Signal processing system and electronic device

Abstract

Provided in the embodiments of the present application are a signal processing system and an electronic device. The system comprises: an input signal module, a displacement model, a filter coefficient generation module, a group delay compensation module, a tunable equalizer, an amplifier and an audio transducer, wherein the input signal module is connected to both the group delay compensation module and the displacement model, the displacement model is connected to the filter coefficient generation module, the group delay compensation module is connected to the tunable equalizer, the tunable equalizer is connected to the amplifier, the amplifier is connected to the audio transducer, and the filter coefficient generation module is connected to the tunable equalizer. In the technical solution provided in the embodiments of the present application, frequency response variations related to driver displacement can be corrected within a wide frequency range, thereby reducing distortion, especially intermodulation distortion.

Description

A signal processing system and electronic device Technical Field

[0001] This application relates to the field of acoustic technology, and more particularly to a signal processing system and electronic device. Background Technology

[0002] One of the main mechanisms that produce intermodulation distortion in the mid and high frequencies (i.e., much higher than the driver base resonance) is the change in frequency response caused by the movement of the loudspeaker diaphragm. Technical issues

[0003] While existing technologies can address the nonlinearity of the transducer itself, they cannot resolve the issue of mid-to-high frequency response caused by variations in system parameters. Limiting low-frequency components to mitigate mid-to-high frequency modulation distortion inevitably leads to sound coloration and distortion. Technical solutions

[0004] In view of this, embodiments of this application provide a signal processing system and electronic device for correcting frequency response variations related to driver displacement over a wide frequency range, thereby reducing distortion, particularly intermodulation distortion.

[0005] On one hand, embodiments of this application provide a signal processing system applied to an electronic device, including: an input signal module, a displacement model, a filter coefficient generation module, a group delay compensation module, an adjustable equalizer, an amplifier, and an audio converter;

[0006] The input signal module is connected to the group delay compensation module and the displacement model, respectively. The displacement model is connected to the filter coefficient generation module. The group delay compensation module is connected to the adjustable equalizer. The adjustable equalizer is connected to the amplifier. The amplifier is connected to the audio converter. The filter coefficient generation module is connected to the adjustable equalizer.

[0007] Optionally, it further includes: a low-pass filter, wherein the displacement model is connected to the low-pass filter, and the low-pass filter is connected to the filter coefficient generation module.

[0008] Optionally, the low-pass filter includes a first low-pass filter, and the system further includes a port velocity model and a second low-pass filter. The input signal module is connected to the port velocity model, the port velocity model is connected to the second low-pass filter, and the second low-pass filter is connected to the filter coefficient generation module.

[0009] Optionally, the low-pass filter includes a first low-pass filter, and the system further includes a third low-pass filter, a high-pass filter, and an adder;

[0010] The group delay compensation module is connected to the third low-pass filter, the third low-pass filter is connected to the adder, the group delay compensation module is connected to the high-pass filter, the high-pass filter is connected to the adjustable equalizer, the adjustable equalizer is connected to the adder, and the adder is connected to the amplifier.

[0011] Optionally, the low-pass filter includes a first low-pass filter, the adjustable equalizer includes an adjustable high-frequency equalizer, and the system further includes: a third low-pass filter, an adjustable low-frequency equalizer, a high-pass filter, and an adder.

[0012] The group delay compensation module is connected to the third low-pass filter, the third low-pass filter is connected to the adjustable low-frequency equalizer, the adjustable low-frequency equalizer is connected to the adder, the group delay compensation module is connected to the high-pass filter, the high-pass filter is connected to the adjustable high-frequency equalizer, the adjustable high-frequency equalizer is connected to the adder, and the adder is connected to the amplifier; the filter coefficient generation module is connected to both the adjustable low-frequency equalizer and the adjustable high-frequency equalizer.

[0013] Optionally, the filter coefficient generation module is used to obtain filter coefficients through an interpolation lookup table, an interpolation function, or machine learning.

[0014] Optionally, the group delay compensation module is used to compensate for the group delay of the displacement model and the low-pass filter.

[0015] Optionally, the audio converter includes a speaker, headphones, or an electroacoustic or electromechanical transducer.

[0016] Optionally, the displacement model includes a calculation implementation model of the loudspeaker diaphragm displacement.

[0017] On the other hand, embodiments of this application provide an electronic device including the signal processing system described above. Beneficial effects

[0018] The technical solution provided in this application includes an input signal module, a displacement model, a filter coefficient generation module, a group delay compensation module, an adjustable equalizer, an amplifier, and an audio converter. The input signal module is connected to both the group delay compensation module and the displacement model. The displacement model is connected to the filter coefficient generation module. The group delay compensation module is connected to the adjustable equalizer. The adjustable equalizer is connected to the amplifier. The amplifier is connected to the audio converter. The filter coefficient generation module is also connected to the adjustable equalizer. The technical solution provided in this application can correct driver displacement-related frequency response changes over a wide frequency range, thereby reducing distortion, especially intermodulation distortion. Attached Figure Description

[0019] To more clearly illustrate the technical solutions of the embodiments of this application, the drawings used in the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments of this application. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0020] Figure 1 is a schematic diagram of the structure of a loudspeaker with a resonator provided in an embodiment of this application;

[0021] Figure 2 is a schematic diagram of the measurement of displacement-related frequency response in a miniature loudspeaker with a resonant front cavity provided in an embodiment of this application;

[0022] Figure 3 is a schematic diagram of the two-tone intermodulation spectrum provided in an embodiment of this application;

[0023] Figure 4 is a schematic diagram of the multi-frequency sound spectrum provided in the embodiment of this application;

[0024] Figure 5 is a schematic diagram of a signal processing system provided in an embodiment of this application;

[0025] Figure 6 is a schematic diagram of another signal processing system provided in an embodiment of this application;

[0026] Figure 7 is a schematic diagram of another signal processing system provided in an embodiment of this application;

[0027] Figure 8 is a schematic diagram of another signal processing system provided in an embodiment of this application;

[0028] Figure 9 is a schematic diagram of the structure of a time-varying second-order IIR equalizer block provided in an embodiment of this application;

[0029] Figure 10 is a schematic diagram showing the change of response with diaphragm movement according to an embodiment of this application;

[0030] Figure 11 is a schematic diagram of equalizing the position-related equalizer response in Figure 10 to the nominal target according to an embodiment of this application;

[0031] Figure 12 is a schematic diagram of the signal spectrum and distortion spectrum provided in the embodiments of this application;

[0032] Figure 13 is a schematic diagram of another signal processing system provided in an embodiment of this application;

[0033] Figure 14 is a schematic diagram of another signal processing system provided in an embodiment of this application. The best embodiment of the present invention

[0034] To better understand the technical solution of this application, the embodiments of this application will be described in detail below with reference to the accompanying drawings.

[0035] It should be understood that the described embodiments are merely some, not all, of the embodiments in this application. All other embodiments obtained by those skilled in the art based on the embodiments in this application without inventive effort are within the scope of protection of this application.

[0036] The terminology used in the embodiments of this application is for the purpose of describing particular embodiments only and is not intended to be limiting of this application. The singular forms “a,” “the,” and “the” used in the embodiments of this application and the appended claims are also intended to include the plural forms unless the context clearly indicates otherwise.

[0037] It should be understood that the term "and / or" used in this article is merely a description of the relationship between related objects, indicating that three relationships can exist. For example, A and / or B can represent: A existing alone, A and B existing simultaneously, or B existing alone. Additionally, the character " / " in this article generally indicates that the preceding and following related objects have an "or" relationship.

[0038] Since the modulation distortion mechanisms associated with frequency variations in information processing systems have not been discussed in the scientific literature, there are no methods in related technologies to address distortion caused by changes in mid-to-high frequency response. The technical solutions in related technologies are related to reducing distortion caused by transducer nonlinearity, which affects system performance, especially near the fundamental resonance of the driver. Some nonlinear echo cancellation techniques in related technologies share similar characteristics with transducer nonlinearity compensation techniques; an overview of the most common solutions is presented.

[0039] One of the main mechanisms that produce intermodulation distortion in the mid and high frequencies (i.e., well above the driver's fundamental resonance) is the change in frequency response caused by the movement of the loudspeaker diaphragm. In typical miniature loudspeaker applications, the loudspeaker radiates to the outside air through the resonant cavity, and the change in response is largely caused by the geometry of the loudspeaker's front cavity and related acoustic changes. The change in overall sensitivity also contributes due to the position dependence of the magnetic flux seen by the voice coil.

[0040] Transducer nonlinearity is caused by one or more of the following factors: nonlinear force factor (Bl), which is a function of both displacement (the primary factor) and drive current (a secondary factor); nonlinear compliance; and nonlinear mechanical damping. Voice coil resistance is another cause of nonlinearity, considered a necessary condition for successful nonlinear prediction and compensation. However, unlike other nonlinear factors, it does not depend on the instantaneous value of displacement but rather on the heating and cooling history of the voice coil, thus requiring specialized models and compensation methods. Other nonlinear mechanisms can be compensated using memoryless methods. The estimation of voice coil resistance is well-known from loudspeaker protection algorithms. One method of nonlinear compensation is to use mirror filters, implemented as polynomial nonlinear functions that are the inverse of the observed transducer nonlinearity, using appropriate pre-emphasis and de-emphasis filters around the nonlinear element. The use of nonparametric methods (i.e., model parameters not directly described by physical variables) has wide applications in nonlinear system identification.

[0041] The technical solutions in related technologies mainly address the nonlinearity of the transducer itself, focusing primarily on low-frequency performance rather than the mid-to-high frequency response caused by parameter variations in the information processing system. Acoustic solutions to this problem exist, but their scope is limited to specific applications, and they cannot compensate for the impact of transducer nonlinearity and acoustic parameter variations on mid-to-high frequency performance. Algorithms that attempt to address mid-to-high frequency modulation distortion by limiting low-frequency content inevitably lead to sound coloration.

[0042] Figure 1 is a schematic diagram of a loudspeaker with a resonator according to an embodiment of this application. As shown in Figure 1, the loudspeaker includes a front cavity 1 and an output port 2. The diaphragm 3 can be shown at the nominal position (solid line), upward (dotted line), and downward (dashed line). The main effect causing the change in frequency response shape is the change in the internal volume of the front cavity 1, which leads to a change in the resonant cavity formed by the front cavity 1 and the output port 2.

[0043] Figure 2 is a schematic diagram of the measurement of displacement-related frequency response in a miniature loudspeaker with a resonant front cavity provided in the embodiment of this application. As shown in Figure 2, the vertical axis is the FR magnitude, the horizontal axis is the cursor 20.0 Hz, 102.85 dB, and the original (damped) effect of bias.

[0044] Measurement results show that strong intermodulation distortion exists around the resonant frequency of front cavity 1. Figure 3 is a schematic diagram of the two-tone intermodulation spectrum provided in the embodiment of this application. The vertical axis of Figure 3 represents the signal level (also known as signal level), in dB, and the horizontal axis represents the frequency (in Hz). The two-tone intermodulation distortion and response changes of the loudspeaker can be seen in Figure 2. Figure 4 is a schematic diagram of the multi-frequency sound spectrum provided in the embodiment of this application. The vertical axis of Figure 4 represents the signal level (in dB), and the horizontal axis represents the frequency (in Hz). The multi-tone distortion of the loudspeaker and its response changes can be seen in Figure 2. Other mechanisms may also cause response changes, especially in larger loudspeakers (i.e., not miniature loudspeakers), which can be addressed by signal processing systems similar to those described in the embodiments of this application. In larger systems, an important mechanism is the change in voice coil inductance with diaphragm movement, which is generally considered one of the main sources of mid-range and high-range intermodulation distortion over a wide frequency range, as well as changes in cavity resonance caused by the air space behind the diaphragm and the holes on the loudspeaker chassis, which typically affect a narrower frequency range. Other mechanisms can also cause displacement-related response changes, therefore the scope of the embodiments of this application is not limited to compensating for distortions that are only related to the anterior cavity.

[0045] To address the technical problems raised in related technologies, embodiments of this application provide a signal processing system that can correct frequency response variations related to driver displacement over a wide frequency range, thereby reducing distortion, particularly intermodulation distortion.

[0046] Figure 5 is a schematic diagram of a signal processing system provided in an embodiment of this application. As shown in Figure 5, the system includes: an input signal module 11, a displacement model 12, a first low-pass filter 13, a filter coefficient generation module 14, a group delay compensation module 15, an adjustable equalizer 16, an amplifier 17, and an audio converter 18.

[0047] The input signal module 11 is connected to the group delay compensation module 15 and the displacement model 12 respectively. The displacement model 12 is connected to the first low-pass filter 13. The first low-pass filter 13 is connected to the filter coefficient generation module 14. The group delay compensation module 15 is connected to the adjustable equalizer 16. The adjustable equalizer 16 is connected to the amplifier 17. The amplifier 17 is connected to the audio converter 18. The filter coefficient generation module 14 is connected to the adjustable equalizer 16.

[0048] In this embodiment of the application, the filter coefficient generation module 14 is used to obtain filter coefficients through interpolation lookup table, interpolation function or machine learning.

[0049] In this embodiment, the group delay compensation module 15 is used to compensate for the group delay of the displacement model 12 and the first low-pass filter 13.

[0050] In this embodiment, the audio converter 18 includes a speaker, headphones, and an electroacoustic or electromechanical transducer. The frequency response of the signal processing system can display frequency response changes related to temporary values ​​of system state parameters, such as transducer displacement, velocity of the acoustic system portion, or internal temperature.

[0051] In this embodiment of the application, the displacement model 12 includes a calculation implementation model for the diaphragm displacement of the loudspeaker.

[0052] In this embodiment, amplifier 17 can be implemented as a basic linear filter, and the filter coefficients for each sample or short frame can be updated using displacement information.

[0053] In this embodiment, the purpose of the filter is to balance the difference between the target frequency response at the stationary position of the diaphragm and the actual frequency response measured or calculated at the stationary position of the diaphragm.

[0054] In this embodiment, under certain circumstances, the flow rate caused by low-frequency signals in the port can significantly affect the loss in the port. In such cases, filter parameter adjustment can benefit from using the port speed as a control parameter. Figure 6 is a schematic diagram of another signal processing system provided in this embodiment. As shown in Figure 6, the system includes: an input signal module 11, a displacement model 12, a first low-pass filter 13, a filter coefficient generation module 14, a group delay compensation module 15, an adjustable equalizer 16, an amplifier 17, and an audio converter 18.

[0055] The input signal module 11 is connected to the group delay compensation module 15 and the displacement model 12 respectively. The displacement model 12 is connected to the first low-pass filter 13. The first low-pass filter 13 is connected to the filter coefficient generation module 14. The group delay compensation module 15 is connected to the adjustable equalizer 16. The adjustable equalizer 16 is connected to the amplifier 17. The amplifier 17 is connected to the audio converter 18. The filter coefficient generation module 14 is connected to the adjustable equalizer 16.

[0056] The system also includes: a port velocity model 19 and a second low-pass filter 20. The input signal module 11 is connected to the port velocity model 19, the port velocity model 19 is connected to the second low-pass filter 20, and the second low-pass filter 20 is connected to the filter coefficient generation module 14.

[0057] In this embodiment, when correcting low-frequency distortion associated with the transducer (in this case, "low frequency" means near the fundamental resonance of the transducer), the position-dependent signal component that causes it also needs to be compensated. Therefore, the compensation process needs to be time-dependent and nonlinear simultaneously. On the other hand, in this embodiment, the nonlinearity is caused by a low-frequency signal, and the signal to be compensated has a higher frequency than the signal component that causes the nonlinearity. Furthermore, the signal to be compensated exhibits a linear relationship with respect to its own signal amplitude, but through a time-varying system. Therefore, the compensation filter needs to be time-dependent, but can be linear with respect to the signal amplitude. This greatly simplifies the design of the correction filter. The nonlinear driving model can be adaptively updated using feedback information, improving the compensation accuracy of all systems. This feedback signal can come from speaker current measurements, speaker current and voltage measurements, or as information from a separate feedback sensor.

[0058] Figure 7 is a schematic diagram of another signal processing system provided in an embodiment of this application. As shown in Figure 7, the system includes: an input signal module 11, a nonlinear time-varying controller 71, a nonlinear driving model 72, an amplifier 17, and an audio converter 18.

[0059] The input signal module 11 is connected to the nonlinear time-varying controller 71 and the nonlinear driving model 72, respectively. The nonlinear time-varying controller 71 is connected to the amplifier 17, the amplifier 17 is connected to the audio converter 18, and the nonlinear driving model 72 is connected to the nonlinear time-varying controller 71. Figure 7 can be seen as the general structure of a traditional nonlinear signal processing system, which is usually used for low-frequency nonlinear compensation of the driver.

[0060] Figure 8 is a schematic diagram of another signal processing system provided in an embodiment of this application. As shown in Figure 8, the system includes: an input signal module 11, a linear time-varying equalizer 81, a linear or nonlinear driving model 82, an amplifier 17, and an audio converter 18.

[0061] The input signal module 11 is connected to the linear time-varying equalizer 81 and the linear or nonlinear driving model 82, respectively. The linear time-varying equalizer 81 is connected to the amplifier 17, the amplifier 17 is connected to the audio converter 18, and the linear time-varying equalizer 81 is connected to the linear or nonlinear driving model 82. Figure 8 can be a general structure of a traditional nonlinear signal processing system, and can also be used for compensation of high-frequency response changes in drivers.

[0062] Traditional low-frequency nonlinear compensation and voice coil temperature compensation algorithms can be used in conjunction with the embodiments of this application, which may improve the performance of the embodiments of this application because the diaphragm displacement can be better predicted. The same driver displacement model can also be used for multiple signal processing systems.

[0063] Time-varying equalizers are typically implemented as dual-group Infinite Impulse Response (IIR) filters, which can consist of multiple filter blocks. Figure 9 shows an example of the adjustable equalizer 16 in Figure 5 or Figure 6, implemented as a second-order filter block in "direct form". An equalizer can consist of one or more such second-order filter blocks, or it can consist of one or more simpler first-order filter blocks. Figure 9 is a schematic diagram of the structure of a time-varying second-order IIR equalizer block provided in an embodiment of this application. The filter coefficient model in Figure 9 can be a lookup table, interpolation function, etc., and the filter coefficients a and b can be updated for each sample.

[0064] Figure 10 is a schematic diagram of the response change with diaphragm movement provided in the embodiment of this application. Figure 1 shows the simplified frequency response of the speaker front cavity considering the driver displacement effect. The frequency scale is normalized to the nominal resonant frequency of the speaker front cavity. As shown in Figure 10, the horizontal axis is the normalized frequency and the vertical axis is the response. Figure 10 shows the schematic diagram of the nominal, downward, and downward responses changing with diaphragm movement. Figure 10 represents the simplified system frequency response behavior in Figure 1, considering only the influence of the main high-frequency cavity resonant movement.

[0065] Figure 11 is a schematic diagram of equalizing the position-dependent equalizer response in Figure 10 to the nominal target according to an embodiment of this application. As shown in Figure 11, the horizontal axis is the normalized frequency and the vertical axis is the response. Figure 11 shows the diaphragm position-dependent equalizer response required to correct the upward and downward responses in Figure 10 to be the same as the nominal response. This equalization can be achieved, for example, by the architecture shown in Figure 9.

[0066] Figure 12 shows the simulation effect of an actual music signal in one embodiment of this application. Figure 12 is a schematic diagram of the signal spectrum and distortion spectrum provided in the embodiment of this application. As shown in Figure 12, the horizontal axis represents frequency, and the vertical axis represents signal level. Figure 12 shows the original signal spectrum, the uncorrected distortion spectrum, and the corrected distortion spectrum. Figure 12 represents the distortion improvement in the simulation. Distortion refers to the difference between the actual output spectrum and the output spectrum of a linear system with the same average frequency response but without distortion. The dotted line (uncorrected distortion spectrum) shows the difference spectrum of the nonlinear loudspeaker simulation model without applying the correction algorithm, while the dashed line (corrected distortion spectrum) shows the situation after applying the correction.

[0067] The driver's displacement model generates an output signal from the input signal module, which corresponds to the expected displacement of the driver within the driver's housing. The precise implementation of this model is not critical to the embodiments of this application. The simplest model can be a low-pass filter; further improvements can use displacement-related nonlinear models, as well as models that consider changes in voice coil temperature, which can be derived from computational thermal models or estimated from driver voltage and current measurements. The model can also be fully adaptive, based on voltage and current information, or, in larger speakers, feedback sensor information.

[0068] Figure 13 is a schematic diagram of another signal processing system provided in an embodiment of this application. As shown in Figure 13, the system includes: an input signal module 11, a displacement model 12, a first low-pass filter 13, a filter coefficient generation module 14, a group delay compensation module 15, an adjustable equalizer 16, an amplifier 17, an audio converter 18, a third low-pass filter 21, and a high-pass filter 22.

[0069] The input signal module 11 is connected to the group delay compensation module 15 and the displacement model 12 respectively. The displacement model 12 is connected to the first low-pass filter 13. The first low-pass filter 13 is connected to the filter coefficient generation module 14. The amplifier 17 is connected to the audio converter 18. The filter coefficient generation module 14 is connected to the adjustable equalizer 16. The group delay compensation module 15 is connected to the third low-pass filter 21. The third low-pass filter 21 is connected to the adder. The group delay compensation module 15 is connected to the high-pass filter 22. The high-pass filter 22 is connected to the adjustable equalizer 16. The adjustable equalizer 16 is connected to the adder. The adder is connected to the amplifier 17.

[0070] In this embodiment, if the frequencies of the high-frequency resonance being corrected are separated, and the corresponding equalizer's frequency range is small, then the equalizer designed to correct the high-frequency response may have a certain adverse effect on the distortion in the low-frequency range. In these cases, a high-pass / low-pass filter can be used to bypass the low-frequency equalization, as shown in Figure 13. Alternatively, separate processing can be applied to the low frequencies, as shown in Figure 14.

[0071] Figure 14 is a schematic diagram of another signal processing system provided in an embodiment of this application. As shown in Figure 14, the system includes: an input signal module 11, a displacement model 12, a first low-pass filter 13, a filter coefficient generation module 14, a group delay compensation module 15, an adjustable high-frequency equalizer 161, an amplifier 17, an audio converter 18, a third low-pass filter 21, a high-pass filter 22, and an adjustable low-frequency equalizer 23.

[0072] The input signal module 11 is connected to the group delay compensation module 15 and the displacement model 12, respectively. The displacement model 12 is connected to the first low-pass filter 13. The first low-pass filter 13 is connected to the filter coefficient generation module 14. The amplifier 17 is connected to the audio converter 18. The group delay compensation module 15 is connected to the third low-pass filter 21. The third low-pass filter 21 is connected to the adjustable low-frequency equalizer 23. The adjustable low-frequency equalizer 23 is connected to the adder. The group delay compensation module 15 is connected to the high-pass filter 22. The high-pass filter 22 is connected to the adjustable high-frequency equalizer 161. The adjustable high-frequency equalizer 161 is connected to the adder. The adder is connected to the amplifier 17. The filter coefficient generation module 14 is connected to the adjustable low-frequency equalizer 23 and the adjustable high-frequency equalizer 161, respectively.

[0073] In this embodiment, the signal processing system may also use analog filters in whole or in part. The analog implementation of the analog filter includes a control path for the displacement model, constructed as a low-pass filter (digital or analog), a nonlinear function mapping the displacement data to filter control parameters (which can be implemented in a fully analog implementation, e.g., using an operational amplifier), and a drive signal path containing both a full-pass filter and a voltage-controlled filter. The actual implementation may also include a hybrid of analog and digital functions, for example, an analog voltage-controlled filter (VCF) with digital displacement model and digitally implemented control lookup control.

[0074] In this application, the application of the algorithm is not limited to frequency response changes caused by system geometry variations. As another example of a source of response variation, particularly in larger loudspeakers, the displacement dependence of the voice coil inductance causes mid-to-high frequency response variations, which are generally considered a major source of mid-to-high frequency intermodulation distortion in loudspeakers. When the driver is subjected to static displacement, the inductive effect can be measured by the frequency response change and therefore compensated for using the same algorithm. Furthermore, when the diaphragm shifts from its static position, the driver sensitivity typically changes, primarily due to changes in the B1 factor, and this effect can be incorporated into a lookup table. Changes in the temperature of the driving voice coil in a dynamic loudspeaker also cause changes in sensitivity and response shape, which can be considered as an additional control parameter.

[0075] Some algorithms reduce "piano distortion" when detecting mid-frequency signals by reducing strong low-frequency content, but the methods they use inevitably alter the tonal balance of the signal.

[0076] In this embodiment, the signal processing system may include a real-time simulation model of the system state parameters and a variable equalization unit, the parameters of which are also controlled in real time using the system model. The parameters of the controllable equalizer unit are adjusted to compensate for the system's frequency response in the nominal (small signal) state and the response determined by the actual displacement position of the sensor moving component and possibly by other system variables (internal temperature, etc.). The equalization unit is located on the signal path connecting the amplifier and the transducer, and this signal path includes a delay compensation unit for compensating for the group delay of the transducer model.

[0077] The technical solution provided in this application includes an input signal module, a displacement model, a filter coefficient generation module, a group delay compensation module, an adjustable equalizer, an amplifier, and an audio converter. The input signal module is connected to both the group delay compensation module and the displacement model. The displacement model is connected to the filter coefficient generation module. The group delay compensation module is connected to the adjustable equalizer. The adjustable equalizer is connected to the amplifier. The amplifier is connected to the audio converter. The filter coefficient generation module is also connected to the adjustable equalizer. The technical solution provided in this application can correct driver displacement-related frequency response changes over a wide frequency range, thereby reducing distortion, especially intermodulation distortion.

[0078] The technical solution provided in this application can reduce distortion while reducing frequency response variations. An alternative method could be based on intentionally altering the tone balance of the signal. The model has low computational complexity. The signal processing system can be applied to a wide range of electroacoustic systems, not just miniature loudspeakers.

[0079] This application provides an electronic device including the signal processing system described above.

[0080] The above description is merely a preferred embodiment of this application and is not intended to limit this application. Any modifications, equivalent substitutions, improvements, etc., made within the spirit and principles of this application should be included within the scope of protection of this application.

Claims

1. A control method for a vibration motor, characterized in that, include: Obtain the audio signal corresponding to the audio content; The audio signal is processed to generate a processed audio signal; Identify the spatial information of the processed audio signal; Based on the spatial information, control parameters for the vibration motor are generated; According to the control parameters, a vibration signal corresponding to the control parameters is output to the vibration motor to control the vibration motor.

2. The method according to claim 1, characterized in that, The generation of control parameters for the vibration motor based on the spatial information includes: The spatial information is calculated using a vibration feedback effect model to generate control parameters for the vibration motor.

3. The method according to claim 1, characterized in that, After processing the audio signal to generate the processed audio signal, the process further includes: The processed audio signal is converted into a vibration effect signal; Identify the spatial information of the processed audio signal; Based on the spatial information, control parameters for the vibration motor are generated; Based on the vibration effect signal and the control parameters, a vibration signal corresponding to the vibration effect signal and the control parameters is output to the vibration motor.

4. The method according to claim 1, characterized in that, After acquiring the audio signal corresponding to the audio content, the process further includes: Identify the spatial information of the audio signal; Based on the spatial information, control parameters for the vibration motor are generated; According to the control parameters, a vibration signal corresponding to the control parameters is output to the vibration motor.

5. The method according to claim 1, characterized in that, After identifying the spatial information of the processed audio signal, the process includes: Based on the spatial information, the audio signal or preset vibration signal is processed to generate a new vibration signal; Based on the spatial information, control parameters for the vibration motor are generated; Based on the control parameters, the new vibration signal is output and distributed to the vibration motor to control the vibration motor.

6. The method according to claim 5, characterized in that, The step of processing the audio signal or preset vibration signal based on the spatial information to generate a new vibration signal includes: Based on the spatial information, the audio signal or preset vibration signal is converted or modified to generate a new vibration signal.

7. A control device for a vibration motor, characterized in that, include: The audio acquisition module is used to acquire the audio signal corresponding to the audio content; An audio signal processing module is used to process the audio signal and generate a processed audio signal; The spatial information analysis module is used to identify the spatial information of the processed audio signal; A multi-motor control parameter generation module is used to generate control parameters for the vibration motor based on the spatial information. A multi-motor control module is used to output a vibration signal corresponding to the control parameters to the vibration motor according to the control parameters, so as to control the vibration motor.

8. The apparatus according to claim 7, characterized in that, The multi-motor control parameter generation module is specifically used to calculate the spatial information using a vibration feedback effect model to generate control parameters for the vibration motor.

9. A storage medium, characterized in that, The storage medium includes a stored program, wherein, when the program is executed, it controls the device containing the storage medium to perform the control method of the vibration motor according to any one of claims 1 to 6.

10. An electronic device comprising a memory and a processor, the memory for storing information including program instructions, and the processor for controlling the execution of the program instructions, characterized in that, When the program instructions are loaded and executed by the processor, they implement the steps of the control method for the vibration motor according to any one of claims 1 to 6.

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