WO2021248520A1 - 扬声器的非线性补偿方法、扬声器设备、装置和存储介质 - Google Patents

扬声器的非线性补偿方法、扬声器设备、装置和存储介质 Download PDF

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WO2021248520A1
WO2021248520A1 PCT/CN2020/096683 CN2020096683W WO2021248520A1 WO 2021248520 A1 WO2021248520 A1 WO 2021248520A1 CN 2020096683 W CN2020096683 W CN 2020096683W WO 2021248520 A1 WO2021248520 A1 WO 2021248520A1
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speaker
signal
linear
loudspeaker
nonlinear
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PCT/CN2020/096683
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English (en)
French (fr)
Inventor
黄远芳
蓝睿智
吴锐兴
叶利剑
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瑞声声学科技(深圳)有限公司
瑞声科技(新加坡)有限公司
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Publication of WO2021248520A1 publication Critical patent/WO2021248520A1/zh

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • H04R3/04Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R9/00Transducers of moving-coil, moving-strip, or moving-wire type
    • H04R9/02Details
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R9/00Transducers of moving-coil, moving-strip, or moving-wire type
    • H04R9/06Loudspeakers
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2203/00Details of circuits for transducers, loudspeakers or microphones covered by H04R3/00 but not provided for in any of its subgroups
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2400/00Loudspeakers
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00Signal processing covered by H04R, not provided for in its groups

Definitions

  • the present invention relates to the technical field of intelligent power amplifiers, and in particular to a method for non-linear compensation of a speaker, a speaker device, a device and a storage medium.
  • a speaker nonlinear compensation method which is applied to a speaker device including a smart power amplifier.
  • the speaker nonlinear compensation method includes: attenuating the original input signal at the current moment to generate and process the input signal; acquiring the speaker state at the current moment Parameters and speaker system parameters, calculate the predistortion signal corresponding to the processed input signal according to the speaker state parameters and the speaker system parameters; obtain the system response parameters and frequency domain response parameters of the smart power amplifier, and according to the system The response parameter filters the predistortion signal to obtain a compensation signal; input the compensation signal into the smart power amplifier, obtain a playback signal, and input the playback signal into a speaker.
  • a speaker device for a speaker device including a smart power amplifier including: an offset processing module, used to attenuate the original input signal at the current moment, to generate and process the input signal; a non-linear module, used to obtain the speaker at the current moment State parameters and speaker system parameters, calculating a predistortion signal corresponding to the processed input signal according to the speaker state parameters and the speaker system parameters; a power amplifier equalization module for obtaining the system response parameters and frequency domain of the smart power amplifier Response parameter, filtering the predistortion signal according to the system response parameter to obtain a compensation signal; a smart power amplifier module for inputting the compensation signal into the smart power amplifier, obtaining a playback signal, and inputting the playback signal into a speaker .
  • a speaker device includes a processor and a memory, the processor is coupled to the memory, and a computer program is stored in the memory, and the processor executes the computer program to implement the method as described above.
  • a readable storage medium storing a computer program, and the computer program can be executed by a processor to implement the method as described above.
  • the speaker device including the smart power amplifier can be nonlinearly compensated to improve the playback sound quality of the speaker.
  • FIG. 1 is a schematic flowchart of a first embodiment of a method for non-linear compensation of a loudspeaker provided by the present invention
  • Figure 2a is a schematic diagram of the waveform of the system response of the smart power amplifier provided by the present invention.
  • Figure 2b is a schematic diagram of the waveform of the inverse system response of the smart power amplifier provided by the present invention.
  • FIG. 3 is a schematic flowchart of a second embodiment of a method for non-linear compensation of a loudspeaker provided by the present invention
  • FIG. 4a is a schematic diagram of SPL (Sound Pressure Level) waveforms using the method described in this embodiment
  • FIG. 4b is a schematic diagram of a THD (Total Harmonic Distortion) waveform using the method described in this embodiment
  • FIG. 5 is a schematic structural diagram of an embodiment of the speaker device provided by the present invention.
  • Fig. 6 is a schematic structural diagram of an embodiment of a speaker device provided by the present invention.
  • FIG. 7 is a schematic structural diagram of an embodiment of a storage medium provided by the present application.
  • nonlinear compensation Due to the size limitation of the micro-speaker, its nonlinear distortion is very obvious in the case of large amplitude. Therefore, it is necessary to perform non-linear compensation for the large-amplitude micro-speaker.
  • General nonlinear compensation is only applicable to fixed power amplifiers, and smart power amplifiers have speaker protection (for example, amplitude protection, temperature protection), voltage boost and other processing, which will make the effect of nonlinear compensation worse or even completely invalid.
  • the present invention provides a non-linear compensation method of a speaker, which can perform non-linear compensation on a speaker device including a smart power amplifier, and improve the playback sound quality of the speaker.
  • FIG. 1 is a schematic flowchart of a first embodiment of a speaker nonlinear compensation method provided by the present invention.
  • the non-linear compensation method of loudspeaker provided by the present invention includes the following steps:
  • S101 Perform attenuation processing on the original input signal at the current moment to generate a processed input signal.
  • the original input signal is attenuated, the speaker displacement response is adjusted, the output displacement of the speaker is limited, and the processed input signal is obtained.
  • One cause of this nonlinear distortion or damage is excessive vibration displacement of the diaphragm coil assembly of the speaker.
  • S102 Acquire the speaker state parameter and the speaker system parameter at the current moment, and calculate the predistortion signal corresponding to the processed input signal according to the speaker state parameter and the speaker system parameter.
  • the original input signal at the current moment and the speaker output feedback signal at the current moment can be acquired, and the speaker state parameters at the current moment can be acquired according to the original input signal at the current moment and the speaker output feedback signal at the current moment.
  • the predistortion signal of the input signal is calculated and processed according to the speaker state parameters and the speaker system parameters at the current moment.
  • the speaker system parameters can be set by the user or have fixed values corresponding to each speaker.
  • the speaker system parameters include linear parameters and non-linear parameters; linear parameters include at least one of voice coil vibration equivalent quality, voice coil DC resistance, voice coil inductance, voice coil factor linearity, stiffness coefficient linear term and force resistance linear term Term; non-linear parameters include at least one of force factor, stiffness coefficient and force resistance.
  • S103 Obtain system response parameters and frequency domain response parameters of the smart power amplifier, filter the predistortion signal according to the system response parameters, and obtain a compensation signal.
  • the frequency response of the predistortion signal is equalized to compensate for the frequency response distortion of the smart power amplifier.
  • a target frequency range such as 100HZ-1000HZ
  • the frequency response of the smart power amplifier is not ideal, and the gain of different frequencies is different.
  • the predistortion signal is directly used for compensation, the compensation effect may be affected. Therefore, the constructed inverse response of the system is used to perform FIR (Finite Impulse Response, finite unit impulse response) filtering operation on the predistortion signal to obtain a compensation signal, which can solve the non-linear compensation failure caused by the non-linear processing of the smart power amplifier. problem.
  • FIR Finite Impulse Response
  • the frequency domain response parameter H of the smart power amplifier can be calculated according to the following formula:
  • FIG. 2a is a schematic diagram of the waveform of the system response of the smart power amplifier provided by the present invention
  • FIG. 2b is a schematic diagram of the inverse system response of the smart power amplifier provided by the present invention.
  • S104 Input the compensation signal into the smart power amplifier, obtain the playback signal, and input the playback signal into the speaker.
  • the compensation signal is input to the smart power amplifier, and the smart power amplifier performs the compensation signal according to preset protection rules, such as amplitude protection, temperature protection, etc., or according to preset processing rules, such as changing voltage and changing current. Adjust, obtain the playback signal, and input the playback signal into the speaker, so that the audio quality output by the speaker is effectively improved.
  • preset protection rules such as amplitude protection, temperature protection, etc.
  • preset processing rules such as changing voltage and changing current. Adjust, obtain the playback signal, and input the playback signal into the speaker, so that the audio quality output by the speaker is effectively improved.
  • the original input signal is attenuated to obtain the processed input signal.
  • the speaker state parameters at the current moment are calculated, and the speaker state parameters are calculated according to the current speaker state parameters.
  • Non-linear distortion and corresponding pre-distortion signals are generated, the frequency response of the pre-distortion signals is equalized, and the frequency response distortion of the smart power amplifier is compensated, which can solve the problem of non-linear compensation failure caused by the non-linear processing of the smart power amplifier and improve the playback sound quality of the speaker.
  • FIG. 3 is a schematic flowchart of a second embodiment of a method for non-linear compensation of a loudspeaker provided by the present invention.
  • the non-linear compensation method of loudspeaker provided by the present invention includes the following steps:
  • S201 Obtain the speaker diaphragm displacement corresponding to the original input signal, and determine whether the speaker diaphragm displacement exceeds a preset maximum displacement.
  • the speaker diaphragm displacement corresponding to the original input signal is obtained according to the nonlinear speaker diaphragm displacement response function, where the nonlinear speaker diaphragm displacement response function is based on the linear parameter voice coil vibration in the speaker system parameters, etc. effective mass m t, voice coil DC resistance R e, a voice coil linear term force factor b 0, stiffness coefficient of the linear term and nonlinear parameters k 0 factor Bl (x), stiffness coefficient k t (x) and mechanical resistance R m (v) is calculated. It can accurately predict the diaphragm displacement of the nonlinear speaker under a specific audio signal.
  • the preset maximum displacement of the speaker and determine whether the speaker diaphragm displacement exceeds the preset maximum displacement. If it exceeds, the original input signal at the current moment needs to be compressed and filtered to avoid degrading the sound quality and avoid damaging the speaker.
  • the signal transmission characteristic at the current moment is a second-order IIR (Infinite Impulse Response) filter
  • the transfer function in the z-domain is as follows:
  • ⁇ t is the characteristic gain of the filter
  • b 1 and b 2 are the speaker feedforward coefficients
  • a 1 and a 2 are the speaker feedback coefficients.
  • ⁇ t , b 1 , b 2 , a 1 , and a 2 can all be calculated and obtained according to the current speaker state parameters.
  • the displacement response of the loudspeaker can be matched with the displacement amplitude of the diaphragm of the loudspeaker according to the signal after the attenuation processing, which can effectively improve the quality of the output audio of the loudspeaker.
  • S203 Obtain the signal loaded on both ends of the speaker at the previous moment at the current moment, and calculate the speaker state parameter at the current moment according to the signal and the speaker system parameter.
  • the signal loaded on both ends of the speaker at the current moment and the previous moment is obtained, for example, a voltage sensor is configured on the smart power amplifier, and the voltage signal at both ends of the speaker is obtained through the voltage sensor.
  • the predistortion signal of the previous moment and the power amplifier gain of the smart power amplifier at the current moment are obtained, and the speaker two at the current moment is estimated based on the predistortion signal of the previous moment at the current moment and the power amplifier gain of the smart power amplifier.
  • the signal loaded at the end Specifically, calculate the voltage signal applied to both ends of the speaker at the current moment and the previous moment according to the following formula:
  • u (t) is a predistortion signal on the time the current time
  • g (t) is the gain of the amplifier intelligent amplifier
  • v e (t) is the voltage across the speaker signal on a loading time of the current time.
  • the loudspeaker system parameters include linear parameters and non-linear parameters. Including a voice coil linear parameter equivalent mass of the vibration n t, voice coil DC resistance R e, voice coil inductance L e, power factor coil linear term b 0, stiffness coefficient of the linear term mechanical resistance and a linear term k 0 r 0.
  • Non-linear parameters include force factor Bl (x), stiffness factor k t (x) and force resistance R m (v).
  • S205 Obtain the system response parameter and the frequency domain response parameter of the smart power amplifier, and calculate the compensation signal corresponding to the predistortion signal according to the system response parameter and the frequency domain response parameter.
  • S206 Input the compensation signal into the smart power amplifier, obtain the playback signal, and input the playback signal into the speaker.
  • steps S204-S205 are basically the same as steps S103-S104 in the first embodiment of the speaker nonlinear compensation method provided by the present invention, and will not be repeated here.
  • Figure 4a is a schematic diagram of SPL (Sound Pressure Level) waveforms using the method of this embodiment
  • Figure 4b is a THD (Total Harmonic Distortion) using the method of this embodiment. ) Schematic diagram of the waveform.
  • the attenuation coefficient at the current moment is calculated according to the current loudspeaker state parameters
  • the current loudspeaker state parameters are calculated according to the signal loaded at both ends of the loudspeaker and the loudspeaker system parameters at the last moment of the current moment.
  • the calculation of the speaker state parameters and the speaker system parameters corresponds to the predistortion signal of the input signal
  • the compensation signal corresponding to the predistortion signal is calculated according to the system response parameters and frequency domain response parameters of the smart power amplifier. Linear compensation improves the playback sound quality of speakers.
  • FIG. 6 is a schematic structural diagram of an embodiment of the speaker device provided by the present invention.
  • the speaker device 10 is applied to a speaker device including a smart power amplifier, and includes: an offset processing module 11, a nonlinear module 12, a power amplifier equalization module 13, and a smart power amplifier module 14.
  • the offset processing module 11 is used to perform attenuation processing on the original input signal at the current moment to generate a processed input signal.
  • the non-linear module 12 is used to obtain the speaker state parameters and the speaker system parameters at the current moment, and calculate the predistortion signal corresponding to the processed input signal according to the speaker state parameters and the speaker system parameters.
  • the power amplifier equalization module 13 is used to obtain the system response parameters and frequency domain response parameters of the smart power amplifier, and filter the predistortion signal according to the system response parameters to obtain the compensation signal.
  • the smart power amplifier module 14 is used to input the compensation signal into the smart power amplifier, obtain the playback signal, and input the playback signal into the speaker.
  • the speaker device 10 also includes an offset prediction module 15 for obtaining the speaker diaphragm displacement corresponding to the original input signal, and determining whether the speaker diaphragm displacement exceeds a preset maximum displacement; if the speaker diaphragm displacement exceeds the preset maximum displacement, obtain the current The attenuation coefficient of the moment, attenuates the original input signal at the current moment.
  • the offset processing module 11 is used to obtain the attenuation coefficient at the current moment, perform compression filtering processing on the original input signal at the current moment according to the attenuation coefficient, and adjust the displacement response of the speaker.
  • the power amplifier equalization module 13 is also used to perform finite-length unit impulse response filtering on the predistortion signal.
  • the loudspeaker device 10 also includes a loudspeaker model module 16 for acquiring the signal loaded on both ends of the loudspeaker at the previous moment of the current moment, and calculating the loudspeaker state parameters at the current moment according to the signal and the loudspeaker system parameters.
  • the speaker model module 16 is also used to obtain the predistortion signal of the previous moment of the current moment and the power amplifier gain of the smart power amplifier, and estimate the previous moment speaker of the current moment according to the predistortion signal of the previous moment of the current moment and the power amplifier gain of the smart power amplifier Signal loaded at both ends.
  • the loudspeaker system parameters include linear parameters and non-linear parameters; linear parameters include voice coil vibration equivalent mass, voice coil DC resistance, voice coil inductance, voice coil factor linearity, stiffness coefficient linear term and force resistance linear term At least one item; the nonlinear parameter includes at least one of a force factor, a stiffness coefficient, and a force resistance; and the speaker state parameter includes at least one of a voice coil current, a diaphragm displacement, and a diaphragm speed.
  • the speaker device in this embodiment attenuates the original input signal to obtain the processed input signal, calculates the speaker state parameters at the current moment based on the original input signal and the received feedback signal, and calculates the speaker based on the current speaker state parameters And generate corresponding predistortion signals, equalize the frequency response of the predistortion signals, compensate for the frequency response distortion of the smart power amplifier, and can perform nonlinear compensation on the speaker equipment including the smart power amplifier, and improve the playback sound quality of the speaker.
  • FIG. 6 is a schematic structural diagram of an embodiment of the speaker device provided by the present application.
  • the touch terminal 20 includes a processor 21 and a memory 22.
  • the processor 21 is coupled to the memory 22.
  • a computer program is stored in the memory 22, and the processor 21 executes the computer program when working to implement the methods shown in FIGS. 1 and 3. The detailed method can be referred to the above, and will not be repeated here.
  • the speaker device in this embodiment attenuates the original input signal to obtain the processed input signal, calculates the speaker state parameters at the current moment based on the original input signal and the received feedback signal, and calculates the speaker based on the current speaker state parameters And generate corresponding predistortion signals, equalize the frequency response of the predistortion signals, compensate for the frequency response distortion of the smart power amplifier, and can perform nonlinear compensation on the speaker equipment including the smart power amplifier, and improve the playback sound quality of the speaker.
  • FIG. 7 is a schematic structural diagram of an embodiment of a storage medium provided by the present application.
  • At least one computer program 31 is stored in the storage medium 30, and the computer program 31 is used to be executed by the processor to implement the methods shown in FIGS.
  • the storage medium 30 may be a storage chip in a terminal, a hard disk, or a mobile hard disk, a USB flash drive, an optical disk, or other tools for writing storage, or a server or the like.
  • the computer program in the storage medium in this embodiment can be used to attenuate the original input signal to obtain the processed input signal, and calculate the speaker state parameters at the current moment based on the original input signal and the received feedback signal. Calculate the nonlinear distortion of the speaker according to the current speaker state parameters, and generate the corresponding predistortion signal, perform frequency response equalization on the predistortion signal, compensate for the frequency response distortion of the smart power amplifier, and can perform nonlinear compensation for the speaker equipment including the smart power amplifier. Improve the playback sound quality of speakers.
  • the present invention not only considers the nonlinear distortion in the actual work of the loudspeaker, but also considers the nonlinear problem of the intelligent power amplifier processing, reasonably predicts the compensation signal, and can perform nonlinear compensation for the loudspeaker equipment including the intelligent power amplifier.
  • the playback sound quality of the speakers is not only considers the nonlinear distortion in the actual work of the loudspeaker, but also considers the nonlinear problem of the intelligent power amplifier processing, reasonably predicts the compensation signal, and can perform nonlinear compensation for the loudspeaker equipment including the intelligent power amplifier.

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Abstract

本发明实施例公开了一种扬声器的非线性补偿方法,应用于包括智能功放的扬声器设备,扬声器的非线性补偿方法包括:对当前时刻的原始输入信号进行位移压缩滤波处理,生成处理输入信号;获取当前时刻的扬声器状态变量和扬声器系统参数,根据扬声器状态变量和扬声器系统参数计算对应于处理输入信号的预失真信号;获取智能功放的系统响应参数和频域响应参数,根据系统响应参数和频域响应参数计算对应于预失真信号的补偿信号;将补偿信号输入智能功放,获取播放信号,将播放信号输入扬声器。本发明还提供了扬声器设备、装置和存储介质。本发明可以能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。

Description

扬声器的非线性补偿方法、扬声器设备、装置和存储介质 技术领域
本发明涉及智能功放技术领域,尤其涉及扬声器的非线性补偿方法、扬声器设备、装置和存储介质。
背景技术
便携式智能终端设备的快速发展和普及,影音娱乐等多媒体应用越来越受到关注。良好的声重放是多媒体应用用户体验重要部分。大振幅微型扬声器和智能功放的广泛使用极大提高了便携式智能终端的声重放效果。但是,由于微型扬声器的尺寸限制,在大振幅情况下,其非线性失真十分明显。所以需要对大振幅微型扬声器进行非线性补偿。一般的非线性补偿只适用于固定功放,而智能功放由于存在扬声器保护(例如,振幅保护、温度保护)、电路升压等处理,会让非线性补偿的效果变差,甚至完全失效。
申请内容
基于此,有必要针对上述问题,提出了扬声器的非线性补偿方法、扬声器设备、装置和存储介质。
一种扬声器的非线性补偿方法,应用于包括智能功放的扬声器设备,所述扬声器的非线性补偿方法包括:对当前时刻的原始输入信号进行衰减处理,生成处理输入信号;获取当前时刻的扬声器状态参数和扬声器系统参数,根据所述扬声器状态参数和所述扬声器系统参数计算对应于所述处理输入信号的预失真信号;获取所述智能功放的系统响应参数和频域响应参数,根据所述系统响应参数对所述预失真信号进行滤波,获取补偿信号;将所述补偿信号输入所述智能功放,获取播放信号,将所述播放信号输入扬声器。
一种扬声器装置,用于包括智能功放的扬声器设备,包括:偏移处理模块,用于对当前时刻的原始输入信号进行衰减处理,生成处理输入信号;非线性模块,用于获取当前时刻的扬声器状态参数和扬声器系统参数,根据所述扬声器状态参数和所述扬声器系统参数计算对应于所述处理输入信号的预失真信号;功放均衡模块,用于获取所述智能功放的系统响应参数和频域响应参数,根据所述系统响应参数对所述预失真信号进行滤波,获取补偿信号;智能功放模块,用于将所述补偿信号输入所述智能功放,获取播放信号,将所述播放信号输入扬声器。
一种扬声器设备,包括:处理器和存储器,所述处理器耦接所述存储器,所述存储器中存储有计算机程序,所述处理器执行所述计算机程序以实现如上所述的方法。
一种可读存储介质,存储有计算机程序,所述计算机程序能够被处理器执行以实现如上所述的方法。
采用本发明实施例,具有如下有益效果:
通过对原始输入信号进行衰减处理,获取处理输入信号,根据原始输入信号以及接受的反馈信号,计算当前时刻的扬声器状态参数,根据当前扬声器状态参数计算扬声器的非线性失真,并产生相应的预失真信号,对预失真信号进行频响均衡,补偿智能功放的频响失真,能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。
附图说明
为了更清楚地说明本发明实施例或现有技术中的技术方案,下面将对实施例或现有技术描述中所需要使用的附图作简单地介绍,显而易见地,下面描述中的附图仅仅是本发明的一些实施例,对于本领域普通技术人员来讲,在不付出创造性劳动的前提下,还可以根据这些附图获得其他的附图。
其中:
为了更清楚地说明本发明实施例或现有技术中的技术方案,下面将对实施例或现有技术描述中所需要使用的附图作简单地介绍,显而易见地,下面描述中的附图仅仅是本发明的一些实施例,对于本领域普通技术人员来讲,在不付出创造性劳动的前提下,还可以根据这些附图获得其他的附图。
其中:
图1是本发明提供的扬声器的非线性补偿方法的第一实施例的流程示意图;
图2a是本发明提供的智能功放的系统响应的波形示意图;
图2b是本发明提供的智能功放的逆系统响应的波形示意图。
图3是本发明提供的扬声器的非线性补偿方法的第二实施例的流程示意图;
图4a是采用本实施例所述方法的SPL(Sound Pressure Level,声压级)波形示意图;
图4b是采用本实施例所述方法的THD(Total Harmonic Distortion,总谐波失真)波形示意图;
图5是本发明提供的扬声器装置的一实施例的结构示意图;
图6是本发明提供的扬声器设备的一实施例的结构示意图;
图7是本申请提供的存储介质的一实施例的结构示意图。
具体实施方式
下面将结合本发明实施例中的附图,对本发明实施例中的技术方案进行清楚、完整地描述,显然,所描述的实施例仅仅是本发明一部分实施例,而不是全部的实施例。基于本发明中的实施例,本领域普通技术人员在没有作出创造性劳动前提下所获得的所有其他实施例,都属于本发明保护的范围。
由于微型扬声器的尺寸限制,在大振幅情况下,其非线性失真十分明显。所以需要对大振幅微型扬声器进行非线性补偿。一般的非线性补偿只适用于固 定功放,而智能功放由于存在扬声器保护(例如,振幅保护、温度保护)、电压Boost等处理,会让非线性补偿的效果变差,甚至完全失效。
为了解决上述问题,本发明提供了一种扬声器的非线性补偿方法,能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。
请参阅图1,图1是本发明提供的扬声器的非线性补偿方法的第一实施例的流程示意图。本发明提供的扬声器的非线性补偿方法包括如下步骤:
S101:对当前时刻的原始输入信号进行衰减处理,生成处理输入信号。
在一个具体的实施场景中,对原始输入信号进行衰减处理,调整扬声器位移响应,限制扬声器的输出位移,获取处理输入信号。这样可以使得输入扬声器的信号保持低于某一限制。如果该信号振幅过大,则扬声器将产生非线性失真或将被不可修复地损坏。该非线性失真或损坏的一个原因是扬声器的膜片线圈组的过度的振动位移。通过对原始输入信号进行衰减处理,可以有效将信号振幅限制在预设范围内,且保留原始信号的至少部分特征。
S102:获取当前时刻的扬声器状态参数和扬声器系统参数,根据扬声器状态参数和扬声器系统参数计算对应于处理输入信号的预失真信号。
在本实施场景中,可以获取当前时刻的原始输入信号和当前时刻的扬声器输出反馈信号,根据当前时刻的原始输入信号和当前时刻的扬声器输出反馈信号获取当前时刻的扬声器状态参数。根据当前时刻的扬声器状态参数和扬声器系统参数计算处理输入信号的预失真信号。
扬声器系统参数可以由用户设置或者为对应于每个扬声器的固定值。扬声器系统参数包括线性参数和非线性参数;线性参数包括音圈振动等效质量、音圈直流电阻、音圈电感、音圈因数线性想、劲度系数线性项和力阻线性项中的至少一项;非线性参数包括力因数、劲度系数和力阻中的至少一项。
S103:获取智能功放的系统响应参数和频域响应参数,根据系统响应参数对预失真信号进行滤波,获取补偿信号。
在本实施场景中,对预失真信号进行频响均衡,补偿智能功放的频响失真。 具体地说,在一目标频率范围内,例如100HZ-1000HZ,智能功放的频响不理想,且不同的频率的增益不同,若直接采用预失真信号进行补偿则有可能影响补偿的效果。因此,使用构造出来的系统逆响应对预失真信号进行FIR(Finite Impulse Response,有限长单位冲激响应)滤波操作,获取补偿信号,该补偿信号可以解决智能功放非线性处理导致非线性补偿失效的问题。
假设智能功放的系统响应参数为h,则该智能功放的频域响应参数H可以根据以下公式计算:
Figure PCTCN2020096683-appb-000001
构造逆系统响应h inv,使其对应的频域响应H inv=H -1,根据逆系统响应h inv获取对应于预失真信号的补偿信号。请结合参阅图2a和图2b,图2a为本发明提供的智能功放的系统响应的波形示意图,图2b为本发明提供的智能功放的逆系统响应的波形示意图。
S104:将补偿信号输入智能功放,获取播放信号,将播放信号输入扬声器。
在本实施场景中,将补偿信号输入智能功放,智能功放根据预设的保护规则,例如振幅保护、温度保护等,或者根据预设的处理规则,例如改变电压、改变电流等,对补偿信号进行调节,获取播放信号,将播放信号输入扬声器,从而扬声器输出的音频质量得到有效提升。
通过上述描述可知,在本实施例中通过对原始输入信号进行衰减处理,获取处理输入信号,根据原始输入信号以及接受的反馈信号,计算当前时刻的扬声器状态参数,根据当前扬声器状态参数计算扬声器的非线性失真,并产生相应的预失真信号,对预失真信号进行频响均衡,补偿智能功放的频响失真,能够解决智能功放非线性处理导致非线性补偿失效的问题,提升扬声器的播放音质。
请参阅图3,图3是本发明提供的扬声器的非线性补偿方法的第二实施例的流程示意图。本发明提供的扬声器的非线性补偿方法包括如下步骤:
S201:获取原始输入信号对应的扬声器振膜位移,判断扬声器振膜位移是 否超过预设最大位移。
在本实施场景中,根据非线性扬声器振膜位移响应函数获取原始输入信号对应的扬声器振膜位移,其中,非线性扬声器振膜位移响应函数是是根据扬声器系统参数中的线性参数音圈振动等效质量m t,音圈直流电阻R e,音圈力因数线性项b 0,劲度系数线性项k 0和非线性参数因数Bl(x),劲度系数k t(x)和力阻R m(v)计算得到。能够对非线性扬声器在特定音频信号下的振膜位移做出准确预测。
获取扬声器的预设最大位移,判断扬声器振膜位移是否超过预设最大位移,若超过,则需要对对当前时刻的原始输入信号进行压缩滤波处理,避免降低音质以及避免对扬声器造成损伤。
S202:获取当前时刻的衰减系数,根据衰减系数对当前时刻的原始输入信号进行压缩滤波处理,调整扬声器位移响应。
在一个具体的实施场景中,当前时刻的信号传输特性为二阶的IIR(Infinite Impulse Response,无限脉冲响应)滤波器,在z域的传输函数如下列公式:
Figure PCTCN2020096683-appb-000002
其中,σ t是滤波器的特性增益,b 1和b 2为扬声器前馈系数,a 1和a 2是扬声器反馈系数。σ t、b 1、b 2、a 1、a 2均可根据当前扬声器状态参数计算获取。
扬声器位移响应可以根据经过衰减处理后的信号与扬声器的振膜位移幅度相匹配,可以有效提升扬声器输出音频的质量。
S203:获取当前时刻的上一时刻扬声器两端加载的信号,根据信号和扬声器系统参数计算当前时刻的扬声器状态参数。
在本实施场景中,获取当前时刻的上一时刻扬声器两端加载的信号,例如在智能功放上配置电压传感器,通过电压传感器获取扬声器两端的电压信号。在其他实施场景中,获取当前时刻的上一时刻的预失真信号和智能功放的功放增益,根据当前时刻的上一时刻的预失真信号和智能功放的功放增益估算当前时刻的上一时刻扬声器两端加载的信号。具体地说,根据以下公式计算当前时 刻的上一时刻扬声器两端加载的电压信号:
v e(t)=u(t)*g(t)
其中,u(t)为当前时刻的上一时刻的预失真信号,g(t)为智能功放的功放增益,v e(t)为当前时刻的上一时刻扬声器两端加载的电压信号。
S204:根据扬声器状态参数和扬声器系统参数计算对应于处理输入信号的预失真信号。
在本实施场景中,扬声器状态参数为x=[i,x,v] T,其中,i为音圈电流,x为振膜位移,v为振膜速度。扬声器系统参数包括线性参数和非线性参数。线性参数包括音圈振动等效质量n t,音圈直流电阻R e,音圈电感L e,音圈力因数线性项b 0,劲度系数线性项k 0和力阻线性项r 0。非线性参数包括力因数Bl(x),劲度系数k t(x)和力阻R m(v)。
根据以下公式计算预失真信号:
Figure PCTCN2020096683-appb-000003
S205:获取智能功放的系统响应参数和频域响应参数,根据系统响应参数和频域响应参数计算对应于预失真信号的补偿信号。
S206:将补偿信号输入智能功放,获取播放信号,将播放信号输入扬声器。
在本实施场景中,步骤S204-S205与本发明提供的扬声器的非线性补偿方法的第一实施例中的步骤S103-S104基本一致,此处不再进行赘述。
请结合参阅图4a和图4b,图4a是采用本实施例方法的SPL(Sound Pressure Level,声压级)波形示意图,图4b是采用本实施例方法的THD(Total Harmonic Distortion,总谐波失真)波形示意图。
通过上述描述可知,在本实施例中根据当前扬声器状态参数计算获取当前 时刻的衰减系数,根据所当前时刻的上一时刻扬声器两端加载的信号和扬声器系统参数计算当前时刻的扬声器状态参数,根据扬声器状态参数和扬声器系统参数计算对应于处理输入信号的预失真信号,根据智能功放的系统响应参数和频域响应参数计算对应于预失真信号的补偿信号,能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。
请参阅图6,图6是本发明提供的扬声器装置的一实施例的结构示意图。扬声器装置10应用于包括智能功放的扬声器设备,包括:偏移处理模块11、非线性模块12、功放均衡模块13和智能功放模块14。
偏移处理模块11用于对当前时刻的原始输入信号进行衰减处理,生成处理输入信号。非线性模块12用于获取当前时刻的扬声器状态参数和扬声器系统参数,根据扬声器状态参数和扬声器系统参数计算对应于处理输入信号的预失真信号。功放均衡模块13用于获取智能功放的系统响应参数和频域响应参数,根据系统响应参数对预失真信号进行滤波,获取补偿信号。智能功放模块14用于将补偿信号输入智能功放,获取播放信号,将播放信号输入扬声器。
扬声器装置10还包括偏移预测模块15,用于获取原始输入信号对应的扬声器振膜位移,判断扬声器振膜位移是否超过预设最大位移;若扬声器振膜位移超过预设最大位移,则获取当前时刻的衰减系数,对当前时刻的原始输入信号进行衰减处理。
偏移处理模块11用于获取当前时刻的衰减系数,根据衰减系数对当前时刻的原始输入信号进行压缩滤波处理,并调整扬声器位移响应。
功放均衡模块13还用于对预失真信号进行有限长单位冲激响应滤波。
扬声器装置10还包括扬声器模型模块16,用于获取当前时刻的上一时刻扬声器两端加载的信号,根据信号和扬声器系统参数计算当前时刻的扬声器状态参数。
扬声器模型模块16还用于获取当前时刻的上一时刻的预失真信号和智能功放的功放增益,根据当前时刻的上一时刻的预失真信号和智能功放的功放增 益估算当前时刻的上一时刻扬声器两端加载的信号。
其中,扬声器系统参数包括线性参数和非线性参数;线性参数包括音圈振动等效质量、音圈直流电阻、音圈电感、音圈因数线性想、劲度系数线性项和力阻线性项中的至少一项;非线性参数包括力因数、劲度系数和力阻中的至少一项;扬声器状态参数包括音圈电流、振膜位移和振膜速度中的至少一项。
通过上述描述可知,本实施例中扬声器装置通过对原始输入信号进行衰减处理,获取处理输入信号,根据原始输入信号以及接受的反馈信号,计算当前时刻的扬声器状态参数,根据当前扬声器状态参数计算扬声器的非线性失真,并产生相应的预失真信号,对预失真信号进行频响均衡,补偿智能功放的频响失真,能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。
请参阅图6,图6是本申请提供的扬声器设备的一实施例的结构示意图。触控终端20包括处理器21、存储器22。处理器21耦接存储器22。存储器22中存储有计算机程序,处理器21在工作时执行该计算机程序以实现如图1和图3所示的方法。详细的方法可参见上述,在此不再赘述。
通过上述描述可知,本实施例中扬声器设备通过对原始输入信号进行衰减处理,获取处理输入信号,根据原始输入信号以及接受的反馈信号,计算当前时刻的扬声器状态参数,根据当前扬声器状态参数计算扬声器的非线性失真,并产生相应的预失真信号,对预失真信号进行频响均衡,补偿智能功放的频响失真,能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。
请参阅图7,图7是本申请提供的存储介质的一实施例的结构示意图。存储介质30中存储有至少一个计算机程序31,计算机程序31用于被处理器执行以实现如图1和图3所示的方法,详细的方法可参见上述,在此不再赘述。在一个实施例中,存储介质30可以是终端中的存储芯片、硬盘或者是移动硬盘或者优盘、光盘等其他写存储的工具,还可以是服务器等等。
通过上述描述可知,本实施例中存储介质中的计算机程序可以用于通过对原始输入信号进行衰减处理,获取处理输入信号,根据原始输入信号以及接受的反馈信号,计算当前时刻的扬声器状态参数,根据当前扬声器状态参数计算扬声器的非线性失真,并产生相应的预失真信号,对预失真信号进行频响均衡,补偿智能功放的频响失真,能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。
区别于现有技术,本发明中除了考虑扬声器实际工作中的非线性失真,也考虑了智能功放处理的非线性问题,合理预测补偿信号,能够对包括智能功放的扬声器设备进行非线性补偿,提升扬声器的播放音质。
以上所揭露的仅为本发明较佳实施例而已,当然不能以此来限定本发明之权利范围,因此依本发明权利要求所作的等同变化,仍属本发明所涵盖的范围。

Claims (10)

  1. 一种扬声器的非线性补偿方法,其特征在于,包括:
    获取扬声器当前时刻的前一时刻的终端电压;
    根据所述终端电压获取所述扬声器当前时刻的状态变量;
    获取待处理信号,修改所述扬声器原始的线性参数,根据修改后所述线性参数对所述待处理信号进行线性处理,获取线性处理信号;
    根据扬声器的非线性参数和所述当前时刻的状态变量对所述线性处理信号进行非线性处理,获取非线性信号,将所述非线性信号输出至所述扬声器。
  2. 根据权利要求1所述的扬声器的非线性补偿方法,其特征在于,所述扬声器的线性参数包括电阻、电感、力因子、质量、刚度系数、阻尼系数中的至少一项;
    所述扬声器的非线性参数包括总力因子、总刚度系数、总阻尼系数中的至少一项;
    所述当前时刻的状态变量包括所述扬声器的位移、速度和电流中的至少一项。
  3. 根据权利要求1所述的扬声器的非线性补偿方法,其特征在于,所述修改所述扬声器原始的线性参数的步骤,包括:
    增加所述扬声器的阻尼系数以降低所述扬声器的力学品质因子;或
    修改所述扬声器的刚度系数。
  4. 根据权利要求3所述的扬声器的非线性补偿方法,其特征在于,所述修改所述扬声器的刚度系数的步骤,包括:
    获取所述扬声器的响应时间、释放时间、弹性系数变化率和弹性系数变化曲线过渡带对应的位移,根据所述响应时间、所述释放时间、所述弹性系数变化率和所述弹性系数变化曲线过渡带对应的位移修改所述扬声器的刚度系数。
  5. 根据权利要求1所述的扬声器的非线性补偿方法,其特征在于,所述将所述非线性信号输出至所述扬声器的步骤,包括:
    对所述非线性信号进行压缩处理,获取压缩信号,将所述压缩信号输出至所述扬声器。
  6. 根据权利要求1所述的扬声器的非线性补偿方法,其特征在于,所述将所述非线性信号输出至所述扬声器的步骤,包括:
    所述获取待处理信号的步骤,包括:
    获取原始输入信号,获取所述扬声器的预测终端信号,根据所述预测终端信号对所述原始输入信号进行压缩处理,获取所述待处理信号。
  7. 根据权利要求5或6所述的扬声器的非线性补偿方法,其特征在于,所述进行压缩处理的步骤包括:
    获取所述扬声器的响应时间、释放时间、电压阈值和拐点宽度,根据所述应时间、所述释放时间、所述电压阈值和所述拐点宽度进行压缩处理。
  8. 一种扬声器的非线性补偿系统,其特征在于,包括:
    延时模块,用于获取扬声器当前时刻的前一时刻的终端电压;
    状态估计模块,用于根据所述终端电压获取扬声器当前时刻的状态变量;
    线性滤波模块,用于获取待处理信号,修改所述扬声器原始的线性参数,根据修改后所述线性参数对所述待处理信号进行线性处理,获取线性处理信号;
    非线性滤波模块,用于根据扬声器的非线性参数和所述当前时刻的状态变量对所述线性处理信号进行非线性处理,获取非线性信号,将所述非线性信号输出至所述扬声器。
  9. 一种声器的非线性补偿设备,其特征在于,包括:处理器和存储器,所述处理器耦接所述存储器,所述存储器中存储有计算机程序,所述处理器执行所述计算机程序以实现如权利要求1-7任一项所述的方法。
  10. 一种存储介质,其特征在于,存储有计算机程序,所述计算机程序能够被处理器执行以实现如权利要求1-7任一项所述的方法。
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