Signal processing method, device, apparatus, and computer-readable storage medium
By separating the speaker input audio signal into high-frequency and low-frequency signals, adjusting the high-frequency signal gain based on the estimated amplitude, and processing the high-frequency signal voltage to offset nonlinear parameter changes, the speaker intermodulation distortion problem is solved and the purity of the audio signal is improved.
Patent Information
- Application Number
- CN202310945906.X
- Authority / Receiving Office
- CN · China
- Patent Type
- Patents(China)
- Current Assignee / Owner
- Filing Date
- 2023-07-28
- Publication Date
- 2025-09-16
- Estimated Expiration
- 2043-07-28
AI Technical Summary
当至少两种不同频率的音频信号同时输入扬声器时,低频信号导致扬声器的非线性参数变化,引起高频信号的互调失真,影响音频信号的纯净度。
通过获取输入音频信号并将其分为高频信号和低频信号,确定扬声器基于低频信号的估计振幅,根据估计振幅确定高频信号的信号增益,并通过信号增益处理高频信号电压以抵消非线性参数变化,叠加处理后信号以抑制互调失真。
It effectively suppresses intermodulation distortion and improves the purity of the audio signal output by the speaker system.
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Figure CN116896706B_ABST
Abstract
Description
Technical Field
[0001] The present invention relates to the field of loudspeaker technology, and in particular to a signal processing method, apparatus, device, and computer-readable storage medium. Background Art
[0002] When at least two audio signals of different frequencies are simultaneously input into a loudspeaker, the low-frequency signal in the audio signal will cause the loudspeaker to undergo large displacement, causing the speaker's nonlinear parameters, such as the electromechanical coefficient BL(x) and inductance Le(x), to change accordingly. The period of change in these nonlinear parameters coincides with the period of the low-frequency signal. The period of the high-frequency signal in the audio signal is shorter than that of the low-frequency signal. Therefore, the process of the high-frequency signal driving the loudspeaker can be considered a quasi-static process.
[0003] However, due to the nonlinear parameter changes caused by the low-frequency signal, the Ampere driving force of the high-frequency signal changes, causing the high-frequency signal to fluctuate with the same period as the low-frequency signal. That is, the high-frequency signal produces an undesirable periodic envelope. This undesirable periodic envelope occurs in the entire high-frequency band, resulting in the audio signal being rough and having low purity when played. This process is the modulation distortion of the low-frequency signal on the high-frequency signal, also known as intermodulation distortion. Summary of the Invention
[0004] The main purpose of the present invention is to provide a signal processing method, device, equipment and computer-readable storage medium, aiming to suppress intermodulation distortion of audio signals and improve the purity of sound signals.
[0005] To achieve the above object, the present invention provides a signal processing method, which includes the following steps:
[0006] Acquire an input audio signal input to a speaker, and separate the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency;
[0007] determining an estimated amplitude generated by the speaker based on the low-frequency signal, and determining a signal gain of the high-frequency signal based on the estimated amplitude, wherein a gain value of the signal gain is positively correlated with the estimated amplitude;
[0008] The signal voltage of the high-frequency signal is processed by the signal gain to obtain a processed high-frequency signal, and the processed high-frequency signal and the low-frequency signal are superimposed to obtain an audio signal with suppressed intermodulation distortion.
[0009] Optionally, the step of determining the signal gain of the high-frequency signal based on the estimated amplitude includes:
[0010] determining a nonlinear parameter change corresponding to the estimated amplitude based on the estimated amplitude and a nonlinear characteristic curve of the speaker;
[0011] The signal gain of the high-frequency signal is determined based on the nonlinear parameter change amount.
[0012] Optionally, the nonlinear characteristic curve includes a mechanoelectric coefficient curve;
[0013] The step of determining the nonlinear parameter variation corresponding to the estimated amplitude based on the estimated amplitude and the nonlinear characteristic curve of the speaker comprises:
[0014] Determining, based on the estimated amplitude and the electromechanical coefficient curve, an attenuation of the electromechanical coefficient of the voice coil position corresponding to the estimated amplitude relative to the equilibrium position;
[0015] The attenuation of the electromechanical coefficient is used as the nonlinear parameter change corresponding to the estimated amplitude.
[0016] Optionally, the step of determining the estimated amplitude generated by the speaker based on the low-frequency signal comprises:
[0017] Obtaining speaker linear parameters and speaker nonlinear parameters of the speaker;
[0018] An estimated amplitude generated by the speaker based on the low-frequency signal is calculated based on the speaker linear parameters, the speaker nonlinear parameters and a preset state equation.
[0019] Optionally, the step of obtaining speaker linear parameters and speaker nonlinear parameters of the speaker includes:
[0020] Obtain the current and voltage signals of the speaker;
[0021] Performing system identification on the current and voltage signals to obtain identified linear parameters and identified nonlinear parameters;
[0022] The identified linear parameters are used as speaker linear parameters of the speaker, and the identified nonlinear parameters are used as speaker nonlinear parameters.
[0023] Optionally, before the step of separating the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency, the method further comprises:
[0024] Determining a signal frequency response curve of the input audio signal, and determining a reference amplitude based on an amplitude corresponding to a starting frequency point in the signal frequency response curve;
[0025] A reference frequency corresponding to the reference amplitude in the signal frequency response curve is determined, and a preset frequency is determined based on the reference frequency.
[0026] Optionally, the step of determining a preset frequency based on the reference frequency includes:
[0027] If the reference frequency is less than or equal to a preset frequency threshold, the reference frequency is used as the preset frequency;
[0028] If the reference frequency is greater than the frequency threshold, the frequency threshold is used as the preset frequency.
[0029] To achieve the above object, the present invention further provides a signal processing device, comprising:
[0030] An acquisition module, configured to acquire an input audio signal input to the speaker and separate the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency;
[0031] a determination module, configured to determine an estimated amplitude generated by the speaker based on the low-frequency signal, and determine a signal gain of the high-frequency signal based on the estimated amplitude, wherein a gain value of the signal gain is positively correlated with the estimated amplitude;
[0032] The processing module is used to process the signal voltage of the high-frequency signal by the signal gain to obtain a processed high-frequency signal, and to superimpose the processed high-frequency signal and the low-frequency signal to obtain an audio signal with suppressed intermodulation distortion.
[0033] To achieve the above-mentioned purpose, the present invention also provides a signal processing device, which includes: a memory, a processor, and a signal processing program stored in the memory and executable on the processor, wherein the signal processing program implements the steps of the signal processing method described above when executed by the processor.
[0034] In addition, to achieve the above-mentioned purpose, the present invention further proposes a computer-readable storage medium, on which a signal processing program is stored. When the signal processing program is executed by a processor, the steps of the signal processing method described above are implemented.
[0035] In the present invention, an input audio signal of an input speaker is obtained and divided into a high-frequency signal and a low-frequency signal according to a preset frequency; an estimated amplitude generated by the speaker based on the low-frequency signal is determined, and a signal gain of the high-frequency signal is determined based on the estimated amplitude, wherein the gain value of the signal gain is positively correlated with the estimated amplitude; the signal voltage of the high-frequency signal is processed by the signal gain to obtain a processed high-frequency signal, and the processed high-frequency signal and the low-frequency signal are superimposed to obtain an audio signal with suppressed intermodulation distortion.
[0036] The greater the estimated amplitude generated by the speaker based on the low-frequency signal, the greater the change in the nonlinear parameters in the speaker, resulting in a greater change in the Ampere driving force of the high-frequency signal. In the present invention, by setting the gain value of the signal gain to be positively correlated with the estimated amplitude, the greater the estimated amplitude, the greater the signal gain value of the high-frequency signal, that is, the greater the degree of adjustment of the signal voltage of the high-frequency signal, so that the change in the signal voltage of the high-frequency signal can offset the change in the Ampere driving force caused by the nonlinear parameters, so that the Ampere driving force of the high-frequency signal after processing under different estimated amplitudes remains stable, thereby suppressing and eliminating the unexpected envelope generated by the intermodulation of the low-frequency signal to the high-frequency signal, thereby suppressing intermodulation distortion and improving the purity of the audio signal output by the speaker system. BRIEF DESCRIPTION OF THE DRAWINGS
[0037] Figure 1 A schematic diagram of the hardware operating environment involved in an embodiment of the present invention;
[0038] Figure 2 1 is a flow chart of a first embodiment of a signal processing method according to the present invention;
[0039] Figure 3 A nonlinear characteristic curve of BL and displacement related to an embodiment of the signal processing method of the present invention;
[0040] FIG4( a ) is a nonlinear characteristic curve of Le and displacement related to an embodiment of a signal processing method of the present invention;
[0041] FIG4( b ) is a nonlinear characteristic curve of Le and current in accordance with an embodiment of the signal processing method of the present invention;
[0042] Figure 5 A schematic diagram of a system structure involved in an embodiment of a signal processing method of the present invention;
[0043] Figure 6 Schematic diagram of the functional modules of a preferred embodiment of the signal processing device of the present invention.
[0044] The purpose, features and advantages of the present invention will be further described with reference to the accompanying drawings and in conjunction with the embodiments. DETAILED DESCRIPTION
[0045] It should be understood that the specific embodiments described herein are only used to explain the present invention and are not intended to limit the present invention.
[0046] like Figure 1 As shown, Figure 1 It is a schematic diagram of the device structure of the hardware operating environment involved in the embodiment of the present invention.
[0047] It should be noted that the signal processing device in the embodiment of the present invention can be an audio device, such as headphones, smart glasses, head-mounted display devices, smart phones, personal computers and other devices, or it can be a device that establishes a communication connection with the audio device, such as a server, etc., and no specific restrictions are made here.
[0048] like Figure 1 As shown, the signal processing device may include: a processor 1001, such as a CPU, a network interface 1004, a user interface 1003, a memory 1005, and a communication bus 1002. Among them, the communication bus 1002 is used to realize the connection and communication between these components. The user interface 1003 may include a display screen (Display), an input unit such as a keyboard (Keyboard), and the user interface 1003 may also include a standard wired interface and a wireless interface. The network interface 1004 may optionally include a standard wired interface and a wireless interface (such as a WI-FI interface). The memory 1005 may be a high-speed RAM memory or a stable memory (non-volatile memory), such as a disk memory. The memory 1005 may also be a storage device independent of the aforementioned processor 1001.
[0049] Those skilled in the art will understand that Figure 1 The device structure shown in the figure does not constitute a limitation on the signal processing device, and may include more or fewer components than shown in the figure, or combine certain components, or arrange the components differently.
[0050] like Figure 1 As shown, the memory 1005 as a computer storage medium may include an operating system, a network communication module, a user interface module, and a signal processing program. The operating system is a program that manages and controls the hardware and software resources of the device and supports the operation of the signal processing program and other software or programs. Figure 1 In the device shown, the user interface 1003 is mainly used to communicate data with the client; the network interface 1004 is mainly used to establish a communication connection with the server; and the processor 1001 can be used to call the signal processing program stored in the memory 1005 and perform the following operations:
[0051] Acquire an input audio signal input to a speaker, and separate the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency;
[0052] determining an estimated amplitude generated by the speaker based on the low-frequency signal, and determining a signal gain of the high-frequency signal based on the estimated amplitude, wherein a gain value of the signal gain is positively correlated with the estimated amplitude;
[0053] The signal voltage of the high-frequency signal is processed by the signal gain to obtain a processed high-frequency signal, and the processed high-frequency signal and the low-frequency signal are superimposed to obtain an audio signal with suppressed intermodulation distortion.
[0054] Furthermore, the step of determining the signal gain of the high-frequency signal based on the estimated amplitude includes:
[0055] determining a nonlinear parameter change corresponding to the estimated amplitude based on the estimated amplitude and a nonlinear characteristic curve of the speaker;
[0056] The signal gain of the high-frequency signal is determined based on the nonlinear parameter change amount.
[0057] Furthermore, the nonlinear characteristic curve includes a mechanoelectric coefficient curve;
[0058] The step of determining the nonlinear parameter variation corresponding to the estimated amplitude based on the estimated amplitude and the nonlinear characteristic curve of the speaker comprises:
[0059] Determining, based on the estimated amplitude and the electromechanical coefficient curve, an attenuation of the electromechanical coefficient of the voice coil position corresponding to the estimated amplitude relative to the equilibrium position;
[0060] The attenuation of the electromechanical coefficient is used as the nonlinear parameter change corresponding to the estimated amplitude.
[0061] Furthermore, the step of determining the estimated amplitude generated by the speaker based on the low-frequency signal includes:
[0062] Obtaining speaker linear parameters and speaker nonlinear parameters of the speaker;
[0063] An estimated amplitude generated by the speaker based on the low-frequency signal is calculated based on the speaker linear parameters, the speaker nonlinear parameters and a preset state equation.
[0064] Furthermore, the step of obtaining speaker linear parameters and speaker nonlinear parameters of the speaker includes:
[0065] Obtain the current and voltage signals of the speaker;
[0066] Performing system identification on the current and voltage signals to obtain identified linear parameters and identified nonlinear parameters;
[0067] The identified linear parameters are used as speaker linear parameters of the speaker, and the identified nonlinear parameters are used as speaker nonlinear parameters.
[0068] Furthermore, before the step of separating the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency, the processor 1001 may also be configured to call a signal processing program stored in the memory 1005 to perform the following operations:
[0069] Determining a signal frequency response curve of the input audio signal, and determining a reference amplitude based on an amplitude corresponding to a starting frequency point in the signal frequency response curve;
[0070] A reference frequency corresponding to the reference amplitude in the signal frequency response curve is determined, and a preset frequency is determined based on the reference frequency.
[0071] Furthermore, the step of determining the preset frequency based on the reference frequency includes:
[0072] If the reference frequency is less than or equal to a preset frequency threshold, the reference frequency is used as the preset frequency;
[0073] If the reference frequency is greater than the frequency threshold, the frequency threshold is used as the preset frequency.
[0074] Based on the above structure, various embodiments of a signal processing method are proposed.
[0075] Reference Figure 2 , Figure 2 FIG. 1 is a flow chart of a first embodiment of a signal processing method according to the present invention.
[0076] The embodiments of the present invention provide embodiments of the signal processing method. It should be noted that although the logical order is shown in the flowchart, in some cases, the steps shown or described may be performed in an order different from that shown here. In this embodiment, the execution subject of the signal processing method can be an audio device, such as headphones, smart glasses, head-mounted display devices, smart phones, and personal computers, or a device that establishes a communication connection with the audio device, such as a server, etc., which is not limited in this embodiment. For the sake of ease of description, the execution subject is omitted for elaboration of each embodiment. In this embodiment, the signal processing method includes:
[0077] Step S10, obtaining an input audio signal input to a speaker, and dividing the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency;
[0078] In this embodiment, an audio signal input to a speaker is obtained, which is hereinafter referred to as an input audio signal for distinction.
[0079] The input audio signal is divided into a high-frequency signal and a low-frequency signal according to a preset frequency, wherein the preset frequency can be set according to the acoustic model of the speaker or the amplitude curve of the input audio signal, or can be set according to actual needs, and is not limited here.
[0080] The speaker amplitude corresponding to low-frequency signals is larger, while the speaker amplitude corresponding to high-frequency signals is smaller. Intermodulation distortion is actually the modulation of the high-frequency signal by the large amplitude of the low-frequency signal, causing undesirable signal fluctuations in the high-frequency signal and resulting in signal distortion. Therefore, when dividing the high-frequency and low-frequency signals in this embodiment, it is necessary to divide the high-amplitude signal into the low-frequency signal to effectively suppress intermodulation distortion.
[0081] Step S20, determining an estimated amplitude generated by the speaker based on the low-frequency signal, and determining a signal gain of the high-frequency signal based on the estimated amplitude, wherein a gain value of the signal gain is positively correlated with the estimated amplitude;
[0082] In this embodiment, the amplitude of the speaker generated based on the low-frequency signal (hereinafter referred to as the estimated amplitude for clarity) is predicted, that is, the maximum displacement generated by the speaker based on the low-frequency signal. This embodiment does not limit the method for predicting the estimated amplitude. For example, in one feasible implementation, it can be obtained based on a linear displacement prediction method in a nonlinear compensation algorithm; in another feasible implementation, it can also be calculated based on existing methods, which is not limited here.
[0083] In this embodiment, the signal gain of the high-frequency signal is determined based on the estimated amplitude, and the signal gain is used to compensate the voltage signal of the high-frequency signal to offset the change in the Ampere force of the high-frequency signal caused by the low-frequency signal. Since the larger the estimated amplitude generated by the speaker based on the low-frequency signal, the larger the change in the nonlinear parameters in the speaker, resulting in a larger change in the Ampere driving force of the high-frequency signal, the estimated amplitude is set to be positively correlated with the signal gain, so that the change in the signal voltage of the high-frequency signal can offset the change in the Ampere driving force caused by the nonlinear parameters, so that the Ampere driving force of the high-frequency signal after processing under different estimated amplitudes remains stable.
[0084] Specifically, in one feasible implementation, the parameter change of the nonlinear characteristic parameter of the loudspeaker can be determined based on the estimated amplitude, and the signal gain can be determined based on the parameter change; in another feasible implementation, the correspondence between the amplitude and the gain can be pre-set, and the gain corresponding to the estimated amplitude can be determined as the signal gain based on the correspondence; the signal gain can also be determined by other feasible methods, which are not limited here.
[0085] It should be noted that signal gain can be positive or negative, and the sign of the signal gain is related to the Ampere driving force generated by the high-frequency signal based on the voice coil displacement corresponding to the estimated amplitude. If the Ampere driving force generated by the high-frequency signal based on the voice coil displacement corresponding to the estimated amplitude increases, the signal gain becomes negative to offset the increase in Ampere driving force. If the Ampere driving force generated by the high-frequency signal based on the voice coil displacement corresponding to the estimated amplitude decreases, the signal gain becomes positive to offset the decrease in Ampere driving force, thereby maintaining a stable Ampere driving force.
[0086] Step S30 , processing the signal voltage of the high-frequency signal by the signal gain to obtain a processed high-frequency signal, and superimposing the processed high-frequency signal and the low-frequency signal to obtain an audio signal with suppressed intermodulation distortion.
[0087] In this embodiment, the signal voltage of the high-frequency signal is processed using signal gain to obtain a processed high-frequency signal. Based on the Ampere force calculation formula: F = iBL, it can be seen that the Ampere driving force of the high-frequency signal is affected by the current i and the magnetoelectric coefficient BL, where the magnetoelectric coefficient BL is determined by the estimated amplitude of the low-frequency signal. Adjusting the signal voltage can adjust the current, thereby adjusting the Ampere driving force of the high-frequency signal. In this embodiment, the processing direction of the high-frequency signal is determined by the positive or negative signal gain. When the signal gain is positive, the high-frequency signal is gain-processed, that is, amplified; when the signal gain is negative, the high-frequency signal is attenuated.
[0088] After superposition processing, the high-frequency signal and the low-frequency signal are combined to obtain an audio signal with suppressed intermodulation distortion.
[0089] Furthermore, in a feasible implementation manner, before step S10, the following is further included:
[0090] Step S40, determining a signal frequency response curve of the input audio signal, and determining a reference amplitude based on an amplitude corresponding to a starting frequency point in the signal frequency response curve;
[0091] In this embodiment, the signal frequency response curve of the input audio signal is determined, and a reference amplitude is determined based on the amplitude corresponding to the starting frequency point in the signal frequency response curve. In one embodiment, a range of values for the reference amplitude can be determined based on the amplitude corresponding to the starting frequency point, and the reference amplitude is determined from this range of values. In another embodiment, a correspondence between different amplitudes and the reference amplitude can be pre-set, and the reference amplitude corresponding to the amplitude of the starting frequency point can be determined from this correspondence, without limitation. For example, in one embodiment, the reference amplitude can be determined within the range of [X-6dB, X+6dB], where X is the amplitude corresponding to the starting frequency point.
[0092] Step S50 , determining a reference frequency corresponding to the reference amplitude in the signal frequency response curve, and determining a preset frequency based on the reference frequency.
[0093] A reference frequency corresponding to the reference amplitude in the signal frequency response curve is determined, and a preset frequency is determined based on the reference frequency. In one embodiment, the reference frequency may be used as the preset frequency; in another embodiment, the reference frequency may be processed and the processed reference frequency may be used as the preset frequency.
[0094] In this embodiment, by determining the reference frequency based on the amplitude of its actual frequency point and determining the preset frequency based on the reference frequency, this embodiment can distinguish high-frequency signals and low-frequency signals according to the amplitude, so that large-amplitude signals are divided into low-frequency signals and small-amplitude signals are divided into high-frequency signals, thereby improving the suppression effect of intermodulation distortion.
[0095] Furthermore, in a feasible implementation manner, the step S50 includes:
[0096] Step S501: If the reference frequency is less than or equal to a preset frequency threshold, the reference frequency is used as a preset frequency;
[0097] In this embodiment, a frequency threshold is pre-set, and the preset threshold is determined based on the frequency threshold. The frequency threshold can be set according to actual needs. For example, in one feasible embodiment, the frequency threshold can be set to 1 kHz. Based on the auditory characteristics of the human ear at different frequencies, the sound pressure level corresponding to 1 kHz is just the sound pressure level that the human ear can just hear. Therefore, using 1 kHz as the frequency threshold can make the preset frequency more accurate, thereby making the obtained high-frequency and low-frequency signals more accurate.
[0098] Specifically, it is detected whether the reference frequency is less than a frequency threshold. If the reference frequency is less than or equal to a preset frequency threshold, it is considered that the low-frequency signal is accurately divided according to the reference frequency, that is, the low-frequency signal does not contain a small-amplitude high-frequency signal, and the reference frequency is used as the preset frequency.
[0099] Step S502: If the reference frequency is greater than the frequency threshold, the frequency threshold is used as the preset frequency.
[0100] If the reference frequency is greater than the frequency threshold, it is considered that the low-frequency signal divided according to the reference frequency may include a high-frequency signal. In order to ensure the accuracy of the low-frequency signal and the effect of suppressing intermodulation distortion, the frequency threshold is used as the preset frequency.
[0101] In this embodiment, an input audio signal of an input speaker is obtained and divided into a high-frequency signal and a low-frequency signal according to a preset frequency; an estimated amplitude generated by the speaker based on the low-frequency signal is determined, and a signal gain of the high-frequency signal is determined based on the estimated amplitude, wherein the gain value of the signal gain is positively correlated with the estimated amplitude; the signal voltage of the high-frequency signal is processed by the signal gain to obtain a processed high-frequency signal, and the processed high-frequency signal and the low-frequency signal are superimposed to obtain an audio signal with suppressed intermodulation distortion.
[0102] The greater the estimated amplitude generated by the speaker based on the low-frequency signal, the greater the change in the nonlinear parameters in the speaker, resulting in a greater change in the Ampere driving force of the high-frequency signal. In this embodiment, by setting the gain value of the signal gain to be positively correlated with the estimated amplitude, the greater the estimated amplitude, the greater the signal gain value of the high-frequency signal, that is, the greater the degree of adjustment of the signal voltage of the high-frequency signal, so that the change in the signal voltage of the high-frequency signal can offset the change in the Ampere driving force caused by the nonlinear parameters, so that the Ampere driving force of the high-frequency signal after processing under different estimated amplitudes remains stable, thereby suppressing the unexpected envelope caused by the intermodulation of the low-frequency signal to the high-frequency signal, thereby suppressing intermodulation distortion and improving the purity of the audio signal output by the speaker system.
[0103] Furthermore, based on the above-mentioned first embodiment, a second embodiment of the signal processing method of the present invention is proposed. In this embodiment, step S20 includes:
[0104] Step S201, determining a nonlinear parameter variation corresponding to the estimated amplitude based on the estimated amplitude and a nonlinear characteristic curve of the speaker;
[0105] In this embodiment, the signal gain is determined based on the estimated amplitude and the nonlinear characteristic curve of the speaker to improve the accuracy of the signal gain.
[0106] Specifically, the nonlinear parameter change corresponding to the estimated amplitude is determined based on the estimated amplitude and the nonlinear characteristic curve of the speaker. Intermodulation distortion is primarily affected by the speaker's nonlinear characteristics: electromechanical coefficient BL(x) and inductance Le(x). Therefore, the nonlinear characteristic curve in this embodiment can be a BL(x) curve and / or a Le(x) curve, without limitation.
[0107] A nonlinear parameter change corresponding to the estimated amplitude is determined from the nonlinear characteristic curve, wherein the nonlinear parameter change refers to a change in a nonlinear characteristic parameter that affects intermodulation distortion, and the change is relative to the equilibrium position of the loudspeaker voice coil, that is, the position where the displacement is zero.
[0108] In this embodiment, the process of determining the nonlinear parameter change corresponding to the estimated amplitude can be: determining the nonlinear parameter value corresponding to the estimated amplitude from the nonlinear characteristic curve (hereinafter referred to as the target parameter value for distinction); determining the nonlinear parameter value corresponding to the equilibrium position of the voice coil from the nonlinear characteristic curve (hereinafter referred to as the reference parameter value for distinction); and calculating the nonlinear parameter change by subtracting the reference parameter value from the target parameter value.
[0109] Step S202: determining the signal gain of the high-frequency signal based on the nonlinear parameter variation.
[0110] In this embodiment, the signal gain of the high-frequency signal is determined based on the variation of the nonlinear parameter.
[0111] Specifically, in one feasible implementation, the correspondence between different parameter changes and gains may be pre-set, and the gain corresponding to the nonlinear parameter change may be determined from the correspondence as the signal gain of the high-frequency signal; in another feasible implementation, the rate of change of the target parameter value relative to the reference parameter value may be calculated based on the nonlinear parameter change, and the signal gain rate of the high-frequency signal may be determined based on the rate of change, and the signal gain may be calculated based on the signal gain rate.
[0112] Furthermore, in a feasible implementation, after obtaining the high-frequency signal, the signal gain can be adjusted based on characteristics such as the acoustic model of the speaker, so that the adjusted signal gain is more consistent with the actual structure of the speaker, and the high-frequency signal is processed by the adjusted signal gain to improve the effect of suppressing intermodulation distortion.
[0113] Furthermore, in a feasible implementation manner, the nonlinear characteristic curve includes a mechanoelectric coefficient curve, and step S201 includes:
[0114] Step S2011, determining an attenuation of the electromechanical coefficient of the voice coil position corresponding to the estimated amplitude relative to the equilibrium position based on the estimated amplitude and the electromechanical coefficient curve;
[0115] In this embodiment, referring to Figure 3 , Figure 3 The horizontal axis X is the voice coil displacement, the absolute value of the voice coil displacement is the estimated amplitude, and the vertical axis is the BL electromechanical coefficient. The solid line curve is when the height of the voice coil is greater than the air gap depth (i.e. Figure 3 The BL variation curve for the long voice coil with overhang shown in FIG, the dotted curve is the height of the voice coil equal to the air gap depth (ie Figure 3 The BL variation curve under the condition of equal-length shown in . Figure 3It can be seen that no matter the voice coil displacement is positive or negative, the electromechanical coefficient decreases continuously as the voice coil displacement value increases. That is, the electromechanical coefficient decays constantly as the estimated amplitude increases.
[0116] Therefore, in this embodiment, based on the estimated amplitude and the electromechanical coefficient curve, the electromechanical coefficient attenuation of the voice coil position corresponding to the estimated amplitude relative to the equilibrium position is determined.
[0117] Step S2012: Using the attenuation of the electromechanical coefficient as the nonlinear parameter variation corresponding to the estimated amplitude.
[0118] The attenuation of the electromechanical coefficient is used as the nonlinear parameter change corresponding to the estimated amplitude. It should be noted that when the nonlinear characteristic curve includes the electromechanical coefficient curve, since the electromechanical coefficient is constantly attenuated as the estimated amplitude increases, the signal gain determined at this time should be constantly positive. That is, when considering the influence of the electromechanical coefficient on intermodulation distortion, the adjustment of the high-frequency signal should be gain adjustment.
[0119] Furthermore, in one feasible embodiment, the nonlinear characteristic curve may also include an inductance curve. In this embodiment, the specific process of determining the nonlinear parameter change may be: determining a target inductance value corresponding to the voice coil displacement corresponding to the estimated amplitude based on the estimated amplitude and the inductance-displacement curve; determining a target current value corresponding to the target inductance value based on the target inductance value and the inductance-current curve; calculating the target current value minus the current value at the equilibrium position to obtain the inductance change; and using the inductance change as the nonlinear parameter change. In this embodiment, referring to FIG. 4 , the dashed curve in FIG. 4( a ) represents the relationship curve between Le and X-displacement with shorting rings installed (i.e., with shorting rings as shown in FIG. 4( a )), and the solid curve in FIG. 4( a ) represents the relationship curve between Le and X-displacement without shorting rings installed (i.e., without shorting rings as shown in FIG. 4( a )). The dashed curve in Figure 4(a) shows the relationship between Le and X-displacement with shorting rings installed (i.e., with shorting rings, as shown in Figure 4(a)). The solid curve in Figure 4(a) shows the relationship between Le and X-displacement without shorting rings installed (i.e., without shorting rings, as shown in Figure 4(a)). As shown in Figures 4(a) and 4(b), changes in inductance can be positive or negative. Therefore, when considering the impact of inductance on intermodulation distortion, adjustments to high-frequency signals can be either gain or attenuation adjustments.
[0120] Furthermore, in a feasible implementation, the nonlinear parameter change can be determined by combining the electromechanical coefficient and the inductance. For example, the product of the electromechanical coefficient attenuation and the inductance change can be used as the nonlinear parameter change. The specific setting can be made according to actual needs and is not limited here.
[0121] Furthermore, in one feasible implementation, step S20 includes:
[0122] Step S203, obtaining speaker linear parameters and speaker nonlinear parameters of the speaker;
[0123] In this embodiment, linear parameters (hereinafter referred to as speaker linear parameters for distinction) and nonlinear parameters (hereinafter referred to as speaker nonlinear parameters for distinction) of the speaker are obtained.
[0124] The linear parameters of the loudspeaker are the TS signal parameters, which can specifically include parameters such as the voice coil DC resistance Re, the magnetic flux density B in the magnetic gap, the length L of the voice coil wire in the magnetic field, the effective projected area Sd (=πa2) of the diaphragm, as well as Pe(max), the maximum power rating determined by the heat dissipation capacity of the loudspeaker unit, and Vd (=SdXmax), the volume pushed by the diaphragm at maximum amplitude, where Sd is the diaphragm area and Xmax is the maximum amplitude. The nonlinear parameters of the loudspeaker can include parameters such as BL(x), Kms(x), Rms(v), and Le(x). Specifically, the linear parameters and nonlinear parameters of the loudspeaker can be parameters preset in the loudspeaker obtained during the R&D and testing phase. The preset parameters can reduce the amount of calculation and the delay in suppressing intermodulation distortion. They can also be parameters that are updated in real time based on the current and voltage signals generated by the loudspeaker. There is no limitation here.
[0125] Step S204 : obtaining an estimated amplitude generated by the speaker based on the low-frequency signal by calculation based on the speaker linear parameters, the speaker nonlinear parameters, and a preset state equation.
[0126] The estimated amplitude generated by the loudspeaker based on the low-frequency signal is calculated based on the loudspeaker linear parameters, the loudspeaker nonlinear parameters and a preset state equation.
[0127] Among them, the preset state equation is:
[0128]
[0129] where h l(t) is the system response under the loudspeaker linear model, which can be calculated through the loudspeaker linear parameters; U(t) is the input voltage, that is, the voltage of the low-frequency signal obtained after frequency division; α(x) and β(x) are the nonlinear state vectors of the loudspeaker, which can be calculated through the nonlinear parameters of the loudspeaker, such as BL(x), Kms(x), Rms(v) and Le(x).
[0130] Furthermore, in a feasible implementation manner, step S203 includes:
[0131] Step S2031, obtaining the current and voltage signals of the speaker;
[0132] Because loudspeaker system parameters change during operation (especially at extremes), for example, when operating at high voltages, the speaker's temperature rises, causing changes in resistance and softening of the surround, leading to changes in the TS parameter. Such changes can render pre-tested system parameters inaccurate, leading to significant errors in amplitude prediction and reduced high-frequency gain quality. Therefore, in this embodiment, by measuring the current and voltage signals across the loudspeaker in real time, the loudspeaker system parameters are optimized and updated using a loudspeaker model to eliminate errors introduced by extreme speaker operation.
[0133] In this embodiment, the current and voltage signals of the speaker are obtained.
[0134] Step S2032, performing system identification on the current and voltage signals to obtain identified linear parameters and identified nonlinear parameters;
[0135] System identification is performed on the current and voltage signals to obtain identified linear parameters and identified nonlinear parameters. The specific system identification principle is to fit the electrical parameters of the speaker through the current and voltage signals, and update the parameters to minimize the error. It usually includes the following functional modules: a series resistor, which calculates the current in the circuit by measuring the voltage across the resistor to obtain the current and voltage signals; a pilot tone: which adds an extremely low-frequency small signal component to the signal; an error calculation module: which is used to calculate the error between the predicted current and voltage signals and the measured current and voltage signals; and an optimization module: which is used to optimize the speaker system parameters to minimize the above error.
[0136] Step S2033: Using the identified linear parameters as the speaker linear parameters of the speaker, and using the identified nonlinear parameters as the speaker nonlinear parameters.
[0137] The identified linear parameters are used as the speaker linear parameters, and the identified nonlinear parameters are used as the speaker nonlinear parameters. Compared to using predicted linear and nonlinear parameters, this embodiment can make the linear and nonlinear parameters consistent with the actual temperature and humidity, thereby improving the accuracy of the estimated amplitude.
[0138] In this embodiment, the nonlinear parameter change corresponding to the estimated amplitude is determined based on the estimated amplitude and the speaker's nonlinear characteristic curve; the signal gain of the high-frequency signal is then determined based on the nonlinear parameter change. This embodiment improves the accuracy of the signal gain, thereby enhancing the intermodulation distortion suppression effect and improving the purity of the audio signal played by the speaker.
[0139] Further, in one possible implementation, referring to Figure 5 , the speaker in this embodiment is a micro speaker, and the signal processing process can be:
[0140] Perform high and low frequency division on the input audio signal.
[0141] The amplitude of the low-frequency signal is predicted based on the speaker linear parameters (ie, TS parameters) and the speaker nonlinear parameters (ie, nonlinear parameters) to obtain an estimated amplitude. Specifically, the speaker linear parameters and the speaker nonlinear parameters are obtained by system identification based on the current and voltage signals.
[0142] For microspeakers, intermodulation distortion is primarily caused by magnetic field nonlinearity BL(x) due to the small inductance Le(x). Therefore, in this embodiment, the signal gain for high-frequency signal processing is determined based on the estimated amplitude and BL(x). The high-frequency signal is then processed using the signal gain to produce a processed high-frequency signal.
[0143] After superposition processing, the high-frequency signal and the low-frequency signal are combined to obtain an audio signal with suppressed intermodulation distortion.
[0144] In addition, the embodiment of the present invention also provides a signal processing device, referring to Figure 6 , the signal processing device includes:
[0145] An acquisition module 10 is configured to acquire an input audio signal input to a speaker and separate the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency;
[0146] a determination module 20, configured to determine an estimated amplitude generated by the speaker based on the low-frequency signal, and determine a signal gain of the high-frequency signal based on the estimated amplitude, wherein a gain value of the signal gain is positively correlated with the estimated amplitude;
[0147] The processing module 30 is configured to process the signal voltage of the high-frequency signal by using the signal gain to obtain a processed high-frequency signal, and to superimpose the processed high-frequency signal and the low-frequency signal to obtain an audio signal with suppressed intermodulation distortion.
[0148] Furthermore, the determining module 20 is further configured to:
[0149] determining a nonlinear parameter change corresponding to the estimated amplitude based on the estimated amplitude and a nonlinear characteristic curve of the speaker;
[0150] The signal gain of the high-frequency signal is determined based on the nonlinear parameter change amount.
[0151] Furthermore, the nonlinear characteristic curve includes a mechanoelectric coefficient curve; the determination module 20 is further configured to:
[0152] Determining, based on the estimated amplitude and the electromechanical coefficient curve, an attenuation of the electromechanical coefficient of the voice coil position corresponding to the estimated amplitude relative to the equilibrium position;
[0153] The attenuation of the electromechanical coefficient is used as the nonlinear parameter change corresponding to the estimated amplitude.
[0154] Furthermore, the acquisition module 10 is further configured to:
[0155] Obtaining speaker linear parameters and speaker nonlinear parameters of the speaker;
[0156] An estimated amplitude generated by the speaker based on the low-frequency signal is calculated based on the speaker linear parameters, the speaker nonlinear parameters and a preset state equation.
[0157] Furthermore, the acquisition module 10 is further configured to:
[0158] Obtain the current and voltage signals of the speaker;
[0159] Performing system identification on the current and voltage signals to obtain identified linear parameters and identified nonlinear parameters;
[0160] The identified linear parameters are used as speaker linear parameters of the speaker, and the identified nonlinear parameters are used as speaker nonlinear parameters.
[0161] Furthermore, the determining module 20 is further configured to:
[0162] Determining a signal frequency response curve of the input audio signal, and determining a reference amplitude based on an amplitude corresponding to a starting frequency point in the signal frequency response curve;
[0163] A reference frequency corresponding to the reference amplitude in the signal frequency response curve is determined, and a preset frequency is determined based on the reference frequency.
[0164] Furthermore, the determining module 20 is further configured to:
[0165] If the reference frequency is less than or equal to a preset frequency threshold, the reference frequency is used as the preset frequency;
[0166] If the reference frequency is greater than the frequency threshold, the frequency threshold is used as the preset frequency.
[0167] The various embodiments of the signal processing device of the present invention may refer to the various embodiments of the signal processing method of the present invention, and will not be described in detail here.
[0168] In addition, an embodiment of the present invention further provides a computer-readable storage medium, on which a signal processing program is stored. When the signal processing program is executed by a processor, the steps of the signal processing method described below are implemented.
[0169] The various embodiments of the signal processing device and the computer-readable storage medium of the present invention may refer to the various embodiments of the signal processing method of the present invention, and will not be described in detail here.
[0170] It should be noted that, in this document, the terms "comprises," "includes," or any other variations thereof are intended to encompass non-exclusive inclusion, such that a process, method, article, or apparatus comprising a series of elements includes not only those elements but also other elements not explicitly listed, or elements inherent to such process, method, article, or apparatus. In the absence of further limitations, an element defined by the phrase "comprising a ..." does not exclude the presence of other identical elements in the process, method, article, or apparatus comprising the element.
[0171] The serial numbers of the above embodiments of the present invention are for description only and do not represent the advantages or disadvantages of the embodiments.
[0172] Through the description of the above embodiments, those skilled in the art can clearly understand that the above-mentioned embodiment methods can be implemented by means of software plus the necessary general hardware platform, and of course can also be implemented by hardware, but in many cases the former is a better embodiment. Based on this understanding, the technical solution of the present invention, or the part that contributes to the prior art, can be embodied in the form of a software product, which is stored in a storage medium (such as ROM / RAM, magnetic disk, optical disk), and includes a number of instructions for enabling a terminal device (which can be a mobile phone, computer, server, air conditioner, or network device, etc.) to execute the methods described in each embodiment of the present invention.
[0173] The above are only preferred embodiments of the present invention and are not intended to limit the patent scope of the present invention. Any equivalent structure or equivalent process transformation made using the contents of the present invention description and drawings, or directly or indirectly applied in other related technical fields, are also included in the patent protection scope of the present invention.
Claims
1. A signal processing method, characterized in that: The signal processing method comprises the following steps: Acquire an input audio signal input to a speaker, and separate the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency; determining an estimated amplitude generated by the speaker based on the low-frequency signal, and determining a signal gain of the high-frequency signal based on the estimated amplitude, wherein a gain value of the signal gain is positively correlated with the estimated amplitude; The signal voltage of the high-frequency signal is processed by the signal gain to obtain a processed high-frequency signal, and the processed high-frequency signal and the low-frequency signal are superimposed to obtain an audio signal with suppressed intermodulation distortion.
2. The signal processing method according to claim 1, wherein: The step of determining the signal gain of the high-frequency signal based on the estimated amplitude comprises: determining a nonlinear parameter change corresponding to the estimated amplitude based on the estimated amplitude and a nonlinear characteristic curve of the speaker; The signal gain of the high-frequency signal is determined based on the nonlinear parameter change amount.
3. The signal processing method according to claim 2, wherein: The nonlinear characteristic curve includes a mechanoelectric coefficient curve; The step of determining the nonlinear parameter variation corresponding to the estimated amplitude based on the estimated amplitude and the nonlinear characteristic curve of the speaker comprises: determining, based on the estimated amplitude and the electromechanical coefficient curve, an attenuation of the electromechanical coefficient of the voice coil position corresponding to the estimated amplitude relative to the equilibrium position; The attenuation of the electromechanical coefficient is used as the nonlinear parameter change corresponding to the estimated amplitude.
4. The signal processing method according to claim 1, wherein: The step of determining an estimated amplitude of the loudspeaker generated based on the low-frequency signal comprises: Obtaining speaker linear parameters and speaker nonlinear parameters of the speaker; An estimated amplitude generated by the speaker based on the low-frequency signal is obtained by calculation based on the speaker linear parameters, the speaker nonlinear parameters and a preset state equation.
5. The signal processing method according to claim 4, wherein: The step of obtaining the speaker linear parameters and the speaker nonlinear parameters of the speaker includes: Obtain the current and voltage signals of the speaker; Performing system identification on the current and voltage signals to obtain identified linear parameters and identified nonlinear parameters; The identified linear parameters are used as speaker linear parameters of the speaker, and the identified nonlinear parameters are used as speaker nonlinear parameters.
6. The signal processing method according to any one of claims 1 to 5, characterized in that: Before the step of separating the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency, the method further includes: Determining a signal frequency response curve of the input audio signal, and determining a reference amplitude based on an amplitude corresponding to a starting frequency point in the signal frequency response curve; A reference frequency corresponding to the reference amplitude in the signal frequency response curve is determined, and a preset frequency is determined based on the reference frequency.
7. The signal processing method according to claim 6, wherein: The step of determining the preset frequency based on the reference frequency includes: If the reference frequency is less than or equal to a preset frequency threshold, the reference frequency is used as the preset frequency; If the reference frequency is greater than the frequency threshold, the frequency threshold is used as the preset frequency.
8. A signal processing device, characterized in that: The signal processing device includes: An acquisition module, configured to acquire an input audio signal input to the speaker and separate the input audio signal into a high-frequency signal and a low-frequency signal according to a preset frequency; a determination module, configured to determine an estimated amplitude generated by the speaker based on the low-frequency signal, and determine a signal gain of the high-frequency signal based on the estimated amplitude, wherein a gain value of the signal gain is positively correlated with the estimated amplitude; The processing module is used to process the signal voltage of the high-frequency signal by the signal gain to obtain a processed high-frequency signal, and superimpose the processed high-frequency signal and the low-frequency signal to obtain an audio signal with suppressed intermodulation distortion.
9. A signal processing device, characterized in that The signal processing device includes: a memory, a processor, and a signal processing program stored in the memory and executable on the processor. When the signal processing program is executed by the processor, the steps of the signal processing method according to any one of claims 1 to 7 are implemented.
10. A computer-readable storage medium, characterized in that The computer-readable storage medium stores a signal processing program, which, when executed by a processor, implements the steps of the signal processing method according to any one of claims 1 to 7.
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