Signal processing system and electronic equipment

Abstract

The embodiment of the invention provides a signal processing system and electronic equipment. The system comprises 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 respectively connected with the group delay compensation module and the displacement model, the displacement model is connected with the filter coefficient generation module, the group delay compensation module is connected with the adjustable equalizer, the adjustable equalizer is connected with the amplifier, the amplifier is connected with the audio converter, and the filter coefficient generation module is connected with the adjustable equalizer. According to the technical scheme provided by the embodiment of the invention, the frequency response change related to the displacement of the driver can be corrected in a very wide frequency range, so that distortion, particularly intermodulation distortion, is reduced.

Description

Technical Field

[0001] The present invention relates to the field of acoustic technologies, and in particular, to a signal processing system and an electronic device.

Background Art

[0002] One of the main mechanisms for generating intermodulation distortion in the mid-frequency and high-frequency ranges (i.e., far above the driver's fundamental resonance) is the frequency response change caused by the movement of the speaker diaphragm.

[0003] The technical solutions in the related art can solve the non-linearity problem of the transducer itself, but cannot solve the problem of the mid-high frequency response caused by the change of system parameters. If the mid / high frequency modulation distortion is solved by restricting the content of the low-frequency frequency, it is easy to inevitably cause sound coloring and the sound is prone to distortion.

Summary of the Invention

[0004] In view of this, embodiments of the present invention provide a signal processing system and an electronic device, which are used to correct the frequency response change related to the driver displacement in a very wide frequency range, thereby reducing distortion, especially intermodulation distortion.

[0005] On the one hand, an embodiment of the present invention provides a signal processing system, which is applied to an electronic device and includes: 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 converter;

[0006] The input signal module is respectively connected to 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 converter, and the filter coefficient generation module is connected to the tunable equalizer.

[0007] Optionally, it further includes: a low-pass filter, 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 respectively connected to 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 the group delay of the displacement model and the low-pass filter.

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

[0016] Optionally, the displacement model includes a computational implementation model of the diaphragm displacement of the speaker.

[0017] On the other hand, an embodiment of the present invention provides an electronic device including the above signal processing system.

[0018] In the technical solution provided by the embodiment of the present invention, there are 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 respectively connected to 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, and the filter coefficient generation module is connected to the adjustable equalizer. In the technical solution provided by the embodiment of the present invention, the frequency response change related to the driver displacement can be corrected within a very wide frequency range, thereby reducing distortion, especially intermodulation distortion.

Description of the Drawings

[0019] To more clearly illustrate the technical solutions of the embodiments of the present invention, the following will briefly introduce the drawings required for the embodiments. Obviously, the drawings in the following description are only some embodiments of the present invention. For those of ordinary skill in the art, without creative efforts, other drawings can also be obtained based on these drawings.

[0020] Figure 1 Schematic structural diagram of a loudspeaker of a resonator provided by an embodiment of the present invention;

[0021] Figure 2 Schematic measurement diagram of displacement-related frequency response in a micro-loudspeaker with a resonant front cavity provided by an embodiment of the present invention;

[0022] Figure 3 Schematic diagram of two-tone intermodulation spectrum provided by an embodiment of the present invention;

[0023] Figure 4 Schematic diagram of multi-frequency sound spectrum provided by an embodiment of the present invention;

[0024] Figure 5 Schematic structural diagram of a signal processing system provided by an embodiment of the present invention;

[0025] Figure 6 Schematic structural diagram of another signal processing system provided by an embodiment of the present invention;

[0026] Figure 7 Schematic structural diagram of another signal processing system provided by an embodiment of the present invention;

[0027] Figure 8 Schematic structural diagram of another signal processing system provided by an embodiment of the present invention;

[0028] Figure 9 Schematic structural diagram of a time-varying second-order IIR equalizer block provided by an embodiment of the present invention;

[0029] Figure 10 Schematic diagram of the change of response with the movement of the diaphragm provided by an embodiment of the present invention;

[0030] Figure 11 For the embodiment of the present invention, Figure 10 Schematic diagram of equalizing the position-related equalizer response in to the nominal target;

[0031] Figure 12 Schematic diagram of signal spectrum and distortion spectrum provided by an embodiment of the present invention;

[0032] Figure 13 Schematic structural diagram of another signal processing system provided by an embodiment of the present invention;

[0033] Figure 14 This is a schematic structural diagram of another signal processing system provided by an embodiment of the present invention.

Specific Embodiments

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

[0035] It should be clear that the described embodiments are only a part of the embodiments of the present invention, rather than all embodiments. All other embodiments obtained by those of ordinary skill in the art based on the embodiments of the present invention without creative efforts fall within the scope of protection of the present invention.

[0036] The terms used in the embodiments of the present invention are only for the purpose of describing specific embodiments, and are not intended to limit the present invention. The singular forms of "a", "the" and "said" used in the embodiments of the present invention 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 / " used herein is only a description of the association relationship of associated objects, indicating that three relationships may exist. For example, a and / or b may represent: a exists alone, a and b exist simultaneously, and b exists alone. In addition, the character " / " in this article generally represents an "or" relationship between the associated objects before and after.

[0038] Since the modulation distortion mechanism related to the frequency change of the information processing system has not been discussed in the scientific literature, there is no method in the related art for the distortion caused by the change of the mid-high frequency response. The technical solutions in the related art are related to reducing the distortion caused by the converter nonlinearity, which affects the system performance, especially near the fundamental resonance of the driver. Some technical solutions for nonlinear echo cancellation in the related art have similar characteristics to the technical solutions for transducer nonlinear compensation, and an overview of the most common solutions is proposed.

[0039] One of the main mechanisms for generating intermodulation distortion in the mid-frequency and high-frequency (i.e., far above the fundamental resonance of the driver) is the change in the frequency response caused by the movement of the speaker diaphragm. In a typical micro-speaker application, the speaker radiates to the external air through a resonant cavity, and the response change is largely caused by the geometry of the speaker front cavity and the related acoustic changes. Due to the position dependence of the magnetic flux seen by the voice coil, the change in the overall sensitivity also makes a certain contribution.

[0040] Transducer non-linearity is caused by one or more of the following factors: the non-linear force factor (Bl), which is a function of both displacement (the main factor) and drive current (the secondary factor), non-linear compliance, and non-linear mechanical damping. The voice coil resistance is another cause of non-linearity. Considering that it is a necessary condition for successful non-linear prediction and compensation, it does not depend on the instantaneous value of displacement like other non-linear factors, but rather on the heating and cooling history of the voice coil. Therefore, a special model and compensation method are required, while other non-linear mechanisms can be compensated using memoryless methods. The estimation of voice coil resistance is well-known from speaker protection algorithms. One method of non-linear compensation is to use mirror filters, which are implemented as polynomial non-linear functions that are the inverse of the observed transducer non-linearity, with appropriate pre-emphasis and de-emphasis filters used around the non-linear elements. The use of non-parametric (i.e., the model parameters are not directly described by physical variables) can be widely applied in non-linear system identification.

[0041] The technical solutions in the related art mainly solve the non-linearity problem of the transducer itself, mainly focusing on low-frequency performance rather than the mid- and high-frequency frequency response caused by parameter changes in the information processing system. Acoustic solutions to this problem exist, but their scope is limited to a limited range of applications, and they cannot compensate for systems where transducer non-linearity and acoustic parameter changes affect mid / high-frequency performance. Algorithms that solve mid / high-frequency modulation distortion by limiting low-frequency content will inevitably lead to sound coloring.

[0042] Figure 1 The structural schematic diagram of a speaker with a resonator provided by an embodiment of the present invention is shown in Figure 1 As shown, the speaker includes a front cavity 1 and an output port 2. The diaphragm 3 can be shown in the nominal position (solid line), displaced upward (dashed-dotted line), and displaced downward (dashed line). The main effect causing the change in the shape of the frequency response is the change in the internal volume of the front cavity 1, resulting in a change in the resonant cavity formed by the front cavity 1 and the output port 2.

[0043] Figure 2 The measurement schematic diagram of the displacement-related frequency response in a micro speaker with a resonant front cavity provided by an embodiment of the present invention is shown in Figure 2 As shown, the vertical axis is the FR magnitude, and the horizontal axis is the cursor 20.0 hz, 102.85 db, and the original (damped) effect of bias.

[0044] The measurement results show that there is strong intermodulation distortion around the resonant frequency of the front cavity 1. Figure 3 The schematic diagram of the two-tone intermodulation spectrum provided by an embodiment of the present invention. Figure 3The ordinate is the signal level, which can also be called the signal strength, with the unit of dB, and the abscissa is the frequency with the unit of Hz. The double-tone intermodulation distortion and response variation of the speaker can be as Figure 2 shown. Figure 4 This is a schematic diagram of the multi-frequency sound spectrum provided by an embodiment of the present invention. Figure 4 The ordinate is the signal level with the unit of dB, and the abscissa is the frequency with the unit of Hz. The multi-tone distortion of the speaker and its response variation can be as Figure 2 shown. There are other mechanisms that may cause response variations, especially in larger speakers (i.e., not micro speakers), which can be solved by a signal processing system similar to the method described in the embodiments of the present invention. In a larger system, an important mechanism is the change of the voice coil inductance with the movement of the diaphragm, which is generally considered one of the main sources of midrange and high-frequency intermodulation distortion in a wide frequency range, and the change of the cavity resonance caused by the air space behind the diaphragm and the holes on the speaker chassis, which generally affects a narrower frequency range. There are also other mechanisms that can cause displacement-related response variations, so the scope of the embodiments of the present invention is not limited to compensating for only the distortions related to the front cavity.

[0045] To solve the technical problems proposed in the related art, an embodiment of the present invention provides a signal processing system that can correct the displacement-related frequency response variations of the driver in a very wide frequency range, thereby reducing distortion, especially intermodulation distortion.

[0046] Figure 5 This is a schematic structural diagram of a signal processing system provided by an embodiment of the present invention, as Figure 5 shown. 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 respectively connected to the group delay compensation module 15 and the displacement model 12. 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 the embodiment of the present invention, the filter coefficient generation module 14 is used to obtain the filter coefficients through an interpolation lookup table, an interpolation function, or machine learning.

[0049] In an embodiment of the present invention, 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 an embodiment of the present invention, the audio converter 18 includes a speaker, headphones, electroacoustic or electromechanical transducers. Among them, the frequency response of the signal processing system can display frequency response variations related to the temporary values of the system state parameters, such as transducer displacement, the velocity of parts of the acoustic system, or the internal temperature.

[0051] In an embodiment of the present invention, the displacement model 12 includes a computational implementation model of the diaphragm displacement of the speaker.

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

[0053] In an embodiment of the present invention, the purpose of the filter is to balance the difference between the target frequency response at the static position of the diaphragm and the actual frequency response measured or calculated at the static displacement diaphragm position.

[0054] In an embodiment of the present invention, in some cases, the flow velocity caused by the low-frequency signal in the port will have a significant impact on the loss in the port. In this case, the adjustment of the filter parameters can benefit from using the port velocity as a control parameter. Figure 6 It is a schematic structural diagram of another signal processing system provided by an embodiment of the present invention, as Figure 6 shown, 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 respectively connected to the group delay compensation module 15 and the displacement model 12, 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, and the filter coefficient generation module 14 is connected to the adjustable equalizer 16.

[0056] The system further 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 an embodiment of the present invention, when correcting low-frequency distortion related to a transducer (in this case, "low-frequency" means close to the fundamental resonance of the transducer), the position-dependent signal part is also the signal that needs to be compensated. Therefore, the compensation process needs to depend on both time and non-linearity. On the other hand, in an embodiment of the present invention, the cause of non-linearity is a low-frequency signal, and the signal to be compensated has a higher frequency than the signal component that causes non-linearity, and the signal to be compensated behaves linearly 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 non-linear drive model can be adaptively updated using feedback information, which can improve the compensation accuracy of all systems. The feedback signal can come from speaker current measurement, speaker current and voltage measurement, or as information from a separate feedback sensor.

[0058] Figure 7 FIG. is a schematic structural diagram of another signal processing system provided by an embodiment of the present invention, as Figure 7 shown. The system includes: an input signal module 11, a non-linear time-varying controller 71, a non-linear drive model 72, an amplifier 17, and an audio converter 18.

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

[0060] Figure 8 FIG. is a schematic structural diagram of another signal processing system provided by an embodiment of the present invention, as Figure 8 shown. The system includes: an input signal module 11, a linear time-varying equalizer 81, a linear or non-linear drive model 82, an amplifier 17, and an audio converter 18.

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

[0062] Traditional low-frequency non-linear compensation and voice coil temperature compensation algorithms can be used in combination with the embodiments of the present invention, which may improve the performance of the embodiments of the present invention because the diaphragm displacement can be better predicted. For multiple signal processing systems, the displacement model of the same driver can also be used.

[0063] A time-varying equalizer is usually implemented as a dual set of Infinite Impulse Response (IIR) filters, which can be composed of multiple filter blocks. Figure 9 For Figure 5 or Figure 6 An example of the tunable equalizer 16 in Figure 9 is implemented as a second-order filter block in "direct form". The equalizer can be composed of one or more such second-order filter blocks, or can be composed of one or more simpler first-order filter blocks. Figure 9 The filter coefficient model in

[0064] Figure 10 can be a look-up table, an interpolation function, etc., and the filter coefficients a and b can be updated for each sample. Figure 1 is a schematic diagram of the change of the response with the diaphragm movement provided by the embodiments of the present invention. Figure 10 In Figure 10 the simplified frequency response of the front cavity of the speaker when considering the driver displacement effect is shown. The frequency scale is normalized to the nominal resonance frequency of the front cavity of the speaker, as Figure 10 shown, the abscissa is the normalized frequency, and the ordinate is the response. Figure 1 In

[0065] Figure 11 is a schematic diagram provided by the embodiments of the present invention for equalizing the position-dependent equalizer response in Figure 10 to the nominal target, as Figure 11 shown, the abscissa is the normalized frequency, and the ordinate is the response. Figure 11 It represents the diaphragm position-dependent equalizer response required to correct the upward and downward responses in Figure 10 to make them the same as the nominal response. This equalization can be achieved through, for example, Figure 9 the architecture of

[0066] The simulation effect of an embodiment of the present invention on an actual music signal is as Figure 12 shown.Figure 12 A schematic diagram of the signal spectrum and the distortion spectrum provided by the embodiments of the present invention, as Figure 12 shown, the abscissa is the frequency, and the ordinate is the signal level. Figure 12 The original signal spectrum, the uncorrected distortion spectrum, and the corrected distortion spectrum are shown in total. Figure 12 It represents the simulated distortion improvement. 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 non-linear speaker simulation model without applying the correction algorithm, while the dashed line (corrected distortion spectrum) is the case after applying the correction.

[0067] The displacement model of the driver generates an output signal from the input signal module, and the output signal corresponds to the expected displacement of the driver in the housing of the driver. The precise implementation of this model is not important for the embodiments of the present invention. The simplest model can be a low-pass filter. Further improvements can use a displacement-related non-linear model, as well as a model considering the change of voice coil temperature. These models can be obtained from a computational thermal model or estimated from the measurement of driver voltage and current. The model can also be fully adaptive, based on voltage and current information, or in a larger speaker, feedback sensor information.

[0068] Figure 13 A schematic structural diagram of another signal processing system provided by the embodiments of the present invention, as Figure 13 shown, 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, a tunable 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 respectively connected to the group delay compensation module 15 and the displacement model 12. 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 tunable 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 an adder. The group delay compensation module 15 is connected to the high-pass filter 22. The high-pass filter 22 is connected to the tunable equalizer 16. The tunable equalizer 16 is connected to the adder. The adder is connected to the amplifier 17.

[0070] In the embodiments of the present invention, if the frequencies of the high-frequency resonances to be corrected are separated, and the frequency range of the corresponding equalizer is not large, then the equalizer designed to correct the high-frequency response may have an 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 equalization of the low-frequency, as Figure 13 shown. Separate processing can also be applied to the low-frequency, as Figure 14 shown.

[0071] Figure 14 FIG. is a schematic structural diagram of another signal processing system provided by the embodiments of the present invention, as Figure 14 shown. 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 respectively connected to the group delay compensation module 15 and the displacement model 12. 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 an 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 respectively connected to the adjustable low-frequency equalizer 23 and the adjustable high-frequency equalizer 161.

[0073] In the embodiments of the present invention, all or part of the signal processing system can also use analog filters. The analog implementation of the analog filter includes the control path of the displacement model, which is configured as a low-pass filter (digital or analog), a non-linear function that maps displacement data to filter control parameters (which can be implemented in a fully analog implementation, for example, using an operational amplifier), and a drive signal path that includes an all-pass filter and a voltage-controlled filter. The actual implementation can also include a hybrid of analog and digital functions. For example, an analog voltage-controlled filter (VCF) controlled by a digital displacement model and a digital implementation of a control lookup.

[0074] In an embodiment of the present invention, the application of the algorithm is not limited to the frequency response changes caused by system geometric variations. As an example of other sources of response changes, especially in larger loudspeakers, the mid-frequency / high-frequency response changes caused by the displacement dependence of the voice coil inductance are generally considered the main source of mid-frequency / high-frequency intermodulation distortion in loudspeakers. When the driver is subject to static displacement, the inductance effect can be measured by frequency response changes, and thus the same algorithm can be used for compensation. In addition, when the diaphragm is displaced from the static position, the sensitivity of the driver usually changes, mainly due to the change of the Bl factor, and this effect can be combined into the look-up table. The change of the driving voice coil temperature in a dynamic loudspeaker also causes changes in sensitivity and response shape, which can be used as an additional control parameter.

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

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

[0077] In the technical solution provided by the embodiment of the present invention, there are 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 respectively connected to 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, and the filter coefficient generation module is connected to the adjustable equalizer. In the technical solution provided by the embodiment of the present invention, the frequency response changes related to the driver displacement can be corrected in a wide frequency range, thereby reducing distortion, especially intermodulation distortion.

[0078] In the technical solution provided by the embodiment of the present invention, distortion reduction can be provided while reducing frequency response changes. Alternative methods can also be based on deliberately changing the tonal balance of the signal. The computational complexity of the model is low. The signal processing system can be applied to a wide range of electroacoustic systems, not just miniature loudspeakers.

[0079] An embodiment of the present invention provides an electronic device including the above-mentioned signal processing system.

[0080] The above are only the preferred embodiments of the present invention and are not intended to limit the present invention. Any modifications, equivalent replacements, improvements, etc. made within the spirit and principles of the present invention shall be included within the scope of protection of the present invention.

Claims

1. A signal processing system, characterized in that, 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; The input signal module is respectively connected to 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, and the filter coefficient generation module is connected to the adjustable equalizer.

2. The system according to claim 1, wherein Further includes: A low-pass filter, the displacement model is connected to the low-pass filter, and the low-pass filter is connected to the filter coefficient generation module.

3. The system according to claim 2, wherein The low-pass filter includes a first low-pass filter, and the system further includes: a port speed model and a second low-pass filter, the input signal module is connected to the port speed model, the port speed model is connected to the second low-pass filter, and the second low-pass filter is connected to the filter coefficient generation module.

4. The system according to claim 2, wherein 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; 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.

5. The system according to claim 2, wherein 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; 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 respectively connected to the adjustable low-frequency equalizer and the adjustable high-frequency equalizer.

6. The system according to claim 1, characterized in that, The filter coefficient generation module is used to obtain filter coefficients through an interpolation lookup table, an interpolation function, or machine learning.

7. The system according to claim 2, wherein The group delay compensation module is used to compensate the group delay of the displacement model and the low-pass filter.

8. The system according to claim 1, wherein The audio converter includes a speaker, headphones, an electroacoustic or electromechanical transducer.

9. The system according to claim 1, wherein The displacement model includes a calculation implementation model of the diaphragm displacement of the speaker.

10. An electronic device, characterized in that, Including the signal processing system according to any one of claims 1 to 9.

Images