtained through demodulation to the baseband processor for processing. The baseband processor processes the low-frequency baseband signal, and then transmits a processed signal to the application processor. The application processor outputs a sound signal by using an audio device (which is not limited to the speaker  170 A, the receiver  170 B, or the like), or displays an image or a video on the display  194 . In some embodiments, the modem processor may be an independent component. In some other embodiments, the modem processor may be independent of the processor  110 , and is disposed in a same device as the mobile communications module  150  or another function module. 
     The wireless communications module  160  may provide a solution, applied to the mobile phone  100 , for wireless communication including a wireless local area network (WLAN) (for example, a Wi-Fi network), BLUETOOTH (BT), a global navigation satellite system (GNSS), frequency modulation (FM), near field communication (NFC), an infrared (IR) technology, and the like. The wireless communications module  160  may be one or more components integrating at least one communications processing module. The wireless communications module  160  receives an electromagnetic wave through the antenna 2, performs frequency modulation and filtering processing on an electromagnetic wave signal, and sends a processed signal to the processor  110 . The wireless communications module  160  may further receive a to-be-sent signal from the processor  110 , perform frequency modulation and amplification on the signal, and convert the signal into an electromagnetic wave for radiation through the antenna 2. 
     In some embodiments, the antenna 1 and the mobile communications module  150  of the mobile phone  100  are coupled, and the antenna 2 and the wireless communications module  160  of the mobile phone  100  are coupled, so that the mobile phone  100  can communicate with a network and another device by using a wireless communications technology. The wireless communications technology may include a Global System for Mobile Communications (GSM), a general packet radio service (GPRS), code-division multiple access (CDMA), wideband code-division multiple access (WCDMA), time-division code-division multiple access (TD-CDMA), Long-Term Evolution (LTE), BT, a GNSS, a WLAN, NFC, FM, an IR technology, and/or the like. The GNSS may include a global positioning system (GPS), a global navigation satellite system (GLONASS), a BeiDou navigation satellite system (BDS), a quasi-zenith satellite system (QZSS), and/or a satellite based augmentation system (SBAS). 
     The mobile phone  100  implements a display function by using the GPU, the display  194 , the application processor, and the like. The GPU is a microprocessor for image processing, and is connected to the display  194  and the application processor. The GPU is configured to: perform mathematical and geometric calculation, and render an image. The processor  110  may include one or more GPUs that execute program instructions to generate or change display information. 
     The display  194  is configured to display an image, a video, and the like. The display  194  includes a display panel. The display panel may be a liquid crystal display (LCD), an organic light-emitting diode (OLED), an active-matrix organic light-emitting diode (AMOLED), a flexible light-emitting diode (FLED), a mini-LED, a micro-LED, a micro-OLED, a quantum dot light emitting diode (QLED), or the like. In some embodiments, the mobile phone  100  may include one or N displays  194 , where N is a positive integer greater than 1. 
     The mobile phone  100  may implement a photographing function by using the ISP, the camera  193 , the video codec, the GPU, the display  194 , the application processor, and the like. 
     The ISP is configured to process data fed back by the camera  193 . For example, during photographing, a shutter is pressed, and light is transmitted to a photosensitive element of a camera through a lens. An optical signal is converted into an electrical signal. The photosensitive element of the camera transmits the electrical signal to the ISP for processing, and converts the electrical signal into a visible image. The ISP may further perform algorithm optimization on noise, brightness, and complexion of the image. The ISP may further optimize parameters such as exposure and a color temperature of a shooting scenario. In some embodiments, the ISP may be disposed in the camera  193 . 
     The camera  193  is configured to capture a static image or a video. An optical image of an object is generated through the lens, and is projected to the photosensitive element. The photosensitive element may be a charge coupled device (CCD) or a complementary metal-oxide-semiconductor (CMOS) photoelectric transistor. The photosensitive element converts an optical signal into an electrical signal, and then transmits the electrical signal to the ISP for converting the electrical signal into a digital image signal. The ISP outputs the digital image signal to the DSP for processing. The DSP converts the digital image signal into a standard image signal in an RGB format, a YUV format, or the like. In some embodiments, the mobile phone  100  may include one or N cameras  193 , where N is a positive integer greater than 1. 
     The digital signal processor is configured to process a digital signal, and may process another digital signal (for example, an audio signal) in addition to a digital image signal. For example, when the mobile phone  100  selects a frequency, the digital signal processor is configured to perform Fourier transform and the like on frequency energy. 
     The video codec is configured to: compress or decompress a digital video. The mobile phone  100  may support one or more video codecs. In this way, the mobile phone  100  can play or record videos in a plurality of coding formats, for example, moving picture experts group (moving picture experts group, MPEG)-1, MPEG-2, MPEG-3, and MPEG-4. 
     The NPU is a neural-network (NN) computing processor that rapidly processes input information by referring to a structure of a biological neural network, for example, by referring to a transfer mode between human brain neurons, and can further perform self-learning continuously. Applications such as intelligent cognition of the mobile phone  100 , for example, image recognition, facial recognition, voice recognition, and text understanding, can be implemented by using the NPU. 
     The external memory interface  120  may be configured to connect to an external memory card such as a micro SD card, to extend a storage capability of the mobile phone  100 . The external memory card communicates with the processor  110  through the external memory interface  120 , to implement a data storage function. For example, files such as music and videos are stored in the external memory card. 
     The internal memory  121  may be configured to store computer-executable program code. The executable program code includes instructions. The processor  110  runs the instructions stored in the internal memory  121 , to perform various function applications of the mobile phone  100  and data processing. The internal memory  121  may include a program storage area and a data storage area. The program storage area may store an operating system, an application required by at least one function (for example, a sound playing function or an image playing function), and the like. The data storage area may store data (for example, audio data or an address book) created during use of the mobile phone  100 , and the like. In addition, the internal memory  121  may include a high-speed random access memory, or may include a nonvolatile memory, for example, at least one magnetic disk storage device, a flash memory, or a universal flash storage (UFS). 
     The mobile phone  100  may implement an audio function such as music playing or recording by using the audio module  170 , the speaker  170 A, the receiver  170 B, the microphone  170 C, the headset jack  170 D, the application processor, and the like. 
     The audio module  170  is configured to convert digital audio information into an analog audio signal for output, and is also configured to convert an analog audio input into a digital audio signal. The audio module  170  may be further configured to encode and decode audio signals. In some embodiments, the audio module  170  may be disposed in the processor  110 , or some function modules of the audio module  170  are disposed in the processor  110 . 
     The speaker  170 A, also referred to as a “horn”, is configured to convert an audio electrical signal into a sound signal. The mobile phone  100  may listen to music or answer a hands-free call through the speaker  170 A. 
     The receiver  170 B, also referred to as an “earpiece”, is configured to convert an audio electrical signal into a sound signal. When a call is answered or voice information is received by using the mobile phone  100 , the receiver  170 B may be put close to a human ear to listen to a voice. 
     The microphone  170 C, also referred to as a “mike” or a “mic”, is configured to convert a sound signal into an electrical signal. When making a call or sending voice information, a user may make a sound by moving the mouth of the user close to the microphone  170 C to enter a sound signal to the microphone  170 C. At least one microphone  170 C may be disposed in the mobile phone  100 . In some other embodiments, two microphones  170 C may be disposed in the mobile phone  100 , to collect a sound signal and further implement a noise reduction function. In some other embodiments, three, four, or more microphones  170 C may alternatively be disposed in the mobile phone  100 , to collect a sound signal, reduce noise, further identify a sound source, implement a directional recording function, and the like. 
     The headset jack  170 D is configured to connect to a wired headset. The headset jack  170 D may be the USB interface  130  or a 3.5 mm open mobile terminal platform (OMTP) standard interface or a cellular telecommunications industry association of the USA (CTIA) standard interface. 
     The pressure sensor  180 A is configured to sense a pressure signal, and may convert the pressure signal into an electrical signal. In some embodiments, the pressure sensor  180 A may be disposed on the display  194 . There are a plurality of types of pressure sensors  180 A, such as a resistive pressure sensor, an inductive pressure sensor, and a capacitive pressure sensor. The capacitive pressure sensor may include at least two parallel plates made of conductive materials. When a force is applied to the pressure sensor  180 A, capacitance between electrodes changes. The mobile phone  100  determines pressure intensity based on a capacitance change. When a touch operation is performed on the display  194 , the mobile phone  100  detects strength of the touch operation by using the pressure sensor  180 A. The mobile phone  100  may also calculate a touch location based on a detection signal of the pressure sensor  180 A. In some embodiments, touch operations that are performed in a same touch position but have different touch operation strength may correspond to different operation instructions. For example, when a touch operation whose touch operation strength is less than a first pressure threshold is performed on a Messages icon, an instruction for viewing an SMS message is executed. When a touch operation whose touch operation strength is greater than or equal to the first pressure threshold is performed on the Messages icon, an instruction for creating an SMS message is executed. 
     The gyro sensor  180 B may be configured to determine a motion posture of the mobile phone  100 . In some embodiments, the gyro sensor  180 B may be used to determine angular velocities of the mobile phone  100  around three axes (namely, x, y, and z axes). The gyro sensor  180 B may be configured to implement image stabilization during photographing. For example, when a shutter is pressed, the gyro sensor  180 B detects an angle at which the mobile phone  100  jitters, obtains, through calculation based on the angle, a distance for which a lens module needs to compensate, and allows a lens to cancel the jitter of the mobile phone  100  through reverse motion, to implement image stabilization. The gyro sensor  180 B may be further used in a navigation scenario and a motion-sensing game scenario. 
     The barometric pressure sensor  180 C is configured to measure barometric pressure. In some embodiments, the mobile phone  100  calculates an altitude based on a value of the atmospheric pressure measured by the barometric pressure sensor  180 C, to assist positioning and navigation. 
     The magnetic sensor  180 D includes a Hall effect sensor. The mobile phone  100  may detect opening and closing of a flip cover by using the magnetic sensor  180 D. In some embodiments, when the mobile phone  100  is a flip phone, the mobile phone  100  may detect opening and closing of a flip cover by using the magnetic sensor  180 D. Further, a feature such as automatic unlocking upon opening of the flip cover is set based on a detected opening or closing state of the flip cover. 
     The acceleration sensor  180 E may detect a magnitude of acceleration of the mobile phone  100  in various directions (usually on three axes). When the mobile phone  100  is still, a value and a direction of gravity may be detected. The acceleration sensor  180 E may be further configured to identify a posture of the electronic device, and is used in an application such as switching between a landscape mode and a portrait mode or a pedometer. 
     The distance sensor  180 F is configured to measure a distance. The mobile phone  100  may measure a distance through infrared or laser. In some embodiments, in a photographing scenario, the mobile phone  100  may measure a distance by using the distance sensor  180 F, to implement quick focusing. 
     The optical proximity sensor  180 G may include, for example, a light-emitting diode (LED) and an optical detector such as a photodiode. The light-emitting diode may be an infrared light-emitting diode. The mobile phone  100  may emit infrared light by using the light-emitting diode. The mobile phone  100  detects reflected infrared light from a nearby object by using the photodiode. When sufficient reflected light is detected, it may be determined that there is an object near the mobile phone  100 . When insufficient reflected light is detected, the mobile phone  100  may determine that there is no object near the mobile phone  100 . The mobile phone  100  may detect, by using the optical proximity sensor  180 G, that the user puts the mobile phone  100  close to an ear for conversation, so that automatic screen-off is implemented to save power. The optical proximity sensor  180 G may also be used in a leather case mode or a pocket mode to automatically unlock or lock the screen. 
     The ambient light sensor  180 L is configured to sense ambient light luminance. The mobile phone  100  may adaptively adjust luminance of the display  194  based on the sensed luminance of the ambient light. The ambient light sensor  180 L may be further configured to automatically adjust a white balance during photographing. The ambient light sensor  180 L may further cooperate with the optical proximity sensor  180 G to detect whether the mobile phone  100  is in a pocket, thereby preventing an accidental touch. 
     The fingerprint sensor  180 H is configured to collect a fingerprint. The mobile phone  100  may use a feature of the collected fingerprint to implement fingerprint-based unlocking, application lock access, fingerprint-based photographing, fingerprint-based call answering, and the like. 
     The temperature sensor  180 J is configured to detect a temperature. In some embodiments, the mobile phone  100  executes a temperature processing policy by using the temperature detected by the temperature sensor  180 J. For example, when the temperature reported by the temperature sensor  180 J exceeds a threshold, the mobile phone  100  reduces performance of a processor near the temperature sensor  180 J, to reduce power consumption and implement heat protection. In some other embodiments, when the temperature is lower than another threshold, the mobile phone  100  heats the battery  142 , to avoid an abnormal shutdown of the mobile phone  100  caused by the low temperature. In some other embodiments, when the temperature is lower than still another threshold, the mobile phone  100  boosts an output voltage of the battery  142 , to avoid an abnormal shutdown caused by a low temperature. 
     The touch sensor  180 K is also referred to as a “touch panel”. The touch sensor  180 K may be disposed on the display  194 , and the touch sensor  180 K and the display  194  form a touchscreen, which is also referred to as a “touch screen”. The touch sensor  180 K is configured to detect a touch operation performed on or near the touch sensor. The touch sensor may transfer the detected touch operation to the application processor, to determine a type of a touch event. A visual output related to the touch operation may be provided on the display  194 . In some other embodiments, the touch sensor  180 K may alternatively be disposed on a surface of the mobile phone  100  and is at a position different from that of the display  194 . 
     The bone conduction sensor  180 M may obtain a vibration signal. In some embodiments, the bone conduction sensor  180 M may obtain a vibration signal of a vibration bone of a human vocal-cord part. The bone conduction sensor  180 M may also be in contact with a human pulse, and receive a blood pressure beating signal. In some embodiments, the bone conduction sensor  180 M may alternatively be disposed in a headset, to form a bone conduction headset. The audio module  170  may obtain a voice signal through parsing based on the vibration signal that is of the vibration bone of the vocal part and that is obtained by the bone conduction sensor  180 M, to implement a voice function. The application processor may parse heart rate information based on the blood pressure beating signal obtained by the bone conduction sensor  180 M, to implement a heart rate detection function. 
     The button  190  includes a power button, a volume button, and the like. The button  190  may be a mechanical button, or may be a touch-sensitive button. The mobile phone  100  may receive a button input, and generate a button signal input related to a user setting and function control of the mobile phone  100 . 
     The motor  191  may generate a vibration prompt. The motor  191  may be configured to provide an incoming call vibration prompt or a touch vibration feedback. For example, touch operations performed on different applications (for example, photographing and audio playing) may correspond to different vibration feedback effects. The motor  191  may also correspond to different vibration feedback effects for touch operations performed on different areas of the display  194 . Different application scenarios (for example, a time reminder, information receiving, an alarm clock, and a game) may also correspond to different vibration feedback effects. A touch vibration feedback effect may be further customized. 
     The indicator  192  may be an indicator, and may be configured to indicate a charging status and a power change, or may be configured to indicate a message, a missed call, a notification, and the like. 
     The SIM card interface  195  is configured to connect to a SIM card. The SIM card may be inserted into the SIM card interface  195  or plugged from the SIM card interface  195 , to implement contact with or separation from the mobile phone  100 . The mobile phone  100  may support one or N SIM card interfaces, where N is a positive integer greater than 1. The SIM card interface  195  can support a nano-SIM card, a micro-SIM card, a SIM card, and the like. A plurality of cards may be simultaneously inserted into a same SIM card interface  195 . The plurality of cards may be of a same type, or may be of different types. The SIM card interface  195  may be compatible with different types of SIM cards. The SIM card interface  195  may also be compatible with the external memory card. The mobile phone  100  interacts with a network by using the SIM card, to implement functions such as calling and data communication. In some embodiments, the mobile phone  100  uses an eSIM, that is, an embedded SIM card. The eSIM card may be embedded in the mobile phone  100 , and cannot be separated from the mobile phone  100 . 
     It can be understood that, in the embodiments of this application, the terminal device (for example, the mobile phone) may perform some or all steps in this embodiment of this application. These steps or operations are merely examples. In the embodiments of this application, another operation or variations of various operations may be performed. In addition, the steps may be performed in a sequence different from a sequence presented in embodiments of this application, and not all the operations in embodiments of this application are necessarily to be performed. Embodiments of this application may be implemented separately, or may be implemented in any combination. This is not limited in this application. 
     In this embodiment of this application, signal compensation may be performed on the input signal of the speaker to improve sound quality of the speaker, or displacement protection may be performed on the speaker to improve sound quality of the speaker, or displacement protection may be performed on the speaker and signal compensation may be performed on the input signal of the speaker to improve sound quality of the speaker. 
     The following describes in detail a method for improving sound quality of a speaker in an embodiment of this application. 
     As shown in  FIG. 2 , when nonlinear compensation is performed on an input signal of a speaker to improve sound quality of the speaker, the method for improving sound quality of a speaker in this embodiment of this application may include S 101  to S 103 . 
     S 101 : Perform interpolation on a second nonlinear parameter based on direct current resistance of the speaker, to obtain a third nonlinear parameter of the speaker. 
     The second nonlinear parameter is a nonlinear parameter preconfigured in the speaker, and the direct current resistance of the speaker is direct current resistance of a coil of the speaker. 
     Optionally, in this embodiment of this application, the second nonlinear parameter may be obtained by adjusting a first nonlinear parameter of the speaker, and the first nonlinear parameter is a nonlinear parameter of the speaker in an original state (which can be understood as a state in which the speaker is manufactured and is not put into use). Before delivery of the speaker, the first nonlinear parameter of the speaker is first adjusted, a parameter of the speaker is adjusted from the first nonlinear parameter to the second nonlinear parameter, and then the speaker is delivered and is put into use. 
     In this embodiment of this application, a nonlinear parameter of the speaker (for example, the first nonlinear parameter, the second nonlinear parameter, and the third nonlinear parameter) includes at least one of a force factor, mechanical stiffness, inductance, and damping of the speaker. 
     Optionally, with reference to  FIG. 2 , as shown in  FIG. 3 , S 101  may be specifically implemented by using S 1011  and S 1012 . 
     S 1011 : Determine a temperature of the coil of the speaker based on the direct current resistance of the speaker. 
     In this embodiment of this application, an output signal of the speaker, the second nonlinear parameter, and the linear parameter of the speaker are entered into a speaker model, to obtain the direct current resistance of the speaker and a current, displacement, and a velocity that exist at a current moment. The current, the displacement, and the velocity that exist at the current moment are used to perform signal compensation on a next signal in the input signal. 
     Optionally, in this embodiment of this application, the speaker model may be a model in the conventional technology. Details are not described in this embodiment of this application. 
     In this embodiment of this application, a relationship between the temperature of the coil (which may also be referred to as a temperature of a voice coil) of the speaker and the direct current resistance of the coil of the speaker is as follows: 
     
       
         
           
             T 
             = 
             
               
                 
                   1 
                   η 
                 
                 ⁢ 
                 
                   ( 
                   
                     
                       R 
                       
                         R 
                         0 
                       
                     
                     - 
                     1 
                   
                   ) 
                 
               
               + 
               
                 2 
                 ⁢ 
                 5 
               
             
           
         
       
     
     Herein, T is the temperature of the coil of the speaker, R is the direct current resistance of the coil of the speaker, η is a temperature rise coefficient, R o  is direct current resistance of the coil that corresponds to a calibration temperature, and the temperature of the voice coil is usually calibrated at 25 degrees Celsius. 
     After the direct current resistance of the speaker is obtained, the temperature of the coil of the speaker may be obtained based on the foregoing formula. 
     S 1012 : Perform interpolation on the second nonlinear parameter based on the temperature of the coil of the speaker, to obtain the third nonlinear parameter. 
     A nonlinear parameter, for example, a nonlinear parameter Kms(x) in the second nonlinear parameter is used as an example to describe a process of performing interpolation on the second nonlinear parameter to obtain the third nonlinear parameter. 
     First, characteristic curves of the nonlinear parameter Kms(x) that exist when the temperature of the coil of the speaker has different temperature values are obtained. The characteristic curve of Kms(x) is a curve reflecting a relationship between a stiffness coefficient of the speaker and displacement of the speaker. For example, 10 characteristic curves of Kms(x) from 10 degrees Celsius to 55 degrees Celsius are obtained at an interval of 5 degrees Celsius, and data of the 10 characteristic curves is stored. 
     Second, linear interpolation is performed on the characteristic curve of the nonlinear parameter Kms(x) based on the temperature of the coil of the speaker, a temperature threshold 1, and a temperature threshold 2, to obtain a target characteristic curve (the target characteristic curve may be understood as an estimation result of a characteristic curve of the third nonlinear parameter). The temperature threshold 2 is greater than the temperature threshold 1, and the third nonlinear parameter may be understood as a nonlinear parameter corresponding to a current temperature of the coil of the speaker. 
     For example, the temperature of the coil of the speaker is denoted as T, the temperature threshold 1 is denoted as T min , and the temperature threshold 2 is denoted as T max . 
     If T&lt;T min , a characteristic curve corresponding to is used as the target characteristic curve. 
     If T&gt;T max , a characteristic curve corresponding to T max  is used as the target characteristic curve. 
     If T min ≤T≤T max , linear interpolation is performed on a characteristic curve corresponding to and a characteristic curve corresponding to T max  based on the temperature of the coil of the speaker, to generate the target characteristic curve. 
     Finally, polynomial fitting is performed on the target characteristic curve, to obtain each coefficient of a polynomial corresponding to the target characteristic curve. Each coefficient is in a one-to-one correspondence with a nonlinear parameter. Therefore, the third nonlinear parameter may be determined based on each coefficient of the polynomial. 
     For example, for the nonlinear parameter Kms(x), it is assumed that a binomial obtained through fitting is: 
     
       
         
           
             
               f 
               ⁡ 
               
                 ( 
                 x 
                 ) 
               
             
             = 
             
               
                 a 
                 0 
               
               + 
               
                 
                   a 
                   1 
                 
                 ⁢ 
                 x 
               
               + 
               
                 
                   a 
                   2 
                 
                 ⁢ 
                 
                   x 
                   2 
                 
               
               + 
               
                 
                   a 
                   3 
                 
                 ⁢ 
                 
                   x 
                   3 
                 
               
               + 
               
                 
                   a 
                   4 
                 
                 ⁢ 
                 
                   x 
                   4 
                 
               
             
           
         
       
     
     Herein, a coefficient a 1  corresponds to a first order coefficient of the nonlinear parameter Kms(x), a coefficient a 2  corresponds to a second order coefficient of the nonlinear parameter Kms(x), a coefficient a 3  corresponds to a third order coefficient of the nonlinear parameter Kms(x), and a coefficient a 4  corresponds to a fourth order coefficient of the nonlinear parameter Kms(x). 
     A parameter of another type in the third nonlinear parameter, for example, Rm(v), may also be obtained by using a method similar to the linear interpolation method. Details are not described in this embodiment of this application. 
     Optionally, the characteristic curve of the nonlinear parameter may be data in a form of a table, or may be data or a file in another form. This is not limited in this embodiment of this application. 
     The nonlinear parameter of the speaker may change in real time. For example, the nonlinear parameter changes with the temperature of the voice coil of the speaker. In this embodiment of this application, interpolation is performed on the second nonlinear parameter of the speaker based on current direct current resistance of the speaker, to obtain nonlinear resistance (that is, third nonlinear resistance) of the speaker in real time. The third nonlinear parameter has high accuracy. 
     It should be understood that the nonlinear parameter of the speaker may also change with the displacement of the speaker. In this embodiment of this application, a similar linear interpolation method may be used to determine the displacement of the speaker based on the direct current resistance of the speaker, and then interpolation is performed on the second nonlinear parameter of the speaker based on the displacement of the speaker, to obtain the third nonlinear parameter of the speaker. In this case, different from S 1012 , a characteristic curve of the second nonlinear parameter is a characteristic curve corresponding to different displacement. 
     S 102 : Perform signal compensation on a first input signal of the speaker based on the third nonlinear parameter, to obtain a compensated first input signal. 
     Specifically, the performing signal compensation on a first input signal of the speaker based on the third nonlinear parameter includes: entering, into a signal compensation model, the first input signal, the third nonlinear parameter, the linear parameter of the speaker, and a current, displacement, and a velocity that are obtained at a previous moment, to obtain the compensated first input signal. 
     In this embodiment of this application, the determined third nonlinear parameter of the speaker has high accuracy. Therefore, signal compensation is performed on the first input signal based on the third nonlinear parameter, so that there is a good signal compensation effect, and impact of the nonlinear parameter on the input signal can be effectively reduced. 
     S 103 : Perform filtering on the compensated first input signal, to obtain the output signal of the speaker. 
     In this embodiment of this application, the compensated first input signal is denoted as a first signal, and a filtering process of the compensated first input signal includes A1 to A5. 
     A1: Perform filtering on the first signal by using a wavetrap, to obtain a second signal. 
     Optionally, a function expression of the wavetrap may be as follows: 
     
       
         
           
             
               
                 H 
                 z 
               
               ⁡ 
               
                 ( 
                 Z 
                 ) 
               
             
             = 
             
               
                 ( 
                 
                   
                     a 
                     1 
                   
                   + 
                   
                     
                       a 
                       2 
                     
                     ⁢ 
                     
                       z 
                       
                         - 
                         1 
                       
                     
                   
                   + 
                   
                     
                       a 
                       3 
                     
                     ⁢ 
                     
                       z 
                       
                         - 
                         2 
                       
                     
                   
                 
                 ) 
               
               / 
               
                 ( 
                 
                   
                     b 
                     1 
                   
                   + 
                   
                     
                       b 
                       2 
                     
                     ⁢ 
                     
                       z 
                       
                         - 
                         1 
                       
                     
                   
                   + 
                   
                     
                       b 
                       3 
                     
                     ⁢ 
                     
                       z 
                       
                         - 
                         2 
                       
                     
                   
                 
                 ) 
               
             
           
         
       
     
     Herein, a 1 , a 2 , a 3 , b 1 , b 2 , and b 3  are filtering coefficients of the wavetrap. 
     
       
         
           
             
               
                 a 
                 1 
               
               = 
               
                 0.5 
                 * 
                 
                   ( 
                   
                     1 
                     + 
                     μ 
                   
                   ) 
                 
               
             
             , 
             
               
                 a 
                 2 
               
               = 
               
                 
                   - 
                   β 
                 
                 * 
                 
                   ( 
                   
                     1 
                     + 
                     μ 
                   
                   ) 
                 
               
             
             , 
             
               
                 a 
                 3 
               
               = 
               
                 0.5 
                 * 
                 
                   ( 
                   
                     1 
                     + 
                     μ 
                   
                   ) 
                 
               
             
           
         
       
       
         
           
             
               
                 b 
                 1 
               
               = 
               1 
             
             , 
             
               
                 b 
                 2 
               
               = 
               
                 
                   - 
                   β 
                 
                 * 
                 
                   ( 
                   
                     1 
                     + 
                     μ 
                   
                   ) 
                 
               
             
             , 
             
               
                 and 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 
                   b 
                   3 
                 
               
               = 
               μ 
             
           
         
       
       
         
           
             Herein 
             , 
             
               
 
             
             ⁢ 
             
               β 
               = 
               
                 cos 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 
                   w 
                   0 
                 
               
             
             , 
             
               μ 
               = 
               
                 
                   ( 
                   
                     1 
                     + 
                     
                       
                         tan 
                         ⁡ 
                         
                           ( 
                           
                             
                               w 
                               0 
                             
                             2 
                           
                           ) 
                         
                       
                       2 
                     
                     - 
                     
                       tan 
                       ⁡ 
                       
                         ( 
                         
                           
                             B 
                             w 
                           
                           2 
                         
                         ) 
                       
                     
                   
                   ) 
                 
                 / 
                 
                   ( 
                   
                     1 
                     + 
                     
                       
                         tan 
                         ⁡ 
                         
                           ( 
                           
                             
                               w 
                               0 
                             
                             2 
                           
                           ) 
                         
                       
                       2 
                     
                     + 
                     
                       tan 
                       ⁡ 
                       
                         ( 
                         
                           
                             B 
                             w 
                           
                           2 
                         
                         ) 
                       
                     
                   
                   ) 
                 
               
             
             , 
             
               
 
             
             ⁢ 
             and 
           
         
       
       
         
           
             
               w 
               0 
             
             = 
             
               2 
               * 
               π 
               * 
               
                 
                   
                     f 
                     0 
                   
                   
                     f 
                     s 
                   
                 
                 . 
               
             
           
         
       
     
     Herein, f 0  is a resonant frequency of the speaker, f s  is a sampling frequency, and B w  is a digital bandwidth coefficient. 
     A2: Calculate a difference between the first signal and the second signal, to obtain a third signal. 
     A3: Multiply the third signal by a filtering gain, to obtain a fourth signal. 
     The filtering gain is used to perform filtering on the compensated first input signal. 
     A4: Calculate a difference between the first signal and the fourth signal, to obtain a fifth signal, and use the fifth signal as the output signal of the speaker. 
     In this embodiment of this application, filtering is performed on the first input signal, to adjust a velocity that is of a diaphragm of the speaker and that exists near the resonant frequency, so that distortion of the output signal is further reduced, and the sound quality of the speaker can be effectively improved. 
     In this embodiment of this application, the filtering gain may be generated based on the input signal. Specifically, generating the filtering gain may include S 1  and S 2 . 
     S 1 : Determine a maximum value of an absolute voltage value in a current frame. 
     It should be understood that the first input signal includes a plurality of signal frames, each signal frame includes a plurality of input voltages, and a voltage with a largest absolute value in the plurality of input voltages is the maximum value of the absolute voltage value in the current frame. 
     S 2 : Determine the filtering gain based on the maximum value of the absolute voltage value. 
     In this embodiment of this application, the maximum value of the absolute voltage value is denoted as U max , the filtering gain is denoted as α, and the determining the filtering gain based on the maximum value of the absolute voltage value includes: 
     
       
         
           
             
               
                 When 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 
                   U 
                   max 
                 
               
               &lt; 
               
                 U 
                 lowlimit 
               
             
             , 
             
               
 
             
             ⁢ 
             
               α 
               = 
               
                 
                   α 
                   buffer 
                 
                 * 
                 
                   α 
                   smooth 
                 
               
             
             , 
           
         
       
     
     where 
     U lowlimit  is a voltage control lower limit, α buffer  is a filtering gain corresponding to a previous frame, α smooth  is a smoothing coefficient of the filtering gain, and * represents convolution; 
     
       
         
           
             
               
                 when 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 
                   U 
                   uplimit 
                 
               
               ≤ 
               
                 U 
                 max 
               
               ≤ 
               
                 U 
                 lowlimit 
               
             
             , 
             
               
 
             
             ⁢ 
             
               α 
               = 
               
                 
                   
                     α 
                     buffer 
                   
                   * 
                   
                     α 
                     smooth 
                   
                 
                 + 
                 
                   
                     
                       
                         U 
                         max 
                       
                       - 
                       
                         U 
                         lowlimit 
                       
                     
                     
                       
                         U 
                         uplimit 
                       
                       - 
                       
                         U 
                         lowlimit 
                       
                     
                   
                   * 
                   
                     ( 
                     
                       
                         α 
                         uplimit 
                       
                       - 
                       
                         α 
                         lowlimit 
                       
                     
                     ) 
                   
                   * 
                   
                     ( 
                     
                       1 
                       - 
                       
                         α 
                         smooth 
                       
                     
                     ) 
                   
                 
               
             
             , 
           
         
       
     
     where 
     U uplimit  is a voltage control upper limit, α uplimit  is a control upper limit of the filtering gain, and α lowlimit  is a control lower limit of the filtering gain; or 
     
       
         
           
             
               
                 when 
                 ⁢ 
                 
                     
                 
                 ⁢ 
                 
                   U 
                   max 
                 
               
               &gt; 
               
                 U 
                 uplimit 
               
             
             , 
             
               
 
             
             ⁢ 
             
               α 
               = 
               
                 
                   
                     α 
                     buffer 
                   
                   * 
                   
                     α 
                     smooth 
                   
                 
                 + 
                 
                   
                     α 
                     uplimit 
                   
                   * 
                   
                     ( 
                     
                       1 
                       - 
                       
                         α 
                         smooth 
                       
                     
                     ) 
                   
                 
               
             
           
         
       
     
     Therefore, nonlinear compensation performed on the first input signal of the speaker is completed, and the output signal of the speaker is obtained for outputting.  FIG. 4  is a flow block diagram of a method for improving sound quality of a speaker through signal compensation. 
     Optionally, in an embodiment of this application, a first nonlinear parameter of a speaker may be adjusted based on an acoustic signal of the speaker or a displacement signal of the speaker, to obtain a second nonlinear parameter of the speaker. Specifically, as shown in  FIG. 5 , the first nonlinear parameter of the speaker is adjusted by cyclically performing S 201  to S 204 . 
     S 201 : Obtain the acoustic signal of the speaker or the displacement signal of the speaker. 
     In this embodiment of this application, for an input signal, an acoustic signal output by the speaker or a displacement signal of the speaker may be collected. Optionally, the input signal may be a sweep signal or a chirp signal. This is not limited in this embodiment of this application. 
     For an input signal U in (n) of the speaker, the signal U in (n), a nonlinear parameter Pn(t n-1 ) obtained at a previous moment, a linear parameter Pi(t n-1 ), and a current i(n−1), displacement x(n−1), and a velocity v(n−1) that are obtained at the previous moment are input into a nonlinear compensation model, to obtain an output signal U out (n), so as to collect an acoustic signal or a displacement signal corresponding to the output signal. 
     The current i(n−1), the displacement x(n−1), and the velocity v(n−1) that are obtained at the previous moment are obtained by feeding back, to a speaker model, an output signal U out (n−1) existing at the previous moment. Specifically, the output signal U out (n−1) existing at the previous moment, the nonlinear parameter Pn(t n-1 ) obtained at the previous moment, and the linear parameter Pi(t n-1 ) are input into the speaker model, to obtain the current i(n−1), the displacement x(n−1), and the velocity v(n−1). 
     Optionally, in this embodiment of this application, the speaker model may be an existing speaker model, and the speaker model is not described in detail herein. 
     S 202 : Determine a target to-be-adjusted parameter from the first nonlinear parameter of the speaker based on the acoustic signal of the speaker or the displacement signal of the speaker. 
     Specifically, S 202  may be implemented by using S 2021  to S 2023 . 
     S 2021 : Perform Fourier transform on the acoustic signal of the speaker or the displacement signal of the speaker, to obtain harmonic distortion. 
     A Fourier transform result of the acoustic signal or the displacement signal may include N order harmonic distortion. In this embodiment of this application, a nonlinear parameter of the speaker is in a one-to-one correspondence with each order of harmonic distortion in the Fourier transform result of the acoustic signal or the displacement signal. Table 1 shows an example of a correspondence between a nonlinear parameter and harmonic distortion. 
     
       
         
           
               
               
             
               
                 TABLE 1 
               
               
                   
               
               
                 Harmonic distortion 
                 Nonlinear parameter 
               
               
                   
               
             
            
               
                 Second order harmonic 
                 First order coefficient of BL(x), and first 
               
               
                 distortion 
                 order coefficient of Kms(x) 
               
               
                 Third order harmonic 
                 Second order coefficient of BL(x), and second 
               
               
                 distortion 
                 order coefficient of Kms(x) 
               
               
                 Fourth order harmonic 
                 Third order coefficient of BL(x), and third 
               
               
                 distortion 
                 order coefficient of Kms(x) 
               
               
                 Fifth order harmonic 
                 Fourth order coefficient of BL(x), and fourth 
               
               
                 distortion 
                 order coefficient of Kms(x) 
               
               
                   
               
            
           
         
       
     
     S 2022 : Determine a candidate to-be-adjusted parameter from the first nonlinear parameter of the speaker based on the harmonic distortion. 
     In this embodiment of this application, harmonic distortion in a Fourier transform result corresponding to an acoustic signal or a displacement signal collected after signal compensation is performed on an input signal is referred to as first harmonic distortion, and harmonic distortion in a Fourier transform result corresponding to an acoustic signal or a displacement signal collected before compensation is performed on an input signal is referred to as second harmonic distortion. Therefore, the determining a candidate to-be-adjusted parameter from the first nonlinear parameter of the speaker may include: determining the candidate to-be-adjusted parameter based on the first harmonic distortion and the second harmonic distortion. The determining the candidate to-be-adjusted parameter based on the first harmonic distortion and the second harmonic distortion specifically includes the following steps. 
     Step 1: Determine a ratio of each order of harmonic distortion in the second harmonic distortion to each corresponding order harmonic distortion in the first harmonic distortion. 
     For example, it is assumed that the first harmonic distortion includes second order harmonic distortion to fifth order harmonic distortion, and similarly, the second harmonic distortion also includes second order harmonic distortion to fifth order harmonic distortion. A ratio of the second order harmonic distortion in the second harmonic distortion to the second order harmonic distortion in the first harmonic distortion is calculated, and is recorded as a ratio 1, and a ratio 2, a ratio 3, and a ratio 4 are obtained by analogy. Table 2 shows an example of a correspondence among a nonlinear parameter, each order of harmonic distortion, and a ratio of each order of harmonic distortion. 
     
       
         
           
               
               
               
             
               
                 TABLE 2 
               
               
                   
               
               
                   
                 Each order of 
                 A ratio of each 
               
               
                   
                 harmonic 
                 order of harmonic 
               
               
                 Nonlinear parameter 
                 distortion 
                 distortion 
               
               
                   
               
             
            
               
                 First order coefficient 
                 Second order 
                 Ratio 1 
               
               
                 of BL(x), and first order 
                 harmonic 
               
               
                 coefficient of Kms(x) 
                 distortion 
               
               
                 Second order coefficient 
                 Third order 
                 Ratio 2 
               
               
                 of BL(x), and second order 
                 harmonic 
               
               
                 coefficient of Kms(x) 
                 distortion 
               
               
                 Third order coefficient 
                 Fourth order 
                 Ratio 3 
               
               
                 of BL(x), and third order 
                 harmonic 
               
               
                 coefficient of Kms(x) 
                 distortion 
               
               
                 Fourth order coefficient 
                 Fifth order 
                 Ratio 4 
               
               
                 of BL(x), and fourth order 
                 harmonic 
               
               
                 coefficient of Kms(x) 
                 distortion 
               
               
                   
               
            
           
         
       
     
     Step 2: Determine, as the candidate to-be-adjusted parameter, a nonlinear parameter corresponding to each ratio that is of harmonic distortion and that is greater than a preset threshold. 
     Optionally, in this embodiment of this application, the preset threshold may be determined based on an actual use requirement, and each order of harmonic distortion may correspond to a same or different preset threshold. This is not limited in this embodiment of this application. 
     For example, it is assumed that a threshold corresponding to the second order harmonic is denoted as a preset threshold 1, a threshold corresponding to the third order harmonic is denoted as a preset threshold 2, a threshold corresponding to the fourth order harmonic is denoted as a preset threshold 3, and a threshold corresponding to the fifth order harmonic is denoted as a preset threshold 4. When a ratio of the second order harmonic distortion is greater than the preset threshold 1, and a ratio of the fourth order harmonic distortion is greater than the preset threshold 3, the first order coefficient of BL(x), the first order coefficient of Kms(x), the third order coefficient of BL(x), and the third order coefficient of Kms(x) may be determined as candidate to-be-adjusted parameters. 
     S 2023 : Determine the target to-be-adjusted parameter from the candidate to-be-adjusted parameter. 
     In this embodiment of this application, after the candidate to-be-adjusted parameter is determined, a convergence error of each nonlinear parameter in the candidate to-be-adjusted parameter is obtained, the convergence error of each nonlinear parameter in the candidate to-be-adjusted parameter is compared with a preset error threshold corresponding to each nonlinear parameter, and each nonlinear parameter whose convergence error is greater than the error threshold in the candidate to-be-adjusted parameter is determined as the target to-be-adjusted parameter. 
     Optionally, in this embodiment of this application, the preset error threshold may be determined based on an actual use requirement, and each nonlinear parameter may correspond to a same or different preset error threshold. This is not limited in this embodiment of this application. 
     For example, with reference to the example in step 2, determined candidate to-be-adjusted parameters are the first order coefficient of BL(x), the first order coefficient of Kms(x), the third order coefficient of BL(x), and the third order coefficient of Kms(x). A preset error threshold corresponding to the first order coefficient of BL(x) is denoted as a preset error threshold 1, a preset error threshold corresponding to the first order coefficient of Kms(x) is denoted as a preset error threshold 2, a preset error threshold corresponding to the third order coefficient of BL(x) is denoted as a preset error threshold 3, and a preset error threshold corresponding to the third order coefficient of Kms(x) is denoted as a preset error threshold 4. A convergence error of the first order coefficient of BL(x) is greater than the preset error threshold 1, and a convergence error of the third order coefficient of Kms(x) is greater than the preset error threshold 4. Therefore, the first order coefficient of BL(x) and the third order coefficient of Kms(x) are determined as target to-be-adjusted parameters, and another nonlinear parameter does not need to be adjusted. 
     S 203 : Calibrate the target to-be-adjusted parameter in the first nonlinear parameter of the speaker based on a target direction and a target step, to obtain an adjusted first nonlinear parameter. 
     In this embodiment of this application, the target direction may include a forward direction and a reverse direction. The forward direction may be defined as a direction in which the nonlinear parameter is increased, and the reverse direction may be defined as a direction in which the nonlinear parameter is decreased. The target direction may be specifically defined based on an actual requirement. This is not limited in this embodiment of this application. 
     The target step represents an adjustment (increase or decrease) amplitude of the nonlinear parameter, and the target step may include a target step corresponding to the forward direction and a target step corresponding to the reverse direction. 
     Optionally, the target step corresponding to the forward direction may be the same as or different from the target step corresponding to the reverse direction. This is not specifically limited in this embodiment of this application. For example, the nonlinear parameter is adjusted in the forward direction, a corresponding target step is set to 5%, the nonlinear parameter is adjusted in the reverse direction, and a corresponding step is set to 10%. 
     Optionally, in this embodiment of this application, when the first nonlinear parameter includes a plurality of nonlinear parameters, different nonlinear parameters may correspond to a same target step or different target steps. This is not specifically limited in this embodiment of this application. For example, it is assumed that nonlinear parameters include BL(x), Kms(x), and Le(x). Table 3 shows an example of a target step corresponding to each nonlinear parameter. 
     
       
         
           
               
               
               
               
             
               
                   
                 TABLE 3 
               
               
                   
                   
               
               
                   
                 First nonlinear parameter 
                 Target direction 
                 Target step 
               
               
                   
                   
               
             
            
               
                   
                 BL(x) 
                 Forward direction 
                 2% 
               
               
                   
                   
                 Reverse direction 
                 2% 
               
               
                   
                 Kms(x) 
                 Forward direction 
                 3% 
               
               
                   
                   
                 Reverse direction 
                 4% 
               
               
                   
                 Le(x) 
                 Forward direction 
                 3% 
               
               
                   
                   
                 Reverse direction 
                 4% 
               
               
                   
                   
               
            
           
         
       
     
     It should be understood that, after the target to-be-adjusted parameter in the first nonlinear parameter is adjusted, the adjusted first nonlinear parameter is obtained. Further, a convergence error of the first nonlinear parameter is updated based on the adjusted first nonlinear parameter, so that the updated convergence error is used for a corresponding step in a subsequent cycle. 
     S 204 : Perform signal compensation on an input signal of the speaker based on the adjusted first nonlinear parameter, to obtain an output signal of the speaker. 
     It can be understood that, after signal compensation is performed on the input signal of the speaker to obtain the output signal of the speaker, S 201  is performed. To be specific, for the output signal, an acoustic signal or a displacement signal of the speaker is obtained.  FIG. 6  is a flow block diagram of a method for adjusting a parameter of a speaker. 
     The first nonlinear parameter is adjusted, to obtain the adjusted first nonlinear parameter. Therefore, a value of the first nonlinear parameter of the speaker is updated to a value of the adjusted first nonlinear parameter (in other words, the value of the adjusted nonlinear parameter is used to replace the value of the first nonlinear parameter existing before adjustment). Then, the target to-be-adjusted parameter is determined from the first nonlinear parameter based on the acoustic signal of the speaker or the displacement signal of the speaker (the first nonlinear parameter herein is the updated first nonlinear parameter). For a specific process, refer to related descriptions in the foregoing embodiment. Details are not described herein again. 
     In conclusion, S 201  to S 204  are cyclically performed, and an adjusted first nonlinear parameter obtained in the last cycle is used as the second nonlinear parameter. 
     Optionally, in an embodiment of this application, a first nonlinear parameter is obtained through measurement. Specifically, S 301  to S 305  may be cyclically performed, to obtain a first nonlinear parameter of a speaker. 
     S 301 : Input an input current, a nonlinear parameter, and a linear parameter into a lumped-loudspeaker model, to obtain a predicted voltage of the speaker. 
     In this embodiment of this application, the input current is a current generated by the speaker under a driving action of an input signal, and the input signal is an input voltage. For ease of description, the input voltage is referred to as an actual voltage in the following embodiment. 
     It should be understood that, in a first cycle, the nonlinear parameter and the linear parameter in S 301  are initialized parameters, and optionally, may be a randomly initialized nonlinear parameter and linear parameter. The nonlinear parameter and the linear parameter in S 301  in a subsequent cycle are a nonlinear parameter and a linear parameter that are obtained in a previous cycle. 
     S 302 : Obtain an error signal based on the actual voltage and the predicted voltage. 
     The error signal is a difference between the actual voltage and the predicted voltage. 
     S 303 : Input the error signal and the input current into a linear parameter identification model, to obtain a linear parameter of the speaker. 
     S 304 : Decorrelate the error signal, to obtain a decorrelated error signal. 
     In this embodiment of this application, the decorrelating the error signal is to remove a linear signal from the error signal. Optionally, the error signal may be decorrelated in a decorrelation method (by using a decorrelation model) in the conventional technology. This is not limited in this embodiment of this application. 
     S 305 : Input the decorrelated error signal and the input current into a nonlinear parameter identification model, to obtain a nonlinear parameter of the speaker. 
     Optionally, in this embodiment of this application, the lumped-speaker model, the linear parameter identification model, and the nonlinear parameter identification model may all be models provided in the conventional technology. This is not limited in this embodiment of this application. 
     It should be understood that, a linear parameter corresponding to a current cycle is obtained in S 304 , a nonlinear parameter corresponding to the current cycle is obtained in S 305 , and then a next input current, the linear parameter obtained in the current cycle, and the nonlinear parameter obtained in the current cycle are input into the lumped-speaker model, to obtain the predicted voltage. In other words, S 301  to S 305  continue to be performed until a preset quantity of cycles is reached, or the nonlinear parameter and the linear parameter converge until a preset tolerant error is obtained, to obtain a final linear parameter and a nonlinear parameter, and a finally obtained nonlinear parameter is used as the first nonlinear parameter. 
     It should be noted that, S 301  to S 305  are cyclically performed, to obtain a convergence error of a nonlinear parameter. It should be understood that the convergence error of the nonlinear parameter includes a convergence error corresponding to each parameter in the nonlinear parameter. 
     In the method for improving sound quality of a speaker in this embodiment of this application, interpolation may be performed on the second nonlinear parameter of the speaker based on direct current resistance of the speaker, to obtain a third nonlinear parameter of the speaker; signal compensation is performed on a first input signal of the speaker based on the third nonlinear parameter, to obtain a compensated first input signal; and filtering is performed on the compensated first input signal, to obtain an output signal of the speaker. The third nonlinear parameter is obtained by performing interpolation on the second nonlinear parameter based on the direct current resistance of the speaker. The nonlinear parameter (that is, the third nonlinear parameter) is a nonlinear parameter corresponding to a current working state of the speaker, in other words, is a real-time nonlinear parameter. The nonlinear parameter has high accuracy. Therefore, signal compensation can be more effectively performed on the first input signal of the speaker based on the third nonlinear parameter, and filtering is performed on the compensated first input signal, to further reduce signal distortion. In this way, the sound quality of the speaker can be effectively improved. 
     As shown in  FIG. 7 , displacement protection is performed on a speaker, to improve sound quality of the speaker. A method for improving sound quality of a speaker in this embodiment of this application may include S 401  to S 406 . 
     S 401 : Perform first displacement conversion on an input signal of the speaker, to obtain a maximum value of first predicted displacement and an effective value of the first predicted displacement. 
     For example, an input signal U in (t n ) that is of the speaker and that exists at a current moment (a moment t n ) is used as an example. It should be noted that in the following embodiment, the n th  frame of signal in the input signal that is of the speaker and that corresponds to the moment t n  is the n th  frame of signal corresponding to U in (t n ). 
     The effective value of the first predicted displacement is calculated based on the following formula: 
     
       
         
           
             
               
                 X 
                 
                   mean 
                   ⁢ 
                   _ 
                   ⁢ 
                   ts 
                 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               mean 
               ⁡ 
               
                 [ 
                 
                   
                     
                       H 
                       
                         u 
                         ⁢ 
                         x 
                       
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                   * 
                   
                     
                       U 
                       in 
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         n 
                       
                       ) 
                     
                   
                 
                 ] 
               
             
           
         
       
     
     Herein, X mean_ts (t n ) represents the effective value of the first predicted displacement, H ux (t n-1 ) is a mathematical model for performing first displacement conversion, and may also be referred to as a displacement transfer function of the speaker, mean represents calculating an effective value, and * represents convolution. 
     The maximum value of the first predicted displacement is calculated based on the following formula: 
     
       
         
           
             
               
                 X 
                 
                   max 
                   ⁢ 
                   _ 
                   ⁢ 
                   ts 
                 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               max 
               ⁡ 
               
                 [ 
                 
                   
                     
                       H 
                       
                         u 
                         ⁢ 
                         x 
                       
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                   * 
                   
                     
                       U 
                       in 
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         n 
                       
                       ) 
                     
                   
                 
                 ] 
               
             
           
         
       
     
     Herein, X max_ts (t n ) represents the maximum value of the first predicted displacement, max represents calculating a maximum value, and * represents convolution. 
     In this embodiment of this application, a feedback signal (a feedback voltage) that is of the speaker and that exists at a previous moment may be input into a linear parameter identification model, to obtain a linear parameter of the speaker, and further, the displacement transfer function is updated based on the linear parameter. 
     Optionally, the displacement transfer function is as follows: 
     
       
         
           
             
               
                 
                   H 
                   
                     u 
                     ⁢ 
                     x 
                   
                 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     
                       n 
                       - 
                       1 
                     
                   
                   ) 
                 
               
               = 
               
                 
                   L 
                   
                     - 
                     1 
                   
                 
                 ⁢ 
                 
                   { 
                   
                     
                       B 
                       ⁢ 
                       
                         L 
                         ⁡ 
                         
                           ( 
                           
                             t 
                             
                               n 
                               - 
                               1 
                             
                           
                           ) 
                         
                       
                     
                     
                       
                         
                           
                             
                               ( 
                               
                                 
                                   
                                     R 
                                     e 
                                   
                                   ⁡ 
                                   
                                     ( 
                                     
                                       t 
                                       
                                         n 
                                         - 
                                         1 
                                       
                                     
                                     ) 
                                   
                                 
                                 + 
                                 
                                   
                                     
                                       L 
                                       e 
                                     
                                     ⁡ 
                                     
                                       ( 
                                       
                                         t 
                                         
                                           n 
                                           - 
                                           1 
                                         
                                       
                                       ) 
                                     
                                   
                                   ⁢ 
                                   S 
                                 
                               
                               ) 
                             
                             ⁢ 
                             
                               ( 
                               
                                 
                                   
                                     
                                       M 
                                       
                                         m 
                                         ⁢ 
                                         s 
                                       
                                     
                                     ⁡ 
                                     
                                       ( 
                                       
                                         t 
                                         
                                           n 
                                           - 
                                           1 
                                         
                                       
                                       ) 
                                     
                                   
                                   ⁢ 
                                   
                                     S 
                                     2 
                                   
                                 
                                 + 
                               
                             
                           
                         
                       
                       
                         
                           
                             
                               
                                 
                                   
                                     
                                       R 
                                       
                                         m 
                                         ⁢ 
                                         s 
                                       
                                     
                                     ⁡ 
                                     
                                       ( 
                                       
                                         t 
                                         
                                           n 
                                           - 
                                           1 
                                         
                                       
                                       ) 
                                     
                                   
                                   ⁢ 
                                   S 
                                 
                                 + 
                                 
                                   
                                     K 
                                     
                                       m 
                                       ⁢ 
                                       s 
                                     
                                   
                                   ⁡ 
                                   
                                     ( 
                                     
                                       t 
                                       
                                         n 
                                         - 
                                         1 
                                       
                                     
                                     ) 
                                   
                                 
                               
                               ) 
                             
                             + 
                             
                               B 
                               ⁢ 
                               
                                 
                                   L 
                                   2 
                                 
                                 ⁡ 
                                 
                                   ( 
                                   
                                     t 
                                     
                                       n 
                                       - 
                                       1 
                                     
                                   
                                   ) 
                                 
                               
                               ⁢ 
                               S 
                             
                           
                         
                       
                     
                   
                   } 
                 
               
             
             ⁢ 
             
               
 
             
             ⁢ 
             
               
                 P 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     
                       n 
                       - 
                       1 
                     
                   
                   ) 
                 
               
               = 
               
                 [ 
                 
                   
                     B 
                     ⁢ 
                     
                       L 
                       ⁡ 
                       
                         ( 
                         
                           t 
                           
                             n 
                             - 
                             1 
                           
                         
                         ) 
                       
                     
                   
                   , 
                   
                     
                       R 
                       e 
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                   , 
                   
                     
                       L 
                       e 
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                   , 
                   
                     
                       M 
                       ms 
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                   , 
                   
                     
                       R 
                       
                         m 
                         ⁢ 
                         s 
                       
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                   ⁢ 
                   
                       
                   
                   , 
                   
                     
                       K 
                       
                         m 
                         ⁢ 
                         s 
                       
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                 
                 ] 
               
             
           
         
       
     
     Herein, L −1  represents inverse Laplace transform, and P(t n-1 ) represents the linear parameters of the speaker. 
     In this embodiment of this application, the displacement transfer function of the speaker may alternatively be a transfer function of another type, and is not limited to a function type shown in the foregoing formula. 
     S 402 : Perform second displacement conversion on the feedback signal of the speaker, to obtain an effective value of second predicted displacement. 
     The feedback signal of the speaker includes a feedback current and the feedback voltage. 
     For example, the input signal U in (n) that is of the speaker and that exists at the current moment (t n ) is used as an example, and the effective value of the second predicted displacement is calculated based on the following formula: 
     
       
         
           
             
               
                 X 
                 
                   mean 
                   ⁢ 
                   _ 
                   ⁢ 
                   emf 
                 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               mean 
               ⁡ 
               
                 [ 
                 
                   
                     ∫ 
                     0 
                     T 
                   
                   ⁢ 
                   
                     
                       
                         
                           
                             U 
                             m 
                           
                           ⁡ 
                           
                             ( 
                             
                               t 
                               
                                 n 
                                 - 
                                 1 
                               
                             
                             ) 
                           
                         
                         - 
                         
                           
                             
                               R 
                               e 
                             
                             ⁡ 
                             
                               ( 
                               
                                 t 
                                 
                                   n 
                                   - 
                                   1 
                                 
                               
                               ) 
                             
                           
                           ⁢ 
                           
                             
                               I 
                               m 
                             
                             ⁡ 
                             
                               ( 
                               
                                 t 
                                 
                                   n 
                                   - 
                                   1 
                                 
                               
                               ) 
                             
                           
                         
                         - 
                         
                           
                             
                               L 
                               e 
                             
                             ⁡ 
                             
                               ( 
                               
                                 t 
                                 
                                   n 
                                   - 
                                   1 
                                 
                               
                               ) 
                             
                           
                           ⁢ 
                           
                             
                               
                                 dI 
                                 m 
                               
                               ⁡ 
                               
                                 ( 
                                 
                                   t 
                                   
                                     n 
                                     - 
                                     1 
                                   
                                 
                                 ) 
                               
                             
                             
                               d 
                               ⁢ 
                               t 
                             
                           
                         
                       
                       
                         B 
                         ⁢ 
                         
                           L 
                           ⁡ 
                           
                             ( 
                             
                               t 
                               
                                 n 
                                 - 
                                 1 
                               
                             
                             ) 
                           
                         
                       
                     
                     ⁢ 
                     dt 
                   
                 
                 ] 
               
             
           
         
       
     
     Herein, X mean_enf (t n ) represents the effective value of the second predicted displacement, U m (t n-1 ) represents the feedback voltage (that is, a voltage output at the previous moment), and I m (t n-1 ) represents the feedback current (that is, a current output at the previous moment (t n-1 )). 
     A mathematical model for the second displacement conversion (which may also be referred to as an induced electromotive force model of the speaker) may be as follows: 
     
       
         
           
             
               ∫ 
               0 
               T 
             
             ⁢ 
             
               
                 
                   
                     
                       U 
                       m 
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                   - 
                   
                     
                       
                         R 
                         e 
                       
                       ⁡ 
                       
                         ( 
                         
                           t 
                           
                             n 
                             - 
                             1 
                           
                         
                         ) 
                       
                     
                     ⁢ 
                     
                       
                         I 
                         m 
                       
                       ⁡ 
                       
                         ( 
                         
                           t 
                           
                             n 
                             - 
                             1 
                           
                         
                         ) 
                       
                     
                   
                   - 
                   
                     
                       
                         L 
                         e 
                       
                       ⁡ 
                       
                         ( 
                         
                           t 
                           
                             n 
                             - 
                             1 
                           
                         
                         ) 
                       
                     
                     ⁢ 
                     
                       
                         
                           dI 
                           m 
                         
                         ⁡ 
                         
                           ( 
                           
                             t 
                             
                               n 
                               - 
                               1 
                             
                           
                           ) 
                         
                       
                       
                         d 
                         ⁢ 
                         t 
                       
                     
                   
                 
                 
                   B 
                   ⁢ 
                   
                     L 
                     ⁡ 
                     
                       ( 
                       
                         t 
                         
                           n 
                           - 
                           1 
                         
                       
                       ) 
                     
                   
                 
               
               ⁢ 
               dt 
             
           
         
       
     
     Herein, R e (t n-1 ) is direct current resistance. 
     In this embodiment of this application, similarly, the feedback signal (the feedback voltage U m (t n-1 )) that is of the speaker and that exists at the previous moment may be input into the linear parameter identification model, to obtain the linear parameter P(t n-1 ) of the speaker, and further, the induced electromotive force model of the speaker is updated based on the linear parameter. 
     S 403 : Determine a displacement correction gain based on the effective value of the first predicted displacement and the effective value of the second predicted displacement. 
     S 403  may be implemented by using S 4031  and S 4032 . 
     S 4031 : Determine an effective value of third predicted displacement based on the effective value of the first predicted displacement and the effective value of the second predicted displacement. 
     The effective value of the third predicted displacement may be obtained based on the following formula: 
     
       
         
           
             
               
                 X 
                 
                   mean 
                   ⁢ 
                   _ 
                   ⁢ 
                   est 
                 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               KalmanFilter 
               ⁡ 
               
                 [ 
                 
                   
                     
                       X 
                       
                         mean 
                         ⁢ 
                         _ 
                         ⁢ 
                         ts 
                       
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         n 
                       
                       ) 
                     
                   
                   , 
                   
                     
                       X 
                       
                         mean 
                         ⁢ 
                         _ 
                         ⁢ 
                         emf 
                       
                     
                     ⁡ 
                     
                       ( 
                       
                         t 
                         n 
                       
                       ) 
                     
                   
                 
                 ] 
               
             
           
         
       
     
     Herein, X mean_est (t n ) represents the effective value of the third predicted displacement, and KalmanFilter represents a Kalman filter. 
     S 4032 : Determine the displacement correction gain based on the effective value of the first predicted displacement and the effective value of the third predicted displacement. 
     A displacement correction gain G c (t n ) is as follows: 
     
       
         
           
             
               
                 G 
                 c 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               
                 
                   X 
                   
                     mean 
                     ⁢ 
                     _ 
                     ⁢ 
                     est 
                   
                 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     n 
                   
                   ) 
                 
               
               
                 
                   X 
                   
                     mean 
                     ⁢ 
                     _ 
                     ⁢ 
                     ts 
                   
                 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     n 
                   
                   ) 
                 
               
             
           
         
       
     
     S 404 : Determine displacement of the speaker based on the maximum value of the first predicted displacement and the displacement correction gain. 
     Displacement X max_ets (t n ) of the speaker may be obtained based on the following formula: 
     
       
         
           
             
               
                 X 
                 max_est 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               
                 
                   G 
                   c 
                 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     n 
                   
                   ) 
                 
               
               × 
               
                 
                   X 
                   max_ts 
                 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     n 
                   
                   ) 
                 
               
             
           
         
       
     
     S 405 : Determine a signal control gain of the speaker based on the displacement of the speaker and a preset displacement threshold. 
     In this embodiment of this application, the displacement of the speaker is denoted as X max_ets (t n ), the preset displacement threshold is denoted as X th , and the signal control gain of the speaker is as follows: 
     
       
         
           
             
               
                 G 
                 p 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               { 
               
                 
                   
                     
                       
                         
                           
                             X 
                             
                               max 
                               ⁢ 
                               _ 
                               ⁢ 
                               est 
                             
                           
                           ⁡ 
                           
                             ( 
                             
                               t 
                               n 
                             
                             ) 
                           
                         
                         
                           X 
                           th 
                         
                       
                     
                   
                   
                     
                       
                         1 
                         , 
                         
                             
                         
                         ⁢ 
                         others 
                       
                     
                   
                 
                 , 
                 
                   
                     if 
                     ⁢ 
                     
                         
                     
                     ⁢ 
                     
                       
                         X 
                         
                           max 
                           ⁢ 
                           _ 
                           ⁢ 
                           est 
                         
                       
                       ⁡ 
                       
                         ( 
                         
                           t 
                           n 
                         
                         ) 
                       
                     
                   
                   &gt; 
                   
                     X 
                     th 
                   
                 
               
             
           
         
       
     
     S 406 : Perform gain control on the input signal of the speaker based on the signal control gain of the speaker, to obtain an output signal of the speaker. 
     Specifically, the output signal of the speaker may be obtained based on the following formula: 
     
       
         
           
             
               
                 U 
                 out 
               
               ⁡ 
               
                 ( 
                 
                   t 
                   n 
                 
                 ) 
               
             
             = 
             
               
                 
                   G 
                   p 
                 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     n 
                   
                   ) 
                 
               
               × 
               
                 
                   U 
                   in 
                 
                 ⁡ 
                 
                   ( 
                   
                     t 
                     n 
                   
                   ) 
                 
               
             
           
         
       
     
     Optionally, in this embodiment of this application, the input signal may be further multiplied by an optimization coefficient, and the coefficient may be a coefficient related to hardware or a working state of the speaker. 
     In this embodiment of this application, it can be learned, with reference to S 405  and S 406 , that when the displacement of the speaker is greater than the preset displacement threshold, the displacement of the speaker needs to be adjusted (in other words, displacement protection is enabled), so that the displacement does not exceed the preset displacement threshold, to improve a sound effect of the speaker, and improve a subjective experience of a user. Further, a parameter (which is a linear parameter) of the speaker may be identified in real time based on the feedback signal of the speaker, and a first displacement conversion model and a second displacement conversion model are updated, to resolve a problem that displacement protection fails due to a factor such as aging of a component of the speaker. 
     The output signal obtained in S 406  is a digital signal. In this embodiment of this application, a digital-to-analog converter may be used to convert the digital signal into an analog signal, and then the output signal is transmitted, by using an amplifier, to the speaker for playing. 
     Further, the feedback signal of the speaker may be obtained through detection by using a feedback circuit. The feedback signal includes voltage signals and current signals (the voltage signals and the current signals are analog signals) at two ends of the speaker. Then, the voltage signals and the current signals are converted into digital voltages and digital currents by using an analog to digital converter, and then the digital voltages and the digital currents are input into the linear parameter identification model, to obtain the linear parameter (for example, P (t n-1 ) in the foregoing embodiments) of the speaker. 
       FIG. 8  or  FIG. 9  is a flow block diagram of a method for improving a sound effect of a speaker through displacement protection. A difference is as follows: In  FIG. 8 , the displacement correction gain is applied to the maximum value of the first predicted displacement output after first displacement conversion, to obtain the displacement of the speaker. For example, after X max_ts (t n ) is obtained, the displacement of the speaker is obtained based on X max_est  (t n )=G c (t n )×X max_ts (t n ). In  FIG. 9 , the displacement correction gain is applied to a first displacement conversion process. To be specific, the displacement correction gain is directly applied to the first displacement conversion model, to obtain the displacement of the speaker. For example, X max_est (t n )=G c (t n )×max[H ux (t n-1 )*U in (t n )]. Essentially, a procedure shown in  FIG. 8  and a procedure shown in  FIG. 9  correspond to a same method for determining displacement of a speaker. 
     According to the method for improving sound quality of a speaker in this embodiment of this application, first displacement conversion may be performed on the input signal of the speaker, to obtain the maximum value of the first predicted displacement and the effective value of the first predicted displacement. Second displacement conversion is performed on the feedback signal of the speaker, to obtain the effective value of the second predicted displacement. The displacement correction gain is determined based on the effective value of the first predicted displacement and the effective value of the second predicted displacement. Further, the displacement of the speaker is determined based on the maximum value of the first predicted displacement and the displacement correction gain. The signal control gain of the speaker is determined based on the displacement of the speaker and the preset displacement threshold. Gain control is performed on the input signal of the speaker based on the signal control gain of the speaker, to obtain the output signal of the speaker. In the foregoing method, the displacement of the speaker can be determined in real time, and displacement protection is performed on the speaker based on the determined displacement of the speaker (which indicates to perform gain control on the input signal of the speaker based on the signal control gain determined based on the displacement of the speaker), so that a gain of the input signal is reduced, an abrupt change in a speaker volume of the speaker is avoided, it can be ensured that the displacement of the speaker does not exceed an upper safety limit, and the sound effect of the speaker can be improved. 
     Optionally, in an embodiment of this application, a displacement protection method may be applied to signal compensation, to further improve a sound effect of a speaker. Specifically, an output signal obtained in the displacement protection method may be used as the first input signal. Specifically, before S 101 , S 100   a  to S 100   c  may be further included. 
     S 100   a : Determine displacement of the speaker. 
     In this embodiment of this application, the displacement of the speaker may be obtained in S 401  to S 404 . For details, refer to related descriptions in the foregoing embodiment. Details are not described herein again. 
     S 100   b : Determine a signal control gain of the speaker based on the displacement of the speaker and a preset displacement threshold. 
     S 100   c : Perform gain control on a second input signal of the speaker based on the signal control gain of the speaker, to obtain the first input signal. 
     For descriptions of S 100   b  and S 100   c , refer to specific descriptions of S 405  and S 406  in the foregoing embodiment. Details are not described herein again. 
     For example,  FIG. 10  is a flow block diagram of a method for applying displacement protection to signal compensation to improve a sound effect of a speaker. The foregoing method steps may be understood with reference to the flow block diagram of the method. 
     In this embodiment of the present disclosure, function modules of a speaker improvement apparatus may be obtained through division based on the foregoing method examples. For example, each function module may be obtained through division based on each corresponding function, or two or more functions may be integrated into one processing module. The integrated module may be implemented in a form of hardware, or may be implemented in a form of a software function module. It should be noted that division into the modules in embodiments of the present disclosure is an example, and is merely logical function division. There may be another division manner in actual implementation. 
     When each function module is obtained through division based on each corresponding function,  FIG. 11  is a schematic diagram of a possible structure of a sound quality improvement apparatus in the foregoing embodiments. As shown in  FIG. 11 , the sound quality improvement apparatus may include an interpolation module  1001 , a signal compensation module  1002 , and a filtering module  1003 . The interpolation module  1001  may be configured to support the apparatus to perform S 101  (including S 1011  and S 1012 ) in the foregoing method embodiments. The signal compensation module  1002  may be configured to support the apparatus to perform S 102  and S 204  in the foregoing method embodiments. The filtering module  1003  may be configured to support the apparatus to perform S 103  in the foregoing method embodiments. 
     Optionally, as shown in  FIG. 11 , the sound quality improvement apparatus may further include a generation module  1004 , a parameter adjustment module  1005 , and an obtaining module  1006 . The generation module  1004  may be configured to support the apparatus to generate a filtering gain. The parameter adjustment module  1005  may be configured to support the apparatus to perform S 202  (including S 2021  to S 2023 ) and S 203 . The obtaining module  1006  may be configured to support the apparatus to perform S 201 . All related content of the steps in the foregoing method embodiments may be quoted to function descriptions of the corresponding function modules. Details are not described herein again. 
     Optionally, as shown in  FIG. 11 , the sound quality improvement apparatus may further include a displacement determining module  1007 , a control gain determining module  1008 , and a gain control module  1009 . The displacement determining module  1007  may be configured to support the apparatus to perform S 401  to S 404  (S 403  includes S 4031  and S 4032 ) and S 100   a . The control gain determining module  1008  may be configured to support the apparatus to perform S 405  and S 100   b . The gain control module  1009  may be configured to support the apparatus to perform S 406  and S 100   c.    
     When an integrated unit is used,  FIG. 12  is a schematic diagram of a possible structure of an apparatus for improving sound quality of a speaker in the foregoing embodiments. As shown in  FIG. 12 , a sound quality improvement apparatus may include a processing module  2001  and a communications module  2002 . The processing module  2001  may be configured to control and manage an action of the apparatus. For example, the processing module  2001  may be configured to support the apparatus to perform S 101  to S 103 , S 201  to S 204 , S 301  to S 305 , and S 401  to S 406  in the foregoing method embodiments, and/or is configured to execute another process of the technology described in this specification. The communications module  2002  may be configured to support communication between the apparatus and another network entity. Optionally, as shown in  FIG. 12 , the apparatus may further include a storage module  2003 , configured to store program code and data of the apparatus. 
     The processing module  2001  may be a processor or a controller (for example, the processor  110  in  FIG. 1 ), for example, may be a central processing unit (CPU), a general purpose processor, a digital signal processor (DSP), an application-specific integrated circuit (ASIC), a field-programmable gate array (FPGA) or another programmable logic device, a transistor logic device, a hardware component, or any combination thereof. The processing module can implement or execute various example logical blocks, modules, and circuits described with reference to the content disclosed in embodiments of the present disclosure. Alternatively, the processor may be a combination for implementing a computing function, for example, a combination including one or more microprocessors or a combination of a DSP and a microprocessor. The communications module  2002  may be a transceiver, a transceiver circuit, a communications interface, or the like (for example, the mobile communications module  150  or the wireless communications module  160  in  FIG. 1 ). The storage module  2003  may be a memory (for example, the internal memory  121  in  FIG. 1 ). 
     When the processing module  2001  is the processor, the communications module  2002  is the transceiver, and the storage module  2003  is the memory, the processor, the transceiver, and the memory may be connected through a bus. The bus may be a peripheral component interconnect (peripheral component interconnect, PCI) bus, an extended industry standard architecture (extended Industry standard architecture, EISA) bus, or the like. The bus may be classified into an address bus, a data bus, a control bus, and the like. 
     When each function module is obtained through division based on each corresponding function,  FIG. 13  is a schematic diagram of a possible structure of a sound quality improvement apparatus in the foregoing embodiments. As shown in  FIG. 13 , the sound quality improvement apparatus may include a displacement determining module  3001 , a control gain determining module  3002 , and a gain control module  3003 . The displacement determining module  3001  may be configured to support the apparatus to perform S 401  to S 404  (S 403  includes S 4031  and S 4032 ) and S 100   a  in the foregoing method embodiments. The control gain determining module  3002  may be configured to support the apparatus to perform S 405  and S 100   b  in the foregoing method embodiments. The gain control module  3003  may be configured to support the apparatus to perform S 406  and S 100   c  in the foregoing method embodiments. 
     When an integrated unit is used,  FIG. 14  is a schematic diagram of a possible structure of a sound quality improvement apparatus in the foregoing embodiments. As shown in  FIG. 14 , the sound quality improvement apparatus may include a processing module  4001  and a communications module  4002 . The processing module  4001  may be configured to control and manage an action of the apparatus. For example, the processing module  4001  may be configured to support the apparatus to perform S 401  to S 406  and S 100   a  to S 100   c  in the foregoing method embodiments, and/or is configured to execute another process of the technology described in this specification. The communications module  4002  may be configured to support communication between the apparatus and another network entity. Optionally, as shown in  FIG. 14 , the sound quality improvement apparatus may further include a storage module  4003 , configured to store program code and data of the apparatus. 
     The processing module  4001  may be a processor or a controller (for example, may be the processor  110  shown in  FIG. 1 ), for example, may be a CPU, a general-purpose processor, a DSP, an ASIC, an FPGA or another programmable logic device, a transistor logic device, a hardware component, or any combination thereof. The processing module can implement or execute various example logical blocks, modules, and circuits described with reference to the content disclosed in embodiments of the present disclosure. Alternatively, the processor may be a combination for implementing a computing function, for example, a combination including one or more microprocessors or a combination of a DSP and a microprocessor. The communications module  4002  may be a transceiver, a transceiver circuit, a communications interface, or the like (for example, the mobile communications module  150  or the wireless communications module  160  in  FIG. 1 ). The storage module  4003  may be a memory (for example, the internal memory  121  in  FIG. 1 ). 
     When the processing module  4001  is the processor, the communications module  4002  is the transceiver, and the storage module  4003  is the memory, the processor, the transceiver, and the memory may be connected through a bus. The bus may be a PCI bus, an EISA bus, or the like. The bus may be classified into an address bus, a data bus, a control bus, and the like. 
     All or some of the foregoing embodiments may be implemented by using software, hardware, firmware, or any combination thereof. When a software program is used to implement the embodiments, all or some of the embodiments may be implemented in a form of a computer program product. The computer program product includes one or more computer instructions. When the computer instructions are loaded and executed on a computer, all or some of the procedures or functions according to embodiments of this application are generated. The computer may be a general-purpose computer, a dedicated computer, a computer network, or another programmable apparatus. The computer instructions may be stored in a computer-readable storage medium or may be transmitted from a computer-readable storage medium to another computer-readable storage medium. For example, the computer instructions may be transmitted from a website, computer, server, or data center to another website, computer, server, or data center in a wired (for example, a coaxial cable, an optical fiber, or a digital subscriber line (DSL)) or wireless (for example, infrared, radio, or microwave) manner. The computer-readable storage medium may be any usable medium accessible by the computer, or a data storage device, such as a server or a data center, integrating one or more usable media. The usable medium may be a magnetic medium (for example, a floppy disk, a magnetic disk, or a magnetic tape), an optical medium (for example, a digital video disc (DVD)), a semiconductor medium (for example, a solid-state drive (SSD)), or the like. 
     The foregoing descriptions about implementations allow a person skilled in the art to clearly understand that, for the purpose of convenient and brief description, division into the foregoing function modules is taken as an example for illustration. In actual application, the foregoing functions can be allocated to different modules and implemented according to a requirement, that is, an inner structure of an apparatus is divided into different function modules to implement all or part of the functions described above. For a detailed working process of the foregoing system, apparatus, and unit, refer to a corresponding process in the foregoing method embodiments, and details are not described herein again. 
     In the several embodiments provided in this application, it should be understood that the disclosed system, apparatus, and method may be implemented in other manners. For example, the described apparatus embodiment is merely an example. For example, division into the modules or units is merely logical function division and may be other division in actual implementation. For example, a plurality of units or components may be combined or integrated into another system, or some features may be ignored or not performed. In addition, the displayed or discussed mutual couplings or direct couplings or communication connections may be implemented by using some interfaces. The indirect couplings or communication connections between the apparatuses or units may be implemented in electronic, mechanical, or other forms. 
     The units described as separate parts may or may not be physically separate, and parts displayed as units may or may not be physical units, may be located in one position, or may be distributed on a plurality of network units. Some or all of the units may be selected based on actual requirements to achieve the objectives of the solutions of the embodiments. 
     In addition, function units in embodiments of this application may be integrated into one processing unit, or each of the units may exist alone physically, or two or more units may be integrated into one unit. The integrated unit may be implemented in a form of hardware, or may be implemented in a form of a software function unit. 
     When the integrated unit is implemented in the form of a software function unit and is sold or used as an independent product, the integrated unit may be stored in a computer-readable storage medium. Based on such an understanding, the technical solutions of this application essentially, or the part contributing to the conventional technology, or all or some of the technical solutions may be implemented in a form of a software product. The computer software product is stored in a storage medium and includes several instructions for instructing a computer device (which may be a personal computer, a server, a network device, or the like) or a processor to perform all or some of the steps of the methods described in embodiments of this application. The foregoing storage medium includes: any medium that can store program code, for example, a flash memory, a removable hard disk, a read-only memory, a random access memory, a magnetic disk, or an optical disc. 
     The foregoing descriptions are merely specific implementations of this application, but are not intended to limit the protection scope of this application. Any variation or replacement within the technical scope disclosed in this application shall fall within the protection scope of this application. Therefore, the protection scope of this application shall be subject to the protection scope of the claims.