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CN119603607A - Speaker excursion prediction system - Google Patents

Speaker excursion prediction system Download PDF

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Publication number
CN119603607A
CN119603607A CN202311163913.0A CN202311163913A CN119603607A CN 119603607 A CN119603607 A CN 119603607A CN 202311163913 A CN202311163913 A CN 202311163913A CN 119603607 A CN119603607 A CN 119603607A
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China
Prior art keywords
stroke
speaker
circuit
audio signal
sampling frequency
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CN202311163913.0A
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Chinese (zh)
Inventor
张荣贵
余文隆
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Elite Semiconductor Memory Technology Inc
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Elite Semiconductor Memory Technology Inc
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Priority to CN202311163913.0A priority Critical patent/CN119603607A/en
Publication of CN119603607A publication Critical patent/CN119603607A/en
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    • 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
    • H04R3/00Circuits for transducers, loudspeakers or microphones
    • 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
    • H04R2430/00Signal processing covered by H04R, not provided for in its groups
    • H04R2430/03Synergistic effects of band splitting and sub-band processing

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

The invention provides a loudspeaker stroke prediction system which is suitable for a loudspeaker protection circuit. The speaker stroke prediction system includes a low-pass filter circuit, a down-sampling circuit, and an impulse response generation unit. A low pass filter circuit configured to generate an audio signal X LPF (t) passing through the low pass filter circuit from the audio signal X (t) having the first sampling frequency. And a downsampling circuit coupled to the low-pass filter circuit and configured to downsample the first sampling frequency of the audio signal X LPF (t) output after passing through the low-pass filter circuit to a second sampling frequency to generate a downsampled audio signal X LPFDN (t). An impulse response generating unit is coupled to the downsampling circuit and configured to generate a stroke prediction value Y (t) according to the loudspeaker stroke transfer function and the downsampled audio signal X LPFDN (t). Therefore, when the stroke prediction circuit adopts a low-order filter, the problem of the stroke prediction accuracy of the loudspeaker is reduced.

Description

Speaker stroke prediction system
Technical Field
The present invention relates to a stroke estimation circuit of a speaker, and more particularly, to a stroke prediction system of a speaker.
Background
The loudspeaker is generally composed of a magnet, a coil and a diaphragm. When an electric current passes through the coil to generate an electromagnetic field, the coil vibrates with the diaphragm, pushing ambient air to vibrate, and the speaker thereby generates sound. The energy conversion process of the loudspeaker is to convert electric energy into magnetic energy, then convert the magnetic energy into mechanical energy and then convert the mechanical energy into sound.
The stroke excursion of the loudspeaker means an offset (hereinafter referred to as stroke) by which the diaphragm starts to move from a rest position by a driving force until a maximum displacement is reached. The size of the stroke affects the performance of the speaker, such as volume, power and distortion, and is therefore critical to the design and optimization of the speaker. Because temporary or permanent mechanical damage may be caused to the speaker when the speaker exceeds the maximum stroke limit, causing an abnormal or non-audible sound to be generated by the speaker. The mechanical damage may be caused by a collision between the coil, diaphragm or voice coil former and the stationary part. In order to avoid mechanical damage to the loudspeaker, it is necessary to estimate the stroke of the diaphragm accurately.
In terms of stroke prediction of loudspeakers, one common approach is based on acoustic modeling and simulation techniques. By building a physical model of the speaker and the driving system, the vibration and acoustic characteristics of the speaker can be predicted and analyzed. In this case, the speaker is characterized by several basic parameters (e.g., parameters including load characteristics, diaphragm elasticity and damping, etc.), and can also be estimated using measured speaker output current and voltage. A unity impulse response (Impulse Res ponse) filter is built up based on the aforementioned basic parameters of the loudspeaker to process the output response of the loudspeaker.
The unit impulse response of the loudspeaker refers to the transient response output by the loudspeaker when the loudspeaker receives an impulse signal, and is used for evaluating performance indexes such as frequency response, harmonic distortion and the like of the loudspeaker, and the frequency response of the impulse response of the loudspeaker, which is modeled in advance, is subjected to Fourier series analysis through an input audio signal, so that the resonant frequency and bandwidth of the loudspeaker can be obtained, and further, the predicted stroke response under the audio input can be deduced. The stroke value of the loudspeaker is estimated through a stroke prediction circuit of infinite impulse response (infinite impulse response, IIR) or finite impulse response (finite impulse response, FIR) corresponding to the loudspeaker, and the audio signal output to the loudspeaker is limited according to the stroke value, so that the problem that the maximum stroke limit (Xmax) of the loudspeaker is exceeded is avoided.
Disclosure of Invention
However, in the above-described stroke prediction circuit, it is generally necessary to consider factors such as frequency response and system stability. In practical applications, the order of the impulse response is generally related to the required system accuracy, and the stroke prediction circuit can provide higher prediction accuracy by using an impulse response with a higher order, but also can result in increased calculation and storage, and if the filter is adjusted to a low order, the stroke prediction accuracy of the loudspeaker is found to be significantly worse. Applications for estimating the stroke of a loudspeaker require a balance between computational and memory requirements and accuracy to determine the optimal impulse response order.
In the previous case, the 1024-order impulse response generated by the speaker under the same audio input can accurately simulate the predicted stroke response output, and when the impulse response order is reduced to 128 orders, the simulated predicted stroke response output will cause a large error offset, which is a prediction error caused by insufficient resolution. The audio sampling rate refers to the number of times a sound signal is sampled over a period of time. It is typically expressed in units of samples per second in hertz (Hz). Typical audio sampling rates are 44.1kHz, 48kHz, 96kHz, etc. The higher the sampling rate, the higher the sampling accuracy of the sound signal. Referring to fig. 1A, a schematic diagram of a stroke response distribution of a commercially available full-range speaker is shown, and it can be found that most of the sound contributions are below the bandwidth of 3 kHz. Referring to fig. 1B, another schematic diagram of the stroke response distribution of a commercially available full-range speaker is shown, and it can be also found that most of the sound contribution is below the bandwidth of 3 kHz.
In view of the above-mentioned prior art, the present invention provides a speaker stroke prediction system, which can still maintain the effect of high accuracy under the condition of decreasing the impulse response order in cooperation with the proposed system architecture, the audio sampling rate and the speaker stroke response distribution relation.
To achieve the above and other objects, the present invention provides a speaker stroke prediction system including a low pass filter circuit, a downsampling circuit, and an impulse response generating unit. A low-pass filter circuit configured to generate an audio signal X LPF (t) output after passing through the low-pass filter circuit from the audio signal X (t) having the first sampling frequency F S1. And a downsampling circuit coupled to the low-pass filter circuit and configured to downsample the first sampling frequency F S1 of the audio signal X LPF (t) output after passing through the low-pass filter circuit to the second sampling frequency F S2 to generate a downsampled audio signal X LPFDN (t). An impulse response generating unit is coupled to the downsampling circuit and configured to generate a stroke prediction value Y (t) according to the loudspeaker stroke transfer function and the downsampled audio signal X LPFDN (t).
In some embodiments, the first sampling frequency F S1 is 48kHz and the second sampling frequency F S2 is adjustable.
In some embodiments, the second sampling frequency F S2 is 1/8 times the first sampling frequency F S1.
In some embodiments, the second sampling frequency F S2 is a component of 95% of the stroke response that is encompassed within the 0.5 frequency point.
In some embodiments, the impulse response generation unit is an impulse response filter.
In some embodiments, the order of the designed impulse response filter is 128 orders when the second sampling frequency F S2 is 6 kHz.
In some embodiments, the cut-off frequency of the low pass filter circuit is less than half of the second sampling frequency F S2.
In some embodiments, the cut-off frequency of the low pass filter circuit is less than-60 dB at the 0.5 frequency point of the second sampling frequency F S2.
In some embodiments, a stroke conversion circuit is further included, coupled to the impulse response generation unit, configured to generate the speaker stroke transfer function.
In some embodiments, a stroke conversion circuit is coupled to the impulse response generation unit and configured to generate a speaker stroke transfer function. And a protection circuit coupled to the impulse response generation unit and configured to generate a speaker stroke protection value according to the stroke prediction value. And a gain controller coupled to the protection circuit and configured to generate a maximum stroke limited audio signal based on the delayed audio signal and the speaker stroke protection value. And a delay circuit coupled to the gain controller and configured to generate a delayed audio signal from the audio signal.
Therefore, the loudspeaker stroke prediction system of the invention improves the accuracy of the impulse response generating unit at low frequency through the downsampling circuit, and can effectively reduce the order of the filter circuit of the impulse response generating unit, thereby using less memory storage space and lower DSP operand.
Drawings
Fig. 1A is a schematic diagram of a commercially available full range speaker stroke response profile.
Fig. 1B is a schematic diagram of another commercially available full range speaker stroke response profile.
FIG. 2 is a system block diagram of an embodiment of the present invention.
Fig. 3 is a waveform diagram of an input audio signal X (t) according to an embodiment of the present invention.
Fig. 4 is a schematic diagram of the first sampling F S1 frequency and the second sampling F S2 frequency according to an embodiment of the present invention.
FIG. 5 is a schematic diagram of an impulse response of an embodiment of the invention using a finite impulse response filter for 128-order operations.
Fig. 6A is a diagram of a stroke prediction output analog waveform according to an embodiment of the present invention.
FIG. 6B is a waveform diagram of actual stroke output measurement according to an embodiment of the present invention.
FIG. 7 is a system block diagram of another embodiment of the present invention.
Fig. 8 is a system block diagram of a further embodiment of the present invention.
The reference numerals are explained as follows:
101,201,301 low pass filter circuit
102,202,302 Downsampling circuit
103,203,303 Impulse response generation unit
204,304 Stroke conversion circuit
305. Protection circuit
306. Delay circuit
307. Gain controller
F S1 first sampling frequency
F S2 second sampling frequency
Detailed Description
For a full understanding of the objects, features and effects of the present invention, reference should now be made to the following detailed description of the invention taken in conjunction with the accompanying drawings, in which:
Referring to fig. 2, a system block diagram of an embodiment of the present invention is shown. As shown in fig. 2, the speaker stroke prediction system includes a low-pass filter circuit 101, a downsampling circuit 201, and an impulse response generation unit 103.
The low-pass filter circuit 101 is configured to generate an audio signal X LPF (t) output after passing through the low-pass filter circuit 101 from the audio signal X (t) having the first sampling frequency F S1. Referring to fig. 3, a waveform diagram of an input audio signal X (t) according to an embodiment of the invention is shown, wherein a first sampling frequency F S1 of the input audio signal X (t) is 48kHz. Since experiments have found that the audio signal X (t) in the high frequency band contributes little or even no to the stroke prediction. Therefore, the low frequency band of the audio signal X (t) is extracted by the low-pass filter circuit 101 to generate the audio signal X LPF (t) output after passing through the low-pass filter circuit 101.
The downsampling circuit 102 is coupled to the low-pass filter circuit 101. The downsampling circuit 102 is configured to downsample the first sampling frequency F S1 of the audio signal X LPF (t) output after passing through the low-pass filter circuit 101 to the second sampling frequency F S2 to generate a downsampled audio signal X LPFDN (t). Please refer to fig. 4, which illustrates a first sampling F S1 frequency and a second sampling F S2 frequency according to an embodiment of the present invention. It can be found that the second sampling frequency F S2 of the audio signal X LPF (t) output via the low-pass filter circuit 101 and the down-sampling circuit 102 is 6kHz. In addition, the downsampling circuit 102 is helpful for the order design of the impulse response generating unit 103 at the filter circuit. More specifically, the down-sampling circuit 101 may allow the impulse response generating unit 103 to use a filter of a lower order. In some embodiments, the cut-off frequency of the low-pass filter circuit 101 is less than half of the second sampling frequency F S2. Preferably, half of the second sampling frequency F S2 is less than-60 dB.
In some embodiments, the second sampling frequency F S2 is adjustable. The second sampling frequency F S2 is 1/N times the first sampling frequency F S1, N being a positive integer. For example, the second sampling frequency F S2 is 1/8 times the first sampling frequency F S 1. For example, when the first sampling frequency is 48kHz, the second sampling frequency is 6kHz. When the first sampling frequency is 96kHz, the second sampling frequency is 12kHz or less. In some embodiments, the second sampling frequency F S2 is within a 0.5 frequency point of the 95% component of the stroke response that is covered.
The impulse response generating unit 103 is coupled to the downsampling circuit 102. An impulse response generation unit 103 is configured to generate a stroke prediction value Y (t) from the loudspeaker stroke transfer function and the downscaled audio signal X LPFDN (t). More specifically, the impulse response generation unit 103 performs a time-domain convolution (convolution) operation on the speaker stroke transfer function H (t) and the downscaled audio signal X LPFDN (t) to generate a stroke prediction value Y (t) corresponding to the displacement distance. Fig. 5 is a schematic diagram of an impulse response calculated for 128 orders using a finite impulse response filter according to an embodiment of the present invention. The impulse response generation unit 103 may be an impulse response filter, such as a finite impulse response (Finite Impulse Response, FIR) filter or an infinite impulse response (Infinite Impulse Response, IIR) filter. In some embodiments, the order of the finite impulse response filter is 128 orders when the second sampling frequency F S2 is 6 kHz. When the second sampling frequency F S2 is 6kHz, it is applicable to a speaker having a frequency response range of 3 kHz.
Referring to fig. 6A and 6B together, as shown in fig. 6A and 6B, the vertical axis represents the stroke number in millimeters (mm). The horizontal axis is time and the units are seconds(s). Fig. 6A is a diagram of a stroke prediction output analog waveform generated by a speaker stroke prediction system. Fig. 6B is a waveform diagram of the actual output of the stroke of the speaker measured by the laser range finder. It can be seen that the waveforms are very similar, i.e. the speaker stroke prediction results of the embodiments of the present invention have high accuracy.
Referring to fig. 7, a system block diagram of another embodiment of the present invention is shown. As shown in fig. 3, the speaker stroke prediction system includes a low-pass filter circuit 201, a downsampling circuit 202, an impulse response generating unit 203, and a stroke conversion circuit 204. The low-pass filter circuit 201, the down-sampling circuit 202 and the impulse response generating unit 203 disclosed in this embodiment are the same as the low-pass filter circuit 101, the down-sampling circuit 102 and the impulse response generating unit 103 disclosed in fig. 2, and are not described in detail below.
The stroke conversion circuit 204 is coupled to the impulse response generation unit 203. The stroke conversion circuit 204 is configured to generate a speaker stroke transfer function H(s) from the speaker parameters. Since the speaker parameters belong to the frequency domain signal, after inverse Laplace (Laplace) operation, a speaker stroke transfer function H (t) of the time domain signal can be generated to perform a correlation operation with the downsampled audio signal X LPFDN (t), which is also the time domain signal.
Referring to fig. 8, a system block diagram of another embodiment of the present invention is shown. As shown in fig. 8, the speaker stroke prediction system includes a low-pass filter circuit 301, a downsampling circuit 302, an impulse response generation unit 303, a stroke conversion circuit 304, a protection circuit 305, a delay circuit 306, and a gain controller 307. The low-pass filter circuit 301, the downsampling circuit 302, the impulse response generating unit 303 and the stroke converting circuit 304 in this embodiment are the same as the low-pass filter circuit 201, the downsampling circuit 202, the impulse response generating unit 203 and the stroke converting circuit 204 in fig. 7, and will not be described in detail.
The protection circuit 305 is coupled to the impulse response generation unit 303. The protection circuit 305 is configured to generate a speaker stroke protection value to the gain controller 307 based on the stroke prediction value Y (t). A delay circuit 306 configured to generate a delayed audio signal to the gain controller 307 from the audio signal X (t).
The gain controller 307 is coupled to the protection circuit 305 and the delay circuit 306. The gain controller 307 is configured to generate the audio signal Y (t) of the maximum stroke limit from the delayed audio signal and the speaker stroke protection value to avoid that the speaker exceeds the maximum stroke limit (Xmax).
Next, the embodiment of the present invention compares the differences between the prior art and the embodiment of the present invention through experimental results, please refer to table one, as follows:
List one
As shown in the above table, the stroke prediction circuit of the prior art adopts a 1024-order filter circuit, but the embodiment of the present invention only needs to adopt a 128-order filter circuit. In comparison, the present invention can save about 1/8 times of memory space on the storage space of the memory.
In the prior art, a filter circuit with a sampling frequency of 48kHz and 1024 orders is adopted, and for the calculation amount of the DSP, the calculation is carried out by a pulse period of 48Mega to obtain the result of the stroke prediction value. The embodiment of the invention adopts a filter circuit with sampling frequency of 6kHz and 128 orders and a low-pass filter circuit. The low pass filter circuit is estimated to take approximately 1440K (10+1440K) of the pulse period of the DSP to operate. The DSP operand is calculated only by a pulse period of 2208K, and the result of the stroke prediction value Y (t) can be obtained. Overall, the embodiment of the invention can save about 1/24 times of DSP operand. In other words, embodiments of the present invention have less memory usage space and lower DSP operand where high accuracy (> 98%) stroke prediction values are also achievable.
In summary, in the speaker stroke prediction system of the present invention, the down-sampling circuit improves the accuracy of the impulse response generating unit at low frequency, and can effectively reduce the order of the filter circuit of the impulse response generating unit, so as to use less memory storage space and lower DSP operand.
The invention has been disclosed above in terms of preferred embodiments, however, it will be understood by those skilled in the art that the embodiments are merely illustrative of the invention and should not be construed as limiting the scope of the invention. It should be noted that all changes and substitutions equivalent to the described embodiments are considered to be covered by the scope of the present invention. Accordingly, the scope of the invention is defined by the appended claims.

Claims (10)

1.一种扬声器冲程预测系统,其特征在于,包括:1. A speaker stroke prediction system, comprising: 低通滤波电路,被配置为根据具有第一采样频率的音频信号X(t),A low-pass filter circuit is configured to, according to an audio signal X(t) having a first sampling frequency, 生成通过所述低通滤波电路的音频信号XLPF(t);Generate an audio signal X LPF (t) passing through the low-pass filter circuit; 降采样电路,耦接所述低通滤波电路,被配置为将通过所述低通滤波电路后输出的音频信号XLPF(t)的所述第一采样频率降低采样至第二采样频率,以生成被降采音频信号XLPFDN(t);及a down-sampling circuit, coupled to the low-pass filter circuit, configured to down-sample the first sampling frequency of the audio signal X LPF (t) output after passing through the low-pass filter circuit to a second sampling frequency to generate a down-sampled audio signal X LPFDN (t); and 脉冲响应生成单元,耦接所述降采样电路,被配置为根据扬声器冲程转移函数与所述被降采音频信号XLPFDN(t),生成冲程预测数值Y(t)。The impulse response generating unit is coupled to the down-sampling circuit, and is configured to generate a stroke prediction value Y(t) according to a speaker stroke transfer function and the down-sampled audio signal X LPFDN (t). 2.根据权利要求1所述的扬声器冲程预测系统,其特征在于,所述第一采样频率为48kHz,所述第二采样频率为可调整的。2. The speaker stroke prediction system according to claim 1 is characterized in that the first sampling frequency is 48 kHz and the second sampling frequency is adjustable. 3.根据权利要求1所述的扬声器冲程预测系统,其特征在于,所述第二采样频率为所述第一采样频率的1/8倍。3 . The speaker stroke prediction system according to claim 1 , wherein the second sampling frequency is 1/8 times of the first sampling frequency. 4.根据权利要求1所述的扬声器冲程预测系统,其特征在于,所述第二采样频率为0.5倍的频率点内涵盖的冲程响应的95%的分量。4 . The speaker stroke prediction system according to claim 1 , wherein the second sampling frequency is 0.5 times the frequency point covering 95% of the components of the stroke response. 5.根据权利要求1所述的扬声器冲程预测系统,其特征在于,所述脉冲响应生成单元为脉冲响应滤波器。5 . The speaker stroke prediction system according to claim 1 , wherein the impulse response generation unit is an impulse response filter. 6.根据权利要求5所述的扬声器冲程预测系统,其特征在于,当所述第二采样频率为6kHz时,设计所述脉冲响应滤波器的阶数为128阶。6 . The speaker stroke prediction system according to claim 5 , characterized in that when the second sampling frequency is 6 kHz, the order of the impulse response filter is designed to be 128. 7.根据权利要求1所述的扬声器冲程预测系统,其特征在于,所述低通滤波电路的截止频率小于所述第二采样频率的一半。7 . The speaker stroke prediction system according to claim 1 , wherein a cutoff frequency of the low-pass filter circuit is less than half of the second sampling frequency. 8.根据权利要求1所述的扬声器冲程预测系统,其特征在于,所述第二采样频率的一半小于-60dB。8 . The speaker stroke prediction system according to claim 1 , wherein half of the second sampling frequency is less than -60 dB. 9.根据权利要求1所述的扬声器冲程预测系统,其特征在于,还包括冲程转换电路,与所述脉冲响应生成单元耦接,被配置为生成所述扬声器冲程转移函数。9. The speaker stroke prediction system according to claim 1 is characterized in that it also includes a stroke conversion circuit, which is coupled to the impulse response generation unit and is configured to generate the speaker stroke transfer function. 10.根据权利要求9所述的扬声器冲程预测系统,其特征在于,还包括有:10. The speaker stroke prediction system according to claim 9, further comprising: 保护电路,与所述脉冲响应生成单元耦接,被配置为根据所述冲程预测数值Y(t),生成扬声器冲程保护数值;A protection circuit, coupled to the impulse response generation unit, configured to generate a speaker stroke protection value according to the stroke prediction value Y(t); 增益控制器,与所述保护电路耦接,被配置为根据延迟音频信号与所述扬声器冲程保护数值,生成最大冲程限制的音频信号;及a gain controller coupled to the protection circuit and configured to generate an audio signal with a maximum excursion limit according to the delayed audio signal and the speaker excursion protection value; and 延迟电路,与所述增益控制器耦接,被配置为根据所述音频信号生成所述延迟音频信号。The delay circuit is coupled to the gain controller and configured to generate the delayed audio signal according to the audio signal.
CN202311163913.0A 2023-09-11 2023-09-11 Speaker excursion prediction system Pending CN119603607A (en)

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