CN101233783B - Loudspeaker device - Google Patents

Abstract

The loudspeaker device according to the present invention includes: a loudspeaker; a feed-forward processing component for performing feed-forward processing on an electrical signal to be input to the loudspeaker based on a preset filter coefficient, so as to eliminate nonlinear distortion occurring in the loudspeaker; and feedback processing means for detecting vibration of the speaker and performing feedback processing on the electrical signal related to the vibration with respect to the electrical signal to be input to the speaker. The feedback processing section performs feedback processing on the electric signal related to the vibration, thereby canceling the nonlinear distortion occurring in the speaker and making the frequency characteristic according to the vibration of the speaker become a predetermined frequency characteristic .

Description

speaker device

technical field

The present invention relates to a speaker device, and more particularly to a speaker device for canceling distortion occurring in a speaker.

Background technique

Conventionally, it is desirable to faithfully convert electrical signals into sound waves in ordinary speakers without electrical signal processing. However, due to their structural limitations, it is difficult for actual loudspeakers to perform faithful conversions. For example, in the magnetic circuit constituting a loudspeaker, due to its structure, the magnetic flux density in the magnetic gap decreases as the amplitude increases. Then, the force coefficient also decreases as the magnetic flux density decreases. Due to the structure of the support system such as dampers, edges, etc., the stiffness of the support system varies with the magnitude of the vibration amplitude. For these reasons, the amplitude of the speaker is not necessarily proportional to the amplitude of the input electric signal, so there is a problem that nonlinear distortion occurs.

As a method of eliminating the above-described nonlinear distortion, a method utilizing electrical signal processing such as feedforward processing or the like has conventionally been proposed. This processing method is a method in which polynomial approximation is performed on parameters including nonlinear components of the speaker (force coefficient according to magnetic flux density, rigidity of support system, etc.), and filter coefficients are set so as to eliminate The nonlinear distortion for this parameter. An electric signal is input through a filter into a speaker whose filter coefficient is set, thereby canceling nonlinear distortion. However, among the parameters, especially the rigidity of the support system changes all the time and also ages. In other words, parameter values change over time. Therefore, in the above-mentioned feedforward processing, the error between the parameter preset value and the parameter actual value becomes larger with time, so there is a disadvantage that the above-mentioned effect of eliminating distortion is significantly deteriorated.

In order to solve the above problems, in feedforward processing, a method of adaptively updating filter coefficient parameters has been proposed (for example, refer to Patent Document 1). This method will be described below with reference to FIG. 28 . FIG. 28 is a block diagram showing a conventional speaker device 9 which adaptively updates filter coefficient parameters.

In FIG. 28 , a conventional speaker device 9 includes a control section 91 , a parameter detector 92 and a speaker 95 . The parameter detector 92 includes an error circuit 93 and an update circuit 94 . The error circuit 93 includes a filter (not shown), and calculates a pseudo-vibration characteristic at the filter from a signal input from the control section. The error circuit 93 predictively calculates the driving voltage to be applied to the speaker 95 from this pseudo-vibration characteristic. Note that this predicted drive voltage is equivalent to the impedance characteristic when the speaker 95 is driven by current. The error circuit 93 then generates an error signal e(t) by subtracting the actual drive voltage applied to the speaker 95 from the predicted drive voltage. The error signal e(t) is input to the update circuit 94 .

Based on this error signal e(t), the update circuit 94 calculates the parameters to be updated in the control section 91 . The parameters calculated by the update circuit 94 are returned to the filter of the error circuit 93, and the gradient signal Sg is generated by the error circuit 93. The gradient signal Sg generated by the error circuit 93 is output to the update circuit 94 again. Then, the update circuit 94 calculates parameters using the above-mentioned error signal e(t) and the gradient signal Sg so that the error signal e(t) becomes minimum. The parameter at which the error signal e(t) becomes the minimum is output to the control section 91 as a power vector P, and the parameter in the control section 91 is updated. As described above, in the speaker device 9 shown in FIG.

[Patent Document 1] Japanese Patent Laid-Open No. 11-46393

Contents of the invention

The problem to be solved by the present invention

However, the error circuit 93 and the update circuit 94 for updating parameters require complex and large mathematical operations. Also, as mentioned above, the stiffness of the support system varies moment to moment with the magnitude of the electrical signal input to the loudspeaker. In other words, since the conventional speaker device 9 requires complex and numerous mathematical operations, it is extremely difficult for the conventional speaker device 9 to actually perform parameter update processing in order to track the above-mentioned drastic changes in the rigidity of the supporting system. As a result, the conventional speaker device 9 has problems in that the distortion canceling effect cannot be sufficiently obtained, and it lacks practicality. In addition, the conventional speaker device 9 has a problem of lack of cost performance since the conventional speaker device 9 implements a large number of mathematical operations.

Accordingly, an object of the present invention is to provide a speaker device that performs signal processing to track parameter changes in an actual speaker and is capable of performing more stable distortion removal processing.

problem solution

A first aspect is a loudspeaker device comprising: a loudspeaker including a diaphragm; a support system assembly including a rim and a damper for supporting the diaphragm to allow the diaphragm to vibrate; and a voice coil generating a driving force that causes the vibration of the vibrating membrane; a feedforward processing section for performing feedforward processing on an electric signal to be input to the speaker based on a filter coefficient including at least a signal representing the support system fixed parameters for modeling the stiffness of the vibration displacement of the component relative to the vibration displacement of the diaphragm and a vibration displacement characteristic representing a force coefficient applied to the voice coil relative to the vibration displacement of the diaphragm fixed parameters of the modulus, the filter coefficients are set to eliminate the nonlinear component of each parameter; and a feedback processing part for detecting the vibration of the diaphragm and relative to the electrical signal feedback processing of the electrical signal associated with the vibration, wherein the feedback processing component performs feedback processing of the electrical signal associated with the vibration to cancel a vibration displacement characteristic indicative of stiffness of the support system component and make the frequency characteristic related to the vibration of the vibrating membrane become a desired frequency characteristic.

In a second aspect according to the first aspect, the feedback processing section includes an ideal filter for receiving the electric signal to be input to the speaker and converting a frequency characteristic of the received electric signal into a desired a frequency characteristic; a sensor for detecting the vibration of the vibrating membrane; a first adder for obtaining the electrical signal converted by the ideal filter and representing the desired frequency characteristic and detecting the difference between the electric signals related to the vibration, and outputting the electric signal of the difference as an error signal; and a second adder for processing the electric signal processed by the feedforward processing part The error signal is summed and the resulting electrical signal is output to the speaker.

The filter coefficient of the feedforward processing section is a coefficient based on a parameter unique to the speaker; and the feedforward processing section processes the electric signal to be input to the speaker so as to cancel nonlinearity of the parameter portion.

The filter coefficient of the feedforward processing section is a coefficient based on a parameter unique to the speaker; and the parameter is a parameter that changes according to a vibration displacement of the speaker.

In a fifth aspect according to the second aspect, the feedforward processing part includes a cancellation filter for receiving the electric signal to be input to the speaker, and processing the received electric signal based on the filter coefficient. signal; and a linear filter for receiving the electrical signal to be input to the speaker and generating an electrical signal representing a vibration displacement of the vibrating membrane when the vibrating membrane linearly vibrates, and the cancellation filter A detector is referenced to the electrical signal generated by the linear filter and representing the vibration displacement.

In a sixth aspect according to the fifth aspect, the speaker device further includes: a power amplifier provided between the second adder and the speaker for amplifying the electrical signal to be input to the speaker , and the filter coefficients of the cancellation filter, the filter coefficients of the ideal filter, and the filter coefficients of the linear filter are filters multiplied by the reciprocal of the gain value amplified by the power amplifier coefficient.

In a seventh aspect according to the second aspect, the electrical signal detected by the sensor is an electrical signal representing the vibration displacement of the vibrating membrane, and the feedforward processing part refers to The electrical signal obtained and representing the vibration displacement.

In an eighth aspect according to the second aspect, the speaker device further includes a pre-stage filter provided at a stage preceding the feedforward processing part for receiving the electric signal to be input to the speaker, and The received electrical signal is processed based on filter coefficients obtained by subtracting a characteristic of the speaker related to the vibration from the desired frequency characteristic.

In a ninth aspect according to the second aspect, the speaker device further includes a limiter for limiting a level of an electric signal so as not to input an electric signal having a level equal to or higher than a predetermined level to the speaker.

In a tenth aspect according to the second aspect, the speaker device further includes a power amplifier provided between the second adder and the speaker for amplifying the electric signal to be input to the speaker. gain, and the filter coefficient of the feedforward processing part and the filter coefficient of the ideal filter are filter coefficients multiplied by the reciprocal of the gain value amplified by the power amplifier.

In an eleventh aspect according to the first aspect, the feedforward processing means is provided at a position before the speaker and in a feedback loop formed by the feedback processing means.

In a twelfth aspect according to the first aspect, the feedback processing section includes an ideal filter for receiving the electric signal to be input to the speaker and converting a frequency characteristic of the received electric signal into a desired frequency characteristic; a sensor for detecting the vibration of the vibrating membrane; a first adder for obtaining the electric signal converted by the ideal filter and representing the desired frequency characteristic and combined with the electric signal by the a difference between the electric signals related to the vibration detected by the sensor, and outputting the electric signal of the difference as an error signal; and a second adder for comparing the electric signal to be input with the error signal and output the obtained electric signal to the feedforward processing part, and the feedforward processing part performs feedforward processing on the electric signal output by the second adder, and outputs the obtained electric signal to the speaker electrical signal.

In a thirteenth aspect according to the twelfth aspect, the speaker device further includes a low-pass filter provided between the second adder and the feedforward processing section, which has a filter coefficient for The gain of the electric signal to be input to the speaker expresses a characteristic inclining with a gradient of -6dB/oct or less in a frequency band equal to or lower than a first frequency, and the first frequency is equal to or higher than the first frequency determined by the The open-loop transfer characteristic of the feedback loop formed by the feedback processing components described above represents the frequency of the gain spanning frequency.

In a fourteenth aspect according to the twelfth aspect, the speaker device further includes a high-pass filter provided at a stage preceding the feedforward processing section, which has filter coefficients for making the input to the speaker The electrical signal gain represents a characteristic of being inclined with a gradient of 6 dB/oct or more in a frequency band equal to or lower than a second frequency, and the second frequency is equal to or higher than a feedback loop formed by the feedback processing part The open-loop transfer characteristic represents the frequency of gain across frequencies.

In a fifteenth aspect according to the twelfth aspect, the speaker device further includes: a low-pass filter provided between the second adder and the feedforward processing section, having a filter coefficient with A characteristic for causing an electric signal gain to be input to the speaker to express a gradient slope of -6 dB/oct or less in a frequency band equal to or lower than the first frequency; and provided at a stage preceding the feedforward processing part a high-pass filter having filter coefficients for making the gain of the electric signal to be input to the speaker express a characteristic inclined at a gradient of 6 dB/oct or more in a frequency band equal to or lower than the second frequency, and The first and second frequencies are frequencies equal to or higher than a gain crossover frequency represented by an open-loop transfer characteristic of a feedback loop formed by the feedback processing section.

The filter coefficient of the feedforward processing part is a coefficient based on a parameter unique to the loudspeaker; and the feedforward processing part processes the electric signal output from the second adder so as to eliminate the parameter non-linear components.

The filter coefficient of the feedforward processing section is a coefficient based on a parameter unique to the speaker; and the parameter is a parameter that changes according to a vibration displacement of the speaker.

In an eighteenth aspect according to the twelfth aspect, the feedforward processing part includes a cancellation filter for receiving the electrical signal output from the second adder, and processing the received electrical signal based on filter coefficients. an electrical signal; and a linear filter for receiving the electrical signal output from the second adder and generating an electrical signal representing a vibration displacement of the vibrating membrane when the vibrating membrane linearly vibrates, and the obtained The cancellation filter references the electrical signal representative of the vibration displacement produced by the linear filter.

In a nineteenth aspect according to the eighteenth aspect, the speaker device further includes a power amplifier provided between the feedforward processing section and the speaker for amplifying the power input to the speaker. The gain of the signal, and the filter coefficient of the cancellation filter, the filter coefficient of the ideal filter and the filter coefficient of the linear filter are filtered by multiplying the inverse of the gain value amplified by the power amplifier device coefficient.

In a twentieth aspect according to the twelfth aspect, the electrical signal detected by the sensor is an electrical signal representing the vibration displacement of the vibrating membrane, and the feedforward processing part refers to The electrical signal detected and indicative of the vibrational displacement.

In a twenty-first aspect according to the twelfth aspect, the speaker device further includes a pre-filter provided at a position in front of the second adder for receiving the electric signal to be input to the speaker, and processing the received electrical signal based on filter coefficients obtained by subtracting a characteristic of the speaker related to the vibration from the desired frequency characteristic.

In a twenty-second aspect according to the twelfth aspect, the speaker device further includes a limiter for limiting the level of an electric signal so as not to input an electric signal having a level equal to or higher than a predetermined level to the speaker. .

In a twenty-third aspect according to the twelfth aspect, the speaker device further includes a power amplifier provided between the feedforward processing part and the speaker for amplifying the power input to the speaker. The gain of the signal, and the filter coefficient of the feedforward processing part and the filter coefficient of the ideal filter are filter coefficients multiplied by the reciprocal of the gain value amplified by the power amplifier. In a twenty-fourth aspect according to the first aspect, the change in the vibrational displacement characteristic indicative of the stiffness of the support system component is due to a slow change in the material forming the support system component or the material forming the support system component. due to the creep phenomenon of the material. In a twenty-fifth aspect according to the first aspect, the material forming the support system component is cloth or resin.

A twenty-sixth aspect is an integrated circuit for processing electrical signals to be input to a speaker, the speaker comprising: a diaphragm; a support system assembly including a lip and a damper for supporting the diaphragm to allow the a vibrating membrane; and a voice coil that generates a driving force that causes the vibrating membrane to vibrate, the integrated circuit including: a feedforward processing part for forwarding an electric signal to be input to the speaker based on a filter coefficient Feedback processing, the filter coefficients include at least fixed parameters representing the vibration displacement characteristics representing the rigidity of the vibration displacement of the support system components relative to the diaphragm and representing the vibration displacement applied to the voice coil relative to fixed parameters for modeling the vibration displacement characteristics of the force coefficient of the vibration displacement of the vibration membrane, the filter coefficients are set to eliminate the nonlinear component of each parameter; and a feedback processing part for detecting the vibration displacement of the vibration membrane vibration, and feedback-processing the electric signal related to the vibration with respect to the electric signal to be input to the speaker, and the feedback processing part feeds back the electric signal related to the vibration processing so as to eliminate variations in vibration displacement characteristics indicative of rigidity of the support system components and to make frequency characteristics of the vibration by the diaphragm into desired frequency characteristics.

Effect of the present invention

According to the first aspect, most of the nonlinear distortion can be eliminated by feedforward processing based on the filter coefficients set to eliminate the nonlinear component of each parameter. Furthermore, through feedback processing, robust distortion cancellation is possible with respect to, for example, slow variations in the stiffness of the support system of the loudspeaker. In other words, according to the present aspect, the feedforward processing section performs processing based on the above-mentioned filter coefficients, and the feedback processing section performs the above robust distortion cancellation, thereby providing a speaker device capable of performing more stable distortion cancellation with high feasibility, There is no need to update the speaker parameters. Furthermore, according to the present aspect, by feedback processing, it is possible to eliminate variations in vibration displacement characteristics indicating rigidity of support system components, and to approximate frequency characteristics related to speaker vibration to desired frequency characteristics.

According to the second aspect, most of the nonlinear distortion can be eliminated by feed-forward processing based on the filter coefficients set to eliminate the nonlinear component of each parameter, and support with respect to, for example, a speaker can be performed by feedback processing based on the error signal Robust distortion cancellation for slowly varying stiffness of the system. Thus, it is possible to provide a speaker device capable of performing more stable distortion canceling processing with high feasibility. Furthermore, according to the present aspect, the frequency characteristics related to speaker vibration can be approximated to desired frequency characteristics by an ideal filter.

Note that nonlinear distortion occurring at a speaker can be canceled by processing an electrical signal to be input to the speaker so as to cancel the nonlinear component of the parameter.

Furthermore, highly accurate distortion removal processing depending on the vibration displacement of the speaker can be performed.

According to the fifth aspect, processing based on the vibration displacement when the diaphragm vibrates linearly is possible, and more efficient distortion removal processing can be performed.

According to the sixth aspect, even when the voltage that can be processed in the internal operation of the cancellation filter, the ideal filter, and the linear filter is small, processing that maintains the effect of canceling distortion is possible. Also, by providing a power amplifier in the feedback loop, the feedback gain can be made large, and the distortion canceling effect can be improved.

According to the seventh aspect, it is possible to perform distortion canceling processing according to the vibration of an actual speaker.

According to the eighth aspect, in the characteristics related to the vibration output from the speaker, convergence to desired frequency characteristics can be enhanced.

According to the ninth aspect, it is possible to prevent the speaker from being damaged due to excessive input.

According to the tenth aspect, even in the case where the voltage that can be processed in the internal operation of the feedforward processing section and the ideal filter is small, processing that maintains the effect of canceling distortion is possible. Also, by providing a power amplifier in the feedback loop, the feedback gain can be made large, and the distortion canceling effect can be improved.

According to the eleventh aspect, by arranging the feedforward processing means in the feedback loop, even when the amplitude of the speaker becomes large, it is possible to achieve the effect of canceling distortion in the lower frequency band.

According to the twelfth aspect, by arranging the feedforward processing means in the feedback loop, even when the amplitude of the speaker becomes large, it is possible to achieve the effect of canceling distortion in the lower frequency band.

According to the thirteenth aspect, since the gain is lowered by the low-pass filter across frequencies, the effect of canceling distortion can be achieved in a lower frequency band.

According to the fourteenth aspect, since the high-pass filter does not input an electrical signal whose frequency is equal to or lower than the gain crossover frequency, distortion due to an input electrical signal whose frequency is equal to or lower than the gain crossover frequency can be eliminated in advance, and higher The effect of eliminating distortion.

According to the fifteenth aspect, since the gain is lowered by the low-pass filter across frequencies, the effect of canceling distortion can be achieved in a lower frequency band. In addition, since the high-pass filter does not input an electrical signal whose frequency is equal to or lower than the gain spanning frequency, the distortion caused by the input frequency of an electrical signal equal to or lower than the gain spanning frequency can be eliminated in advance, and a higher degree of distortion elimination can be obtained. Effect.

Description of drawings

FIG. 1 is a block diagram showing an exemplary configuration of a speaker device 1 according to the first embodiment.

FIG. 2 is a cross-sectional view of the universal speaker 16 .

FIG. 3 shows an example of the behavior of the force coefficient B1 with respect to the vibration displacement x in the vicinity of the magnetic gap 165 .

Figure 4 shows an example of the relationship between the stiffness K of the support system and the vibration displacement x.

Fig. 5 shows the variation of the stiffness K with respect to the input signal I(t).

FIG. 6 shows desired output characteristics, which are set as filter coefficients of the ideal filter 12 .

FIG. 7 is a block diagram showing an exemplary configuration of the speaker device 1 when the output signal of the sensor 17 is referred to by the nonlinear component canceling filter 10 .

FIG. 8 is a block diagram showing an exemplary configuration of a speaker device 2 according to the second embodiment.

FIG. 9 is a block diagram showing an exemplary configuration in which the input of the linear filter 11 shown in FIG. 8 is changed.

FIG. 10 is a block diagram showing an exemplary configuration of the speaker device 2 when the output signal of the sensor 17 is referred to by the nonlinear component canceling filter 10 .

FIG. 11 is a block diagram showing an exemplary configuration of a speaker device 3 according to the third embodiment.

FIG. 12 shows gain characteristics and phase characteristics of the speaker device 3 .

FIG. 13 shows a configuration for analyzing the frequency characteristics of the speaker device 2 shown in FIG. 10 .

FIG. 14 shows gain characteristics, second-order distortion characteristics, and third-order distortion characteristics when the input of the speaker 16 of FIG. 13 is changed.

FIG. 15 is a block diagram showing an exemplary configuration in which a compensation filter 21 is added to the speaker device 3 shown in FIG. 11 .

Fig. 16 shows the frequency characteristics of the transfer function shown by Equation (18).

FIG. 17 is a block diagram showing an exemplary configuration in which a high-pass filter 22 is added to the speaker device 3 shown in FIG. 11 .

FIG. 18 is a block diagram showing an exemplary configuration in which a compensation filter 21 and a high-pass filter 22 are added to the speaker device 3 shown in FIG. 11 .

FIG. 19 shows analysis results when the input is 20W and 40W.

FIG. 20 shows a feedback loop of the speaker device 2 shown in FIG. 10 .

Figure 21 shows a step input and its response in the feedback loop shown in Figure 20.

Figure 22 shows a step input and its response in the feedback loop shown in Figure 20.

Figure 23 shows a step input and its response in the feedback loop shown in Figure 20.

FIG. 24 is a block diagram showing an exemplary configuration of a speaker device 4 according to the fourth embodiment.

Fig. 25 shows a comparison of frequency characteristics with and without scaling.

FIG. 26 shows an exemplary configuration in which the strength of the power amplifier 23 is correlated with each element.

FIG. 27 is a block diagram showing an exemplary configuration in which the limiter 24 is provided in the speaker device 1 shown in FIG. 1 .

FIG. 28 is a block diagram showing a conventional speaker device 9 .

Explanation of reference signs

1, 2 speaker units

10 Nonlinear component elimination filter

11 Linear filter

12 ideal filter

13, 14 adder

15 Feedback Control Filter

16 speakers

17 sensors

20 pre-filter

21 Compensation filter

22 high pass filter

23 power amplifier

24 limiter

161 voice coil

162 diaphragm

163 magnets

164 magnetic circuit

165 magnetic gap

166 damper

167 edge

Detailed ways

Embodiments of the present invention will be described below with reference to the drawings.

(first embodiment)

Referring to Fig. 1, a speaker device 1 according to a first embodiment of the present invention will be described. FIG. 1 is a block diagram showing an exemplary configuration of a speaker device 1 according to the first embodiment. As shown in FIG. 1 , speaker device 1 includes nonlinear component canceling filter 10 , linear filter 11 , ideal filter 12 , adders 13 and 14 , feedback control filter 15 , speaker 16 and sensor 17 .

Here, referring to FIG. 2 , the reason why nonlinear distortion occurs in the speaker 16 will be described. FIG. 2 is a cross-sectional view of the universal speaker 16 . As shown in FIG. 2 , the speaker 16 includes a voice coil 161 , a diaphragm 162 , a magnet 163 , a magnetic circuit 164 , a damper 166 and an edge 167 . A magnetic gap 165 is formed in the magnetic circuit 164 shown in FIG. 2 . According to Fleming's left-hand rule, when the magnetic flux density in the magnetic gap 165 is B and a current flows in the voice coil 161 , the voice coil 161 and the diaphragm 162 vibrate together along the axial direction of the vibration displacement x. The vibrating membrane 162 is supported by the damper 166 and the edge 167, so the vibrating membrane 162 vibrates stably in the axial direction of the vibration displacement x, thereby emitting sound. Note that the speaker 16 shown in FIG. 2 is an example, and it is not limited thereto. For example, it may be a shielded speaker including canceling magnets, or a speaker including an inner magnet type magnetic circuit. In addition, in FIG. 2, the position where the vibration displacement x is zero indicates the center position of the vibration of the voice coil 161 and the diaphragm 162, corresponding to the origin where the vibration displacement x is zero shown in FIGS. 3 to 5 described later.

In the speaker 16, the causes of nonlinear distortion mainly include three causes. The first reason relates to the magnetic flux density B present in the magnetic gap 165 . FIG. 3 shows an example of the force coefficient B1 with respect to the vibrational displacement x in the vicinity of the magnetic gap 165 . When the amplitude of the voice coil 161 is small, that is, when the absolute value of the vibration displacement x is small (x is around zero), the magnetic flux density B is substantially constant. However, when the amplitude of the voice coil 161 is large, that is, when the absolute value of the vibration displacement x is large, the magnetic flux density B rapidly decreases. This is because it is difficult to form a magnetic flux path since the axial direction of the vibration displacement x is far from the vicinity of the center of the magnetic gap 165 (x is near zero). Therefore, the relationship between the force coefficient B1 obtained from the magnetic flux density B and the vibration displacement x of the voice coil 161 is the relationship shown in FIG. 3 . Note that the behavior of the force coefficient B1 shown in FIG. 3 varies according to the vibration displacement x, expressed as a function B1(x) of the vibration displacement x.

Here, when the input signal current flowing through the voice coil 161 is represented by I(t), the driving force F(t) for vibrating the voice coil 161 is represented by the following equation (1):

F(t)＝B1(x)*I(t) (1)

As shown in FIG. 3 , as the amplitude of the voice coil 161 increases, the force coefficient value B1(x) decreases. Therefore, according to equation (1), when the vibration amplitude is large, the driving force F(t) is not proportional to the level of the input signal I(t). In addition, if the driving force F(t) is not proportional to the level of the input signal I(t), obviously, the vibration displacement x is also not proportional to the level of the input signal I(t). Therefore, nonlinear distortion occurs in the speaker 16 .

The second reason involves support systems such as dampers 166, edges 167, and the like. The damper 166 and the edge 167 are not stretched infinitely due to their shape, and begin to tighten when stretched to a certain extent. Fig. 4 shows an example of the relation characteristic of the stiffness K of the support system and the vibration displacement x. As shown in FIG. 4, when the amplitude of the voice coil 161 is small, that is, when the absolute value of the vibration displacement x is small, the rigidity K is substantially constant. However, when the amplitude of the voice coil 161 is large, that is, when the absolute value of the vibration displacement x is large, the rigidity K becomes large. Therefore, when the amplitude becomes large, the value of the rigidity K changes, and the vibration displacement x is not proportional to the driving force F(t). Furthermore, if the vibration displacement x is not proportional to the driving force F(t), according to the above equation (1), the vibration displacement x is not proportional to the level of the input signal I(t). As a result, speaker 16 is distorted nonlinearly.

Fig. 5 shows the variation of the stiffness K characteristic with respect to the input signal I(t). As shown in Fig. 5, the characteristic of the stiffness K varies with the magnitude of the I(t) level, and does not constantly provide a constant curve. Since both the damper 166 and the edge 167 are made of materials such as cloth, resin, etc., the characteristics of the rigidity K shown in FIG. 4 may change even due to slow changes in materials and creep phenomena. Even for these reasons the vibration displacement x is not proportional to the level of the input signal I(t), so that the loudspeaker 16 is distorted nonlinearly.

A third reason involves the electrical impedance characteristics of the voice coil 161 . A high-permeability material such as iron is used for the magnetic circuit of the speaker. Therefore, the inductance component included in the voice coil 161 varies with the magnitude of the amplitude. When an electric signal is input to the voice coil 161, it generates heat. Therefore, the resistance component of the voice coil 161 changes with time. Due to these factors, the current flowing through the voice coil 161 is distorted, and the speaker 16 is distorted nonlinearly. Due to the above three main reasons, the loudspeaker 16 suffers from non-linear distortion.

Note that when the speaker 16 is driven by a constant voltage, the relationship between the input signal voltage E(t) input to the speaker 16 and the vibration displacement x is roughly expressed by the following equation (2):

B1*E(t)/Ze＝K*x(t)+(r+B1 ² /Ze)*dx(t)/dt+m*d ² x(t)/dt ² (2)

Note that in equation (2), the rigidity of the support system is represented by K, the mechanical resistance of the speaker 16 by r, the electrical impedance of the voice coil 161 by Ze, and the vibration system mass by m.

Here, among the above three causes, the influence of the force coefficient B1 and the stiffness K parameter is large especially in the nonlinear distortion occurring in the low frequency band. When expressing the force coefficient B1 and rigidity K shown in FIGS. 3 and 4 as a function of the vibration displacement x in equation (2), the following equation (3) is provided.

B1(x)*E(t)/Ze

=K(x)*x(t)+(r+B1(x) ² /Ze)*dx(t)/dt+m*d ² x(t)/dt ² (3)

Furthermore, when B1(x) and K(x) are polynomially approximated with respect to the vibration displacement x and modeled, the following equations (4) and (5) are provided.

B1(x)＝A0+A1*x+A2*x ² +A3*x ³ +... (4)

K(x)＝K0+K1*x+K2*x ² +K3*x ³ +... (5)

In the above equation (4) and the above equation (5), A0 and K0 are linear component parameters independent of the vibration displacement x. Then, when both equation (4) and equation (5) are divided into linear components and nonlinear components and expressed, they are represented as equation (6) and equation (7), respectively.

B1(x)＝A0+Ax (6)

K(x)＝K0+Kx (7)

Note that Ax is a nonlinear component of B1(x), and Kx is a nonlinear component of K(x). Thus, when B1(x) and K(x) in Equation (3) are substituted into Equation (6) and Equation (7), Equation (8) is provided.

(A0+Ax)*E(t)/Ze

=(K0+Kx)*x(t) ten [r ten (A0+Ax) ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ² (8)

Operation processing of the speaker device 1 shown in FIG. 1 will be described below. In the speaker device 1 according to the present embodiment, roughly speaking, feedforward processing is performed by the nonlinear component canceling filter 10 and the linear filter 11, and the filter is controlled by the ideal filter 12, the sensor 17, the adder 14, the feedback 15 and adder 13 perform feedback processing. Therefore, the nonlinear component canceling filter 10 and the linear filter 11 correspond to feedforward processing means of the present invention. Also, the ideal filter 12, the sensor 17, the adder 14, the feedback control filter 15, and the adder 13 correspond to feedback processing means of the present invention.

Feedforward processing by the nonlinear component canceling filter 10 and the linear filter 11 will be described. The electric signal is input into the nonlinear component removing filter 10 , the linear filter 11 and the ideal filter 12 as input signals. The processing of the ideal filter 12 will be described later.

The nonlinear component removal filter 10 processes the input signal to remove the nonlinear component of the modeled parameter based on predetermined filter coefficients, which are vibration displacement x(t )acquired. Then, the signal processed by the nonlinear component removing filter 10 is output to the adder 13 . The predetermined filter coefficients set at the nonlinear component canceling filter 10 will be described below.

The operating equation of speaker 16 is shown in equation (8) above. From the above equation (8), the working equation excluding the nonlinear components (B1x and Kx) of the parameters, that is, the working equation in the linear operation in which nonlinear distortion does not occur is the following equation (9).

A0*E(t)/Ze＝K0*x(t)+[r+A0 ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ² (9)

Therefore, when equation (9) is subtracted from equation (8), a working equation including only the nonlinear component of the loudspeaker is obtained as equation (10).

Ax*E(t)/Ze＝Kx*x(t)+[(2*A0*Ax+A0 ² )/Ze]*dx(t)/dt (10)

Furthermore, when Equation (10) is subtracted from Equation (8), a working equation from which the nonlinear component of the loudspeaker is removed is obtained as Equation (11).

(A0+Ax)*E(t)/Ze-Ax*E(t)/Ze

=(K0+Kx)*x(t)+[r+(A0+Ax) ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ²

-Kx*x(t)+[(2*A0*Ax+A0 ² )/Ze]*dx(t)/dt (11)

Here, Equation (11) is expressed as Equation (12) when making the right side of Equation (11) equal to the right side of Equation (8) (the original operating equation of the speaker 16).

(A0+Ax)*E(t)/Ze-Ax*E(t)/Ze+Kx*x(t)+

[(2*A0*Ax+A0 ² )/Ze]*dx(t)/dt

=(K0+Kx)*x(t)+[r+(A0+Ax) ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ² (12)

Rearranging the left side of equation (12), equation (13) is obtained. The left side of equation (13) is the filter coefficient for eliminating the nonlinear component of the parameter.

(A0+Ax)/Ze*[E(t)-Ze/(A0+Ax)*(Ax/Ze*E(t)-

(2*A0*Ax+Ax ² )/Ze*dx(t)/dt-Kx*x Ct))]

=(K0+Kx)*x(t)+[r+(A0+Ax) ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ² (13)

Note that among the above filter coefficients, the parameters A0 and Ax with respect to the above-mentioned force coefficient B1, the parameters K0 and Kx with respect to the rigidity K, and the electrical impedance Ze are parameters unique to the connected speaker 16, and are parameters constituting a nonlinear Preset parameters of the filter coefficients of the component canceling filter 10. Furthermore, as seen from the left side of equation (13), the value of the vibration displacement x(t) is required as a parameter required for the filter coefficient of the nonlinear component canceling filter 10 . The vibration displacement x(t) is generated by a linear filter 11 described next.

On the assumption that the speaker 16 performs linear operation from the input signal, the linear filter 11 produces a vibration displacement x(t) based on preset filter coefficients. In other words, the linear filter 11 generates vibration displacement x(t) in pseudo-linear operation. As mentioned above, the operating equation in the linear operation of the loudspeaker 16 is as described in equation (9). Therefore, after obtaining the transfer function by Laplace transforming Equation (9), the following Equation (14) is obtained. The right side of equation (14) is the filter coefficient of the linear filter 11 . Note that X(s) represents the transfer function of the vibration displacement x(t), and E(s) represents the transfer function of the input signal voltage.

x(s)/E(s)＝(A0/Ze)/[K0+s*(r+A0 ² /Ze)+s ² *m] (14)

As described above, as shown in Equation (8), through the feed-forward processing of the nonlinear component removal filter 10 and the linear filter 11, the nonlinearity of the modeled force coefficient BI(x) and stiffness K(x) is eliminated. linear component. Therefore, nonlinear distortion attributable to these nonlinear components can be eliminated. In addition, the feed-forward process removes the non-linear components, allowing the speaker 16 to operate linearly. Since the nonlinear component cancellation filter 10 refers to the vibration displacement x(t) when the loudspeaker 16 operates linearly, a more efficient cancellation of distortion effects is obtained.

The feedback processing performed by the ideal filter 12, the sensor 17, the adder 14, the feedback control filter 15, and the adder 13 will be described below.

The ideal filter 12 is a filter having a transfer function F(s) of a desired output characteristic when the characteristic (hereinafter referred to as output characteristic) according to the vibration of the speaker 16 is a desired output characteristic as a filter coefficient. In other words, the ideal filter 12 is a filter that converts the frequency characteristics of an input signal into desired output characteristics. Here, a signal whose frequency characteristic is converted into a desired output characteristic refers to a desired characteristic signal f(t). The desired characteristic signal f(t) is output to the adder 14 . Note that the output characteristics of the speaker 16 include various characteristics such as vibration displacement characteristics, velocity characteristics, acceleration characteristics (sound pressure characteristics), and the like. For example, as shown in FIG. 6 , it is assumed that the sound pressure frequency characteristic (acceleration characteristic) of the actual speaker 16 is the characteristic shown in A in FIG. 6 . FIG. 6 shows desired output characteristics, which are set as filter coefficients of the ideal filter 12 . In FIG. 6, when the sound pressure frequency characteristic of the loudspeaker 16 is made into a flat characteristic curve having a widened frequency band like the characteristic shown in B, the transfer function F(s) of the characteristic shown in B can be set to be ideal. Filter coefficients for filter 12.

The sensor 17 detects the vibration of the speaker 16 and outputs a detection signal y(t) having an output characteristic of the speaker 16 . The detection signal y(t) output by the sensor 17 is appropriately amplified and output to the adder 14 . Note that the sensor 17 is, for example, a microphone, a laser displacement meter, an accelerometer, or the like. Here, the signal characteristic output to the adder 14 is the same type of output characteristic that the above-mentioned desired characteristic signal f(t) has. In other words, in the ideal filter 12, when the output characteristic of the desired characteristic signal f(t) is, for example, the vibration displacement characteristic of the speaker 16, the signal output to the adder 14 is a signal of the vibration displacement characteristic. Note that in this case, a sensor that detects the vibration of the speaker 16 and outputs its vibration displacement may be used as the sensor 17 . Alternatively, even if a sensor that outputs the velocity characteristic or acceleration characteristic of the speaker 16 is used as the sensor 17, a differentiating circuit and an integrating circuit may be appropriately provided between the sensor 17 and the adder 14 so that the signal output to the adder 14 A characteristic is converted into a vibration displacement characteristic.

Note that the sound pressure frequency characteristic of a speaker is a characteristic proportional to the acceleration characteristic. Therefore, when the characteristic of the desired characteristic signal f(t) output from the ideal filter 12 represents the acceleration characteristic of the speaker 16, and the sensor 17 is an accelerometer, and the characteristic of the signal output from the sensor 17 represents the acceleration characteristic, the distortion is eliminated The effect of the effect becomes the highest.

Hereinafter, for the sake of explanation, it is assumed that the characteristic of the detection signal y(t) output from the sensor 17 is of the same type as the desired characteristic signal f(t) output by the ideal filter 12 . In other words, consider a case where there is no need to provide a differentiating circuit and an integrating circuit between the sensor 17 and the adder 14 .

The adder 14 subtracts the detection signal y(t) output by the sensor 17 from the desired characteristic signal f(t) output by the ideal filter 12, and outputs the subtraction signal (f(t)-y( t)) as the error signal e(t). The gain and the like of the error signal e(t) are adjusted by the feedback control filter 15 , and the error signal e(t) is returned and input to the adder 13 . Then, the output signal of the nonlinear component canceling filter 10 and the error signal e(t) output from the feedback control filter 15 are added by the adder 13 and output to the speaker 16 . Note that the feedback control filter 15 is basically a filter for adjusting gain, that is, an amplifier, and as the gain becomes larger, the effect of eliminating distortion becomes larger.

Here, as mentioned above, the rigidity K of the support system ages. Also, as shown in Fig. 5, the characteristics of the stiffness K vary with the input size. In this case, the output characteristic of the speaker 16 also changes. On the other hand, the sensor 17 detects the changed output characteristic of the speaker 16, and the above error signal e(t) is a signal of difference between the detection signal y(t) output by the sensor 17 and the desired characteristic signal r(t). Then, the above slow change of the rigidity K and the change of its characteristic due to the input size are reflected in the error signal e(t). The error signal e(t) is returned and input to the adder 13 through the feedback control filter 15, thereby canceling the above-mentioned slow variation of the rigidity K and the change of its characteristic due to the magnitude of the input.

As mentioned above, the feedback processing that can be performed by the ideal filter 12, the sensor 17, the adder 14, the feedback control filter 15, and the adder 13, for a slow change in the stiffness K of the supporting system and its characteristic change due to the input size , performing robust distortion cancellation processing.

As the above-mentioned third cause of nonlinear distortion, the change in the electrical impedance characteristic of the voice coil 161 (especially the change due to heat generation) is also included in the above-mentioned error signal e(t). Then, nonlinear distortion caused by this variation can be eliminated by the above feedback processing.

During the generation of the error signal e(t), a signal f(t) with the desired output characteristic (transfer function F(s)) is used at the ideal filter 12 . The output characteristics of the actual speaker 16 can be approximated to the desired output characteristics described above by performing feedback processing on the error signal e(t).

As described above, according to the speaker device 1 of the present embodiment, most of the nonlinear distortion of the speaker can be eliminated by feedforward processing, and it is possible to respond to slow changes in rigidity of the support system and changes in its characteristics due to input magnitudes by feedback processing Perform robust distortion removal processing. Accordingly, an adaptive parameter update circuit requiring a complicated and large amount of calculation is not required, an increase in cost is prevented, and a speaker device capable of performing more stable distortion canceling processing with high feasibility can be provided.

Note that the above-described feedback control filter 15 may also have characteristics such as a low-pass filter or the like in addition to gain adjustment. For example, there are cases where the mid-frequency and high-frequency characteristics of the speaker 16 are significantly disturbed, and when the error signal e(t) is fed back as it is, there is a fear that oscillation will occur. At this time, the feedback control filter 15 is made to have the characteristics of a low-pass filter to remove intermediate frequency and high frequency components, thereby preventing oscillation. In the speaker device 1 shown in FIG. 1, the feedback control filter 15 can be omitted if oscillation caused by the error signal e(t) is not feared and gain adjustment is not required.

In the above-mentioned nonlinear component removal filter 10, the nonlinear distortion attributable to the force coefficient B1 and rigidity K of the support system is eliminated using the filter coefficients shown in Equation (13) derived from Equation (8), But it doesn't stop there. Furthermore, in Equation (8), the above-described electrical impedance characteristic Ze of voice coil 161 is reflected as a function Ze(x) of vibration displacement x, and a filter coefficient considering electrical impedance characteristic Ze can be set according to Equation (14). Therefore, in the feedforward processing performed by the nonlinear component removing filter 10 and the linear filter 11, nonlinear distortion caused by a change in the electrical impedance characteristic Ze based on the vibration displacement x(t) can be eliminated.

In addition, the above-described nonlinear component removing filter 10 refers to the vibration displacement x(t) in pseudo-linear operation generated by the linear filter 11, but it may directly refer to the output signal of the sensor 17 as shown in FIG. 7 . In other words, the linear filter 11 can be omitted by directly referring to the output of the sensor 17 . In this case, the vibration displacement x(t) is the vibration displacement x(t) of the actual speaker, and the nonlinear component elimination filter 10 can perform processing according to the vibration displacement of the actual speaker. Note that FIG. 7 is a block diagram showing an exemplary configuration of the speaker device 1 when the output signal of the sensor 17 is referred to by the nonlinear component canceling filter 10 . At this time, since the signal referenced by the nonlinear component removing filter 10 is the vibration displacement x(t), the sensor 17 may be a sensor that detects the vibration displacement characteristic of the speaker 16 . Also, even if the signal detected by the sensor 17 is a velocity characteristic or an acceleration characteristic, the vibration displacement characteristic can be appropriately obtained using a differential circuit and an integrating circuit.

(second embodiment)

Referring to Fig. 8, a speaker device 2 according to a second embodiment of the present invention will be described. FIG. 8 is a block diagram showing an exemplary configuration of a speaker device 2 according to the second embodiment. In FIG. 8, the speaker device 2 includes a nonlinear component elimination filter 10, a linear filter 11, an ideal filter 12, an adder 13, an adder 14, a feedback control filter 15, a speaker 16, a sensor 17, and a pre-filter device 20. As shown in FIG. 8 , the speaker device 2 according to the present embodiment is different from the speaker device 1 shown in FIG. 1 described above in that it additionally has a pre-stage filter 20 . The differences will be mainly described below. Since the nonlinear component canceling filter 10, the linear filter 11, the ideal filter 12, the adder 13, the adder 14, the feedback control filter 15, the speaker 16, and the sensor 17 are the same as those described in the first embodiment, therefore The same reference numerals are used and descriptions thereof will be omitted.

The front-stage filter 20 is located immediately before the nonlinear component removing filter 10 and the linear filter 11, and processes an electric signal as an input signal based on predetermined filter coefficients. The signal processed by the pre-stage filter 20 is input to the nonlinear component removing filter 10 and the linear filter 11 . Here, the filter coefficient of the front-stage filter 20 is F(s)/P(s), where the transfer function F(s) of the desired output characteristic as the filter coefficient of the ideal filter 12 is divided by the actual loudspeaker 16 The transfer function P(s) of the output characteristics in linear operation. Note that the output characteristics of the transfer function P(s) are of the same type as the desired output characteristics of the ideal filter 12 . In other words, as described in the first embodiment, for example, when the transfer function F(s) is based on the vibration displacement characteristics of the speaker 16, the transfer function P(s) is a function based on the vibration displacement characteristics of the speaker 16 when it operates linearly.

Here, the transfer function of the input signal voltage input to the pre-filter 20 is represented by E(s). At this time, the output signal of the pre-filter 20 becomes E(s)*F(s)/P(s). When the output signal is output by the speaker 16 through the nonlinear component canceling filter 10, the output signal is multiplied by the transfer function P(s) of the speaker 16, so that the output characteristic of the speaker 16 finally becomes E(s)*F(s ). In other words, the output characteristic of the speaker 16 is converted into the target characteristic F(s). At this time, the transfer function of the detection signal y(t) output by the sensor 17 becomes E(s)*F(s). Also, the input signal that becomes the transfer function E(s) is input to the ideal filter 12 . At this time, since the filter coefficient of the ideal filter 12 is F(s), the transfer function of the output signal F(t) of the ideal filter 12 becomes E(s)*F(s). In the adder 14 , the above detection signal y(t) is subtracted from the output signal f(t) from the ideal filter 12 . At this time, the transfer functions of the output signal f(t) and the detection signal y(t) are both E(s)*F(s) and the same, and the error signal e(t) becomes zero.

For example, assume that the transfer function of the loudspeaker changes from P(s) to P'(s) due to slow changes in the stiffness K of the support system, etc. At this time, the transfer function Y(s)/E(s) of the speaker device 2 shown in FIG. 8 becomes Equation (15). Note that Y(s) is obtained by performing Laplace transform on the output signal Y(t) of the speaker 16 . E(s) is obtained by performing a Laplace transform on the input signal voltage.

Y(s)/E(s)＝(P'(s)*[1+P(s)])/(P(s)*[1+P'(s)])*F(s) (15 )

It can be seen from the above equation (15) that when the transfer function P(s) of the loudspeaker 16 is constant (when P'(s)=P(s)), the right side of the equation (15) becomes F(s) . In other words, the output characteristic of the speaker 16 converges to the desired characteristic F(s).

Next, in the speaker device 1 shown in FIG. 1 without the front-stage filter 20, wherein the transfer function is P(s) when the speaker 16 works linearly, the transfer function Y of the speaker device 1 shown in FIG. 1 ( s)/E(s) becomes equation (16).

Y(s)/E(s)＝(P(s)*[1+F(s)])/[1+P(s)] (16)

It can be seen from the above equation (16) that when the transfer function P(s) of the loudspeaker 16 is constant (when P'(s)=P(s)), the right side of the equation (16) does not become F(s ). In other words, the output characteristics of the speaker 16 do not converge to the desired characteristic F(s).

If the transfer function of the speaker 16 is changed from P(s) to P'(s), the transfer function Y(s)/E(s) of the speaker device 1 shown in FIG. 1 becomes Equation (17).

Y(s)/E(s)＝(P’(s)*[1+F(s)])/[1+P’(s)] (17)

Therefore, in the speaker device 1 shown in FIG. 1, as shown in equation (16) and equation (17), the output characteristic of the speaker 16 becomes a characteristic approximated to F(s) by providing the ideal filter 12, However, no matter how the transfer function of the speaker 16 changes, its output characteristics do not converge to the desired characteristic F(s). On the other hand, in the speaker device 2 shown in FIG. 8, by providing the front-stage filter 20, the output characteristic of the speaker 16 converges to F(s) at least when the transfer function of the speaker is not changed. In other words, the front-stage filter 20 functions to enhance the convergence of the output characteristics of the speaker 16 to desired output characteristics.

As described above, by providing the front-stage filter 20, the speaker device 2 according to the present embodiment can enhance convergence to a desired output characteristic (transfer function F(s)).

Note that, similarly to the first embodiment, the above-described feedback control filter 15 may have characteristics such as a low-pass filter in addition to gain adjustment. In the speaker device 2 shown in FIG. 8, the feedback control filter 15 may be omitted if oscillation caused by the error signal e(t) is unlikely and gain adjustment is not required.

In the above-described nonlinear component removal filter 10, similarly to the first embodiment, the force coefficient attributable to the support system is removed using the filter coefficients shown in equation (13) derived from equation (8) Non-linear distortion of B1 and rigid K, but this is not limited to this. Furthermore, in Equation (8), the above-described electrical impedance characteristic Ze of voice coil 161 is reflected as a function Ze(x) of vibration displacement x, and a filter coefficient considering electrical impedance characteristic Ze can be set according to Equation (14).

FIG. 8 shows a configuration in which the input of the linear filter 11 is connected to the output of the pre-filter 20, but this is not limited thereto. Even if a configuration as shown in FIG. 9 is provided in which the input of the linear filter 11 is the same as the input of the preceding filter 20 and the ideal filter 12, the same effect as that obtained by the configuration shown in FIG. 8 can be obtained. Note that FIG. 9 is a block diagram showing an exemplary configuration in which the input of the linear filter 11 shown in FIG. 8 is changed.

Similar to in the first embodiment, the above-mentioned nonlinear component canceling filter 10 refers to the vibration displacement x(t) in pseudo-linear operation produced by the linear filter 11, but it may directly refer to the sensor 17 as shown in FIG. output signal. In other words, the linear filter 11 can be omitted by directly referring to the output of the sensor 17 . Note that FIG. 10 is a block diagram showing an exemplary configuration of the speaker device 2 when the nonlinear component canceling filter 10 refers to the output signal of the sensor 17 . At this time, since the signal referenced by the nonlinear component removing filter 10 is the vibration displacement x(t), the sensor 17 may be a sensor that detects the vibration displacement characteristic of the speaker 16 . Also, even if the signal detected by the sensor 17 is a velocity characteristic or an acceleration characteristic, the vibration displacement characteristic can be appropriately obtained using a differential circuit and an integrating circuit.

(third embodiment)

Referring to Fig. 11, a speaker device 3 according to a third embodiment of the present invention will be described. FIG. 11 is a block diagram showing an exemplary configuration of a speaker device 3 according to the third embodiment. In FIG. 11 , speaker device 3 includes nonlinear component canceling filter 10 , ideal filter 12 , adder 13 , adder 14 , feedback control filter 15 , speaker 16 , sensor 17 and prefilter 20 . The speaker device 3 according to the present embodiment is different from the speaker devices 1 and 2 shown in FIGS. , the speaker device can extend the frequency band in which the distortion canceling effect is obtained to the low frequency band.

The above differences will be mainly described below with reference to FIG. 11 . In FIG. 11 , as the speaker device 3 , an exemplary configuration is shown in which the position of the nonlinear component canceling filter 10 relative to the speaker device 2 is changed. Note that in FIG. 11 , the symbols associated with the inputs and outputs of the adders 13 and 14 are different from those shown in FIG. 10 . However, if they are allocated so that the phase relationship is the same, the same operation and the same effect can be provided even if each symbol is any one of them. Since the nonlinear component canceling filter 10, ideal filter 12, adder 13, adder 14, feedback control filter 15, speaker 16, and sensor 17 are the same as those described in the first and second embodiments, the The same reference numerals are used and descriptions thereof will be omitted.

The non-linear component removing filter 10 is located between the adder 13 and the speaker 16 . In other words, nonlinear component canceling filter 10 is located in a feedback loop formed by sensor 17 , adder 14 , feedback control filter 15 , adder 13 and speaker 16 . In this case, the units of the nonlinear component canceling filter 10 and the speaker 16 can be regarded as controlled objects in the linear two-degree-of-freedom control.

Here, as described in the first embodiment, the nonlinear component canceling filter 10 cancels the nonlinear component of the modeled stiffness K, and functions to cancel the nonlinear distortion generated by the speaker 16 . Therefore, the above-mentioned controlled object can be regarded as an object whose nonlinear distortion of the speaker 16 is canceled by the nonlinear component canceling filter 10 to some extent. By positioning such a controlled object in the feedback loop, the change in the rigidity K shown in FIG. 4 with respect to the vibration displacement x becomes smaller in the feedback loop. In other words, this means that the rigidity K does not significantly change even when the amplitude of the speaker 16 becomes large. Also, since the change in rigidity K becomes small, the change in the lowest resonance frequency f0 becomes small.

On the other hand, in the speaker device 2 shown in FIG. 10, the nonlinear component canceling filter 10 is not in the feedback loop. Therefore, in the speaker device 2 shown in FIG. 10 , the above-mentioned controlled object is the speaker 16 , not an object from which nonlinear distortion has been canceled to some extent in the feedback loop as described above.

As described above, in the case of paying attention to the processing in the feedback loop, in the speaker device 3 according to the present embodiment, the change in the lowest resonance frequency f0 of the speaker 16 becomes small compared with the speaker device 2 shown in FIG. 10 .

The above will be described more specifically below with reference to the gain characteristics G1 to G4 and the phase characteristic P of the speaker device 3 shown in FIG. 12 . FIG. 12 shows gain characteristics and phase characteristics of the speaker device 3 . Note that the gain characteristics G1 to G4 shown in Fig. 12 are open-loop transfer characteristics. A gain characteristic G1 shown by a solid line in FIG. 12 shows the sound pressure frequency characteristic of the speaker 16 , that is, a characteristic proportional to the acceleration characteristic. Gain characteristics G2 to G4 indicated by dotted lines will be described later.

From the gain characteristic G1, it can be seen that the gain is attenuated with a gradient of -12 dB/oct in the frequency band of the lowest resonance frequency f0 or lower. From the phase characteristic P shown in FIG. 12, it can be seen that the phase shifts by 90° at the lowest resonance frequency f0. At the lowest resonance frequency f0 or lower, it is seen that the phase shift approaches 180° as the frequency becomes smaller. At the lowest resonance frequency f0 or higher, it is seen that the phase shift approaches 0° as the frequency becomes larger.

Here, in the feedback control filter 15 shown in FIG. 11 , a case where the gain of the error signal e(t) input to the adder 13 is adjusted is considered. In this case, according to the magnitude of the gain adjusted by the feedback control filter 15, the gain characteristic G1 becomes the gain characteristic G2, G3, or G4 indicated by a dotted line in FIG. 12 . Note that the magnitude of the input to the speaker 16 varies with the magnitude of the gain adjusted by the feedback control filter 15 . By changing the magnitude of the speaker input, the magnitude of the amplitude of the speaker 16 is changed. Here, as described above, in the speaker device 3, even when the amplitude of the speaker 16 becomes large, the change in the lowest resonance frequency f0 is small. Therefore, each of the lowest resonance frequencies of the gain characteristics G2, G3, and G4 indicated by broken lines in FIG. 12 is a value close to F0.

Next consider evaluated values as gain margin and phase margin. The gain margin indicates how negative the gain of the open-loop transfer characteristic changes when the phase of the open-loop transfer characteristic is 180°. Note that the frequency at the 180° phase is referred to as the phase crossing frequency fpc. The phase tolerance indicates how much negative value the phase of the open-loop transfer characteristic changes relative to 180° when the gain of the open-loop transfer characteristic is 0dB. Note that the frequency at 0dB gain is called the gain crossover frequency fgc.

Here, the frequency characteristics of the feedback loop of the speaker device 2 shown in FIG. 10 are analyzed. In the feedback loop of the speaker device 2 shown in FIG. 10, since a signal representing the normal acceleration characteristic is fed back, the frequency characteristic is significantly changed, and then analysis becomes difficult. An ideal filter 12 is added as shown in FIG. 13, and an analysis of frequency characteristics is considered. In other words, the ideal filter 12 is added, and the analysis is performed in a state where the frequency characteristics do not change. FIG. 13 shows a configuration for analyzing the frequency characteristics of the speaker device 2 shown in FIG. 10 .

FIG. 14 shows sound pressure frequency characteristics, second-order distortion characteristics, and third-order distortion characteristics after changing the input size of speaker 16 of FIG. 13 . More specifically, as shown in FIG. 14 , sound pressure frequency characteristics, secondary distortion characteristics, and tertiary distortion characteristics when the input to speaker 16 is 1V, 5W, 10W, 20W, and 40W are shown. As seen from FIG. 14, as the input becomes larger, the levels of the secondary and tertiary distortions become large. This is because the stiffness becomes greater as the input becomes larger, causing the gain to rise across frequency fgc. Therefore, the lower limit frequency of the frequency band in which the distortion canceling effect is obtained is proportional to the gain crossing frequency fgc.

Referring again to FIG. 12 , the reason why the speaker device 3 can extend the frequency band in which the distortion canceling effect is obtained to the low frequency band will be described below. In FIG. 12, when the feedback control filter 15 adjusts to increase the gain, the gain characteristic G1 becomes the characteristic shown by the gain characteristic G2. At this time, the gain crossover frequency fgc2 in the gain characteristic G2 becomes a frequency lower than the gain crossover frequency fgc1. This is because, as described above, in the speaker device 3, even when the magnitude of the amplitude of the speaker 16 changes, the change in the lowest resonance frequency f0 is small. Therefore, the speaker device 3 obtains the result that the frequency band in which the effect of canceling distortion is obtained is extended to a low frequency band proportional to the gain crossover frequency fgc2.

On the other hand, as described above, in the speaker device 2 shown in FIG. 10 , the nonlinear component canceling filter 10 is not in the feedback loop. Therefore, in the speaker device 2 shown in FIG. 10, when the input of the speaker 16 becomes large, that is, when the feedback control filter 15 adjusts to increase the gain, the gain characteristic G1 becomes as shown in the gain characteristic G2'. characteristic. In other words, the value of rigidity K becomes large, and the lowest resonance frequency f0 is raised to F0'. . Furthermore, the gain crossover frequency increases to the gain crossover frequency fgc2' as the lowest resonant frequency f0 increases. Accordingly, the speaker device 2 obtains the result that the frequency band in which the effect of canceling distortion is obtained is shifted to a high frequency band proportional to the gain crossover frequency fgc2'.

Note that in FIG. 12, when the feedback control filter 15 makes adjustments to reduce the gain, the gain characteristic G1 becomes the characteristic shown by the gain characteristic G3. At this time, the gain crossover frequency fgc3 in the gain characteristic G3 becomes a frequency higher than the gain crossover frequency fgc1. In other words, when the feedback control filter 15 adjusts to reduce the gain, the gain characteristic changes from the gain characteristic G1 to the gain characteristic G3, and the gain cross frequency fgc1 rises to the gain cross frequency fgc3. When the feedback control filter 15 makes adjustments to further reduce the gain, the gain characteristic G1 becomes the characteristic shown by the gain characteristic G4. According to the gain characteristic G4, the gain value is negative over the entire frequency band. Therefore, when the gain characteristic is G4, the feedback processing is fully stabilized. However, by reducing the feedback gain, the effect of reducing distortion becomes smaller. For the speaker device 2 shown in FIG. 10 , it is true that the distortion reduction effect becomes small due to these gain characteristics G3 and G4. In the control system using the loudspeaker 16, the phase does not become 180°, and there is no phase crossing frequency fpc. Most of the situation is the same for speaker devices 1 to 3 . Since the phase never becomes 180°, the above phase margin value is always negative.

As described above, according to the speaker device 3 shown in FIG. 11 , by placing the nonlinear component canceling filter 10 in the feedback loop, the lowest resonance frequency f0 of the speaker 16 is lower than that of the speaker device 2 shown in FIG. 10 . changes became smaller. Since the variation of the lowest resonant frequency f0 of the loudspeaker 16 becomes smaller, the variation of the gain across the frequency fgc becomes smaller. Therefore, even if the input becomes large, the speaker device 3 shown in FIG. 11 can achieve the effect of canceling distortion in a lower frequency band than in the speaker device 2 shown in FIG. 10 .

Note that, for the speaker device 3 shown in FIG. 11 , as shown in FIG. 15 , a compensation filter 21 may be added immediately before the nonlinear component canceling filter 10 . FIG. 15 is a block diagram showing an exemplary configuration in which a compensation filter 21 is added to the speaker device 3 shown in FIG. 11 .

The compensation filter 21 increases the level in the low frequency band in the open-loop transfer characteristic of the speaker device 3 . In other words, the compensation filter 21 corresponds to the low-pass filter of the present invention. More specifically, the compensation filter 21 has a filter coefficient H expressed by a transfer function such as Equation (18).

H＝k*(1+1/(T*s)) (18)

Note that T=1/(2*π*fmax).

Here, K represents a gain, and fmax represents an inflection point frequency of the frequency characteristic. The inflection point frequency indicates the frequency at which the gradient of the frequency characteristic changes. For example, assume that the corner frequency is the frequency at the point where the gain changes from 0dB to 3dB. The frequency characteristic of the transfer function shown in Equation (18) becomes the characteristic shown in FIG. 16 . FIG. 16 shows the gain characteristics and phase characteristics of the compensation filter and the gain characteristics ( G5 and G6 ) and phase characteristics ( P5 and P6 ) of the speaker device 3 . According to the gain characteristic of the speaker device 3 shown in FIG. 16, the dotted line gain characteristic G5 shown in FIG. 16 is changed from the filter characteristic of the compensation filter 21 to the gain characteristic G6 shown by the solid line. Since the level in the low frequency band rises in the absence of the phase cross frequency fpc, the gain cross frequency fgc can be approximated as DC. Then, since the frequency at which the above-mentioned effect of removing distortion is lowered, the effect of removing distortion is prevented from deteriorating when the input is large, and the effect of removing distortion can be realized into a lower frequency band.

The above-mentioned corner frequency is set to a frequency at least higher than the gain crossing frequency fgc. Although the degree of equation (18) is 1, it is not limited thereto. It can be a first order or higher order transfer function as long as the gain can be reduced across frequency fgc. If the order of the equation (18) becomes high, the gradient in which the gain rises at the corner frequency or lower in the filter characteristic of the compensation filter 21 becomes steep. Then, the gain characteristic of the speaker device 3 can decrease the gain across the frequency fgc as the order of equation (18) becomes higher. However, the number of times can be appropriately designed in consideration of phase characteristics. Note that when the filter coefficient of the compensation filter 21 is the first order, the filter characteristic of the compensation filter 21 exhibits such a characteristic that the characteristic is -6dB/oct gradient slope.

Note that, to the speaker device 3 shown in FIG. 11 , it is also possible to add a high-pass filter 22 as shown in FIG. 17 . FIG. 17 is a block diagram showing an exemplary configuration in which a high-pass filter 22 is added to the speaker device 3 shown in FIG. 11 .

The high-pass filter 22 prevents in advance a signal having a frequency equal to or lower than the gain crossing frequency fgc from being input. Thus, at least the cutoff frequency needs to be equal to or higher than the gain crossover frequency fgc. Since the cut-off characteristic becomes excellent as the number of times increases, the number of times can be selected for convenience of design. When the filter coefficient of the high-pass filter 22 is one-order, the filter characteristic of the high-pass filter 22 exhibits such a characteristic that it slopes with a gradient of +6dB/oct in a frequency band equal to or lower than the above-mentioned cutoff frequency . Note that the high-pass filter 22 may have a cutoff characteristic sloped with a gradient of +6 dB/oct or higher. In this case, the signal whose frequency is equal to or lower than the gain crossing frequency fgc is further cut off, and the effect of canceling distortion is not deteriorated.

Note that, for the speaker device 3 shown in FIG. 11 , it is also possible to add a compensation filter 21 and a high-pass filter 22 as shown in FIG. 18 . FIG. 18 is a block diagram showing an exemplary configuration in which a compensation filter 21 and a high-pass filter 22 are added to the speaker device 3 shown in FIG. 11 .

Here, FIG. 19 shows the speaker apparatus 3 in FIG. 11 , the speaker apparatus 3 to which only the high-pass filter 22 is added in FIG. 17 , and the speaker apparatus to which the high-pass filter 22 and the compensation filter 21 are added in FIG. 18 The frequency characteristic analysis results of each of the 3. FIG. 19 shows analysis results when the input is 20W and 40W.

As can be seen from the figure, among the secondary and tertiary distortions shown in FIG. 19, the secondary and tertiary distortions of the speaker device 3 shown in FIG. 18 to which the high-pass filter 22 and the compensation filter 21 are added are the smallest . In other words, as shown in the analysis results, the speaker device 3 to which the high-pass filter 22 and the compensation filter 21 are added shown in FIG. 18 is the device that provides the highest distortion canceling effect.

Note that in the above description of FIG. 12, there is no phase crossing frequency fpc, and the phase margin is always negative. Here, when the gain margin and the phase margin are negative, the feedback process is unstable and oscillation occurs. Therefore, in the case that the phase crossing frequency fpc does not exist and the phase margin is always negative, how to stabilize the feedback process will become a problem. On the other hand, verify by referring to the step response. Note that, for simplicity, the analysis is performed using the feedback loop of the speaker device 2 shown in FIG. 10 . FIG. 20 shows a feedback loop of the speaker device 2 shown in FIG. 10 . Although the processing of the ideal filter 12 is a part of the feedback processing, if the processing of the ideal filter 12 is focused, the processing of the ideal filter 12 is a processing of outputting an input electric signal to the adder 14 and corresponds to feedforward processing. The ideal filter 12 is modeled based on this in the actual loudspeaker 16 as a secondary vibration system. Therefore, the processing of the ideal filter 12 is always stable, but does not affect the stability of the feedback processing described above. Therefore, the processing of the ideal filter 12 may not be considered when evaluating the stability of the feedback processing.

The resulting step response in the feedback loop shown in FIG. 20 is shown in FIGS. 21 to 23 . Figure 21 shows that in the configuration shown in Figure 20, when the stiffness Kx (non-linear component of the aforementioned stiffness K(x)) is 20000, the phase margin is -0.849°, and the gain crossover frequency fgc is 5.4 Hz Step input and its response. Fig. 22 shows the step input and its response when the stiffness Kx is 5000, the phase margin is -1.7°, and the gain crossover frequency fgc is 2.7 Hz in the configuration shown in Fig. 20 . Fig. 23 shows the step input and its response when the stiffness Kx is 1200, the phase margin is -3.46°, and the gain crossover frequency fgc is 1.3 Hz in the configuration shown in Fig. 20 .

Referring to each of the step responses shown in Figures 21 to 23, it can be seen that all step responses converge over time. Therefore, even when the phase crossing frequency fpc does not exist and the phase is negative at the gain crossing frequency fgc, oscillation does not occur and the stability is high.

Note that in FIGS. 21 to 23, since the analysis is performed using the feedback loop of the speaker device 2 shown in FIG. 10, as the rigidity Kx increases, the gain across the frequency fgc also increases. As the gain across frequency fgc increases, the frequency of the converged waveform of the step response also increases.

(fourth embodiment)

Referring to Fig. 24, a speaker device 4 according to a fourth embodiment of the present invention will be described. FIG. 24 is a block diagram showing an exemplary configuration of a speaker device 4 according to the fourth embodiment. The speaker device 4 according to the present embodiment is different from the speaker devices 1 to 3 according to the above first to third embodiments in that it further has a power amplifier 23 . In FIG. 24, for example, the speaker device 4 includes a nonlinear component canceling filter 10, a linear filter 11, an ideal filter 12, an adder 13, an adder 14, a feedback control filter 15, a speaker 16, a sensor 17, a front stage filter 20 and power amplifier 23.

In order to put the speaker devices according to the above first to third embodiments into practical use, a power amplifier for driving the speaker 16 is required. Here, when there is an element (such as the nonlinear component canceling filter 10, etc.) A power amplifier 23 is provided immediately in front of the loudspeaker 16 .

In FIG. 24 , the output signal of the adder 13 for canceling nonlinear distortion is amplified by the power amplifier 23 . For example, assume that the gain of the power amplifier 23 is 10 times, and the input voltage of the speaker device 4 shown in FIG. 24 is 1V. In this case, the output voltage of the power amplifier 23 becomes 10V. Here, when the input of the nonlinear component canceling filter 10 is 1V, the nonlinear component canceling filter 10 generates a signal in which the nonlinear distortion is canceled when the input of the speaker 16 is 1V. Thus, when the output signal of the adder 13 is amplified to 10V, there arises a problem that it does not match the magnitude of the nonlinear distortion of the speaker 16.

Therefore, the scale of each parameter constituting the filter coefficient that each element has needs to be adjusted so that the output signal amplified by the power amplifier 23 corresponds to the level of nonlinear distortion of the speaker 16 . The process of adjusting the size of each parameter is hereinafter referred to as scaling processing.

The operating principle of the speaker device 4 shown in FIG. 24 will be described below. Note that in the following description, it is assumed that the gain of the power amplifier 23 is 10 times. The operating equation of speaker 16 is expressed by equation (8) described above.

(A0+Ax)*E(t)/Ze

=(K0+Kx)*x(t)+[r+(A0+Ax) ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ² (8)

Here, since the gain of the power amplifier 23 is 10 times, each parameter is multiplied by 1/10. Thus, Equation (8) is scaled down to a 1/10 model and becomes Equation (19).

1/10*(A0+Ax)*E(t)/(1/10*Ze)

＝1/10*(K0+Kx)*x(t)+[1/10*r+{1/10(A0+Ax)} ² /(1/10*Ze)]*dx(t)/dt

+1/10*m*d ² x(t)/dt ² (19)

The above equation (19) is rearranged into equation (20).

(A0+Ax)*E(t)/0.1/Ze

=(K0+Kx)*x(t)+[r+(A0+Ax) ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ² (20)

This means that when the input voltage E is 1V, the operation is possible when a voltage of 10V is applied.

Next, from the result of the above equation (13), the non-linear component removing filter 10 generates the voltage Eff(t) as shown in the equation (21) so as to remove the non-linear component.

Eff(t)＝

[E(t)-Ze/(A0+Ax)*(Ax/Ze*E(t)-(2*A0*Ax+Ax ² )/Ze*dx(t)/dt-Kx*x(t) )] (twenty one)

Here, considering similarly to equation (19), each parameter of equation (21) can be multiplied by 1/10 to obtain an output for eliminating nonlinear distortion, which corresponds to applying 10V when the input voltage E is 1V voltage when the speaker works. Thus, Equation (21) becomes Equation (22).

Eff(t)＝

[E(t)-(1/10*Ze)/{1/10*(A0+Ax)}*{(1/10*Ax)/(1/10*Ze)*E(t)

-(2*1/10*A0*1/10*Ax+(1/10*Ax) ² )}/(1/10*Ze)*dx(t)/dt-1/10*Kx*x(t ))] (twenty two)

Furthermore, Equation (22) above is codified into Equation (23).

Eff(t)＝

[E(t)/0.1-Ze/(A0+Ax)*(Ax/Ze*E(t)/0.1-(2*A0*Ax+Ax ² )/Ze*dx(t)/dt-Kx* x(t))] (23)

The operation of the speaker 16 to which the voltage Eff(t) expressed by Equation (23) is input becomes Equation (24) derived from Equation (13) above.

(A0+Ax)/Ze*[E(t)/0.1-Ze/(A0+Ax)*(Ax/Ze*E(t)/0.1-(2*A0*Ax+Ax ² )/Ze*dx (t)/dt-Kx*x(t))]

=(K0+Kx)*x(t)+[r+(A0+Ax) ² /Ze]*dx(t)/dt+m*d ² x(t)/dt ² (24)

In other words, when the input voltage E(t) is 1V, since E(t)/0.1 is 10V, the operation and processing are the same as when the voltage is amplified to 10V by the amplifier gain, so-called scaling processing is possible.

Therefore, if the gain of the power amplifier 23 is represented by G, in the case of performing scaling processing, each parameter can be multiplied by 1/G as shown in equation (25).

Eff(t)＝

[E(t)-(1/G*Ze)/{1/G*(A0+Ax)}*{(1/G*Ax)/(1/G*Ze)*E(t)

-(2*1/G*A0*1/G*Ax+(1/G*Ax) ² )}/(1/G*Ze)*dx(t)/dt-1/G*Kx*x(t ))] (25)

Note that the front-stage filter 20 , the ideal filter 12 , and the linear filter 11 can perform the same scaling processing as the nonlinear component canceling filter 10 described above.

As described above, by performing scaling processing, it is possible to make the magnitude of the output voltage of the nonlinear component canceling filter 10 correspond to the magnitude of the input voltage of the speaker 16 output from the power amplifier 23 when the power amplifier 23 is located immediately in front of the speaker 16 . Furthermore, a feedforward processing section such as the nonlinear component canceling filter 10 can react when the voltage at which the feedforward processing section can perform internal processing is limited.

In addition, FIG. 25 shows a comparison of frequency characteristics with and without scaling processing. As shown in Figure 25, it can be seen that the level of secondary and tertiary distortion with scaling processing is smaller, and the effect of eliminating distortion is higher. This is because the feedback gain is increased by adding the power amplifier 23 to the feedback processing section, and the same effect as described for the gain characteristic G2 in FIG. 12 is obtained.

Note that, as shown in FIG. 26, the volume of the power amplifier 23 can be associated with the nonlinear component elimination filter 10, the linear filter 11, the ideal filter 12, the feedback control filter 15, and the pre-stage filter 20, and can be Volume information Vol is returned to each element. Thus, the coefficient 1/G in the above equation (25) can be adaptively changed. Note that the volume information Vol represents information on gain values.

Note that in the speaker devices 1 to 4 described in the first to fourth embodiments, a limiter 24 may be additionally provided. Thus, the speaker 16 can be prevented from being damaged due to a large input. FIG. 27 is a block diagram showing an exemplary configuration in which the limiter 24 is provided in the speaker device 1 shown in FIG. 1 . In FIG. 27, the limiter 24 limits the input signal level to be equal to or lower than the level at which the speaker 16 is damaged. Therefore, even when a large input signal is input, a signal having a level equal to or higher than the level set by the limiter 24 is not input to the speaker 16, thereby preventing the speaker 16 from being damaged. Note that the position of the limiter 24 is not limited to the position shown in FIG. 27 , for example, it may be between the input of the non-linear component elimination filter 10 and the adder 13 or between the output of the adder 13 and the input of the loudspeaker 16. . In other words, the limiter 24 may be located anywhere where the limiter 24 can limit the input to the speaker 16 .

In the speaker devices 1 to 4 described in the first to fourth embodiments, the nonlinear component canceling filter 10, the linear filter 11, the ideal filter 12, the adder 13, the adder 14, the feedback control filter 15. The pre-stage filter 20, the compensation filter 21, the high-pass filter 22, the power amplifier 23 and the limiter 24 are formed as an integrated circuit. At this time, the integrated circuit includes an output terminal for outputting an electrical signal to the speaker 16 , a first input terminal for inputting an electrical signal, and a second input terminal for inputting a detection signal of the sensor 17 . In the first to fourth embodiments described above, circuits for performing each function described above are integrated into a small package, and, for example, an audio signal processing circuit DSP (Digital Signal Processor) or the like is formed, thereby realizing the present invention . Also, the nonlinear component canceling filter 10, the linear filter 11, and the ideal filter 12 can be formed as integrated circuits, and each function can be realized by a DSP. This is effective where the processing time of the DSP adversely affects the feedback processing and the effectiveness is reduced.

Industrial Applicability

The speaker device according to the present invention can be used in a speaker device which performs signal processing to follow the parameter change of an actual speaker and can perform more stable distortion canceling processing, and can also be used in a speaker device according to the present invention For thin speakers, etc.

Claims (22)

1. speaker unit comprises:

Loud speaker, it comprises: vibrating membrane; The back-up system assembly that comprises edge and damper is used to support described vibrating membrane to allow described vibrating membrane vibration; And voice coil loudspeaker voice coil, its generation causes the actuating force of described vibrating membrane vibration;

The feed-forward process parts, be used for the signal of telecommunication that will be input to described loud speaker being carried out feed-forward process based on filter coefficient, described filter coefficient comprises that at least described filter coefficient is configured to eliminate each nonlinearity in parameters component to representing that described back-up system assembly carries out the preset parameter of modeling and expression is applied to the preset parameter that the vibration displacement characteristic with respect to the force coefficient of the vibration displacement of described vibrating membrane of described voice coil loudspeaker voice coil is carried out modeling with respect to the vibration displacement characteristic of the rigidity of the vibration displacement of described vibrating membrane; And

The feedback processing parts are used to detect described vibration vibration of membrane, and carry out feedback processing with respect to the described signal of telecommunication that will the be input to described loud speaker pair signal of telecommunication relevant with described vibration,

The wherein said feedback processing parts pair described signal of telecommunication relevant with described vibration carries out feedback processing, thereby variation and the feasible frequency characteristic relevant with the described vibration of described vibrating membrane of eliminating the vibration displacement characteristic of the rigidity of representing described back-up system assembly become expected frequency characteristic.

2. speaker unit according to claim 1, wherein

Described feedback processing parts comprise:

Ideal filter is used to receive the described signal of telecommunication that will be input to described loud speaker and also converts the frequency characteristic of the received signal of telecommunication to expected frequency characteristic;

Transducer is used to detect the described vibration of described vibrating membrane;

First adder, be used to obtain by described ideal filter conversion and represent described expected frequency characteristic the described signal of telecommunication and with by described sensor to the relevant described signal of telecommunication of described vibration poor, and the signal of telecommunication of exporting described difference is as error signal; And

Second adder is used for the described signal of telecommunication and described error signal addition with described feed-forward process parts processing, and the resulting signal of telecommunication is outputed to described loud speaker.

3. speaker unit according to claim 2, wherein

Described feed-forward process parts comprise:

Eliminate filter, be used to receive the described signal of telecommunication that will be input to described loud speaker, and handle the signal of telecommunication that is received based on described filter coefficient; And

Linear filter is used to receive the described signal of telecommunication that will be input to described loud speaker, and produces the signal of telecommunication of the vibration displacement of described vibrating membrane when being illustrated in described vibrating membrane linear oscillator, and

Described elimination filter is with reference to the described signal of telecommunication of the described vibration displacement of expression of described linear filter generation.

4. speaker unit according to claim 3 also comprises the power amplifier that is provided between described second adder and the described loud speaker, the gain that is used to amplify the described signal of telecommunication that will be input to described loud speaker,

The filter coefficient of the filter coefficient of wherein said elimination filter, the filter coefficient of described ideal filter and described linear filter is the filter coefficient that is multiplied by the inverse of the yield value that described power amplifier amplifies.

5. speaker unit according to claim 2, wherein

The described signal of telecommunication by described sensor is the signal of telecommunication of the described vibration displacement of the described vibrating membrane of expression, and

Described feed-forward process parts with reference to by described sensor to and the described signal of telecommunication of representing described vibration displacement.

6. speaker unit according to claim 2, also comprise the prime filter that is provided in described feed-forward process parts previous stage, be used to receive the described signal of telecommunication that will be input to described loud speaker, and based on handling the signal of telecommunication that is received by the filter coefficient that characteristic obtained that deducts the described loud speaker relevant with described vibration from described expected frequency characteristic.

7. speaker unit according to claim 2 also comprises amplitude limiter, is used to limit the level of the signal of telecommunication, in order to avoid be equal to or higher than the signal of telecommunication of predetermined level to described loud speaker incoming level.

8. speaker unit according to claim 2 also comprises the power amplifier that is provided between described second adder and the described loud speaker, the gain that is used to amplify the described signal of telecommunication that will be input to described loud speaker,

The filter coefficient of wherein said feed-forward process parts and the filter coefficient of described ideal filter are the filter coefficients that is multiplied by the inverse of the described yield value that described power amplifier amplifies.

9. speaker unit according to claim 1, wherein said feed-forward process parts be provided in the position before the described loud speaker and the feedback loop that is provided in to form by described feedback processing parts in.

10. speaker unit according to claim 1, wherein

Described feedback processing parts comprise:

Ideal filter is used to receive the described signal of telecommunication that will be input to described loud speaker and also converts the frequency characteristic of the received signal of telecommunication to expected frequency characteristic;

Transducer is used to detect the described vibration of described vibrating membrane;

First adder, be used to obtain by described ideal filter conversion and represent described expected frequency characteristic the described signal of telecommunication and with by described sensor to the relevant described signal of telecommunication of described vibration poor, and the signal of telecommunication of exporting described difference is as error signal; And

Second adder, the described signal of telecommunication that is used for being input to described loud speaker and described error signal adduction mutually output to described feed-forward process parts with the resulting signal of telecommunication, and

Described feed-forward process parts carry out feed-forward process to the described signal of telecommunication of described second adder output, and export the resulting signal of telecommunication to described loud speaker.

11. speaker unit according to claim 10, also comprise the low pass filter that is provided between described second adder and the described feed-forward process parts, it has filter coefficient, the characteristic that the frequency band that is used for making the signal of telecommunication gain table that will be input to described loud speaker to be shown in being equal to or less than first frequency tilts with-6dB/oct or littler gradient

Wherein said first frequency is the frequency that is equal to or higher than the gain crossover frequency that the open loop transmission characteristic of the feedback loop that is formed by described feedback processing parts represents.

12. speaker unit according to claim 10, also comprise the high pass filter that is provided in described feed-forward process parts previous stage, it has filter coefficient, the characteristic that the frequency band that is used for making the signal of telecommunication gain table that will be input to described loud speaker to be shown in being equal to or less than second frequency tilts with 6dB/oct or bigger gradient

Wherein said second frequency is the frequency that is equal to or higher than the gain crossover frequency that the open loop transmission characteristic of the feedback loop that is formed by described feedback processing parts represents.

13. speaker unit according to claim 10 also comprises:

Be provided in the low pass filter between described second adder and the described feed-forward process parts, it has filter coefficient, the characteristic that the frequency band that is used for making the signal of telecommunication gain table that will be input to described loud speaker to be shown in being equal to or less than first frequency tilts with-6dB/oct or littler gradient, and

Be provided in the high pass filter of described feed-forward process parts previous stage, it has filter coefficient, the characteristic that the frequency band that is used for making the signal of telecommunication gain table that will be input to described loud speaker to be shown in being equal to or less than second frequency tilts with 6dB/oct or bigger gradient,

Wherein said first and second frequencies are the frequencies that are equal to or higher than the gain crossover frequency that the open loop transmission characteristic of the feedback loop that is formed by described feedback processing parts represents.

14. speaker unit according to claim 10, wherein

Described feed-forward process parts comprise:

Eliminate filter, be used to receive the described signal of telecommunication, and handle the signal of telecommunication that is received based on described filter coefficient from described second adder output; And

Linear filter is used to receive from the described signal of telecommunication of described second adder output, and produces the signal of telecommunication of the vibration displacement of described vibrating membrane when being illustrated in described vibrating membrane linear oscillator, and

Described elimination filter is produce and the described signal of telecommunication that represent described vibration displacement with reference to described linear filter.

15. speaker unit according to claim 14 also comprises the power amplifier that is provided between described feed-forward process parts and the described loud speaker, the gain that is used to amplify the described signal of telecommunication that will be input to described loud speaker,

The filter coefficient of the filter coefficient of wherein said elimination filter, the filter coefficient of described ideal filter and described linear filter is the filter coefficient that is multiplied by the inverse of the yield value that described power amplifier amplifies.

16. speaker unit according to claim 10, wherein

By described sensor to the described signal of telecommunication be the signal of telecommunication of described vibration displacement of the described vibrating membrane of expression, and

Described feed-forward process parts with reference to by described sensor to and the described signal of telecommunication of representing described vibration displacement.

17. speaker unit according to claim 10, also comprise the prime filter that is provided in described second adder anterior locations, be used to receive the described signal of telecommunication that will be input to described loud speaker, and based on handling the signal of telecommunication that is received by the filter coefficient that characteristic obtained that deducts the described loud speaker relevant with described vibration from described expected frequency characteristic.

18. speaker unit according to claim 10 also comprises amplitude limiter, is used to limit the level of the signal of telecommunication, in order to avoid be equal to or higher than the signal of telecommunication of predetermined level to described loud speaker incoming level.

19. speaker unit according to claim 10 also comprises the power amplifier that is provided between described feed-forward process parts and the described loud speaker, the gain that is used to amplify the described signal of telecommunication that will be input to described loud speaker,

The filter coefficient of wherein said feed-forward process parts and the filter coefficient of described ideal filter are the filter coefficients that is multiplied by the inverse of the described yield value that described power amplifier amplifies.

20. speaker unit according to claim 1, the variation of described vibration displacement characteristic of rigidity of wherein representing described back-up system assembly is because form the slow variation of material of described back-up system assembly or the creep that forms the material of described back-up system assembly takes place.

21. speaker unit according to claim 1, the material that wherein forms described back-up system assembly is cloth or resin.

22. an integrated circuit is used to handle the signal of telecommunication that will be input to loud speaker, described loud speaker comprises: vibrating membrane; The back-up system assembly that comprises edge and damper is used to support described vibrating membrane to allow described vibrating membrane vibration; And voice coil loudspeaker voice coil, its generation causes the actuating force of described vibrating membrane vibration, and described integrated circuit comprises:

The feed-forward process parts, be used for the signal of telecommunication that will be input to described loud speaker being carried out feed-forward process based on filter coefficient, described filter coefficient comprises that at least described filter coefficient is configured to eliminate each nonlinearity in parameters component to representing that described back-up system assembly carries out the preset parameter of modeling and expression is applied to the preset parameter that the vibration displacement characteristic with respect to the force coefficient of the vibration displacement of described vibrating membrane of described voice coil loudspeaker voice coil is carried out modeling with respect to the vibration displacement characteristic of the rigidity of the vibration displacement of described vibrating membrane; And

The feedback processing parts are used to detect described vibration vibration of membrane, and carry out feedback processing with respect to the described signal of telecommunication that will the be input to described loud speaker pair signal of telecommunication relevant with described vibration,

The wherein said feedback processing parts pair described signal of telecommunication relevant with described vibration carries out feedback processing, thereby the frequency characteristic of the described vibration of the variation of the vibration displacement characteristic of the rigidity of the described back-up system assembly of elimination expression and the described vibrating membrane of feasible foundation becomes expected frequency characteristic.

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