Patent Translate Powered by EPO and Google Notice This translation is machine-generated. It cannot be guaranteed that it is intelligible, accurate, complete, reliable or fit for specific purposes. Critical decisions, such as commercially relevant or financial decisions, should not be based on machine-translation output. DESCRIPTION JPH1051881 [0001] BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a distortion removing apparatus for removing harmonic distortion and intermodulation distortion occurring in a speaker for reproducing an audio signal and realizing high fidelity reproduction for an input signal. is there. [0002] 2. Description of the Related Art In conventional speakers, when an audio signal input from a CD player or the like to a speaker through an amplifier is reproduced as sound, harmonic distortion generally called non-linear distortion and sound that reproduces the audio signal faithfully The voice of intermodulation distortion was generated. [0003] Since the music signal reproduced from the speaker is felt to be very unpleasant if these nonlinear distortions are added, reducing the distortions is a major issue in designing the speaker. In particular, second-order harmonic distortion and intermodulation distortion (hereinafter referred to simply as second-order distortion for the sake of simplicity) are perceived as unpleasant. Therefore, the present invention aims to reduce second harmonic distortion and intermodulation distortion. 10-04-2019 1 [0004] Heretofore, there have been two major methods of removing these non-linear distortions. One is a method that has been taken for many years, and is a method in which the shape, size, and material of the speaker are devised and designed so as to minimize distortion. The other one is a method that has been proposed in recent years, in which an audio electrical signal output from a sound source such as a CD player is corrected by a distortion removing device and then input to a speaker, and finally from a speaker It is a method of making the emitted sound a sound with less distortion. The inventor studied the method belonging to the latter, and completed the invention of one system (undisclosed) shown in FIG. 7 prior to the present invention. [0005] FIG. 7 is a block diagram modeling the distortion removing device 705 and the speaker 701 in the system of the above-described invention previously completed by the inventor. The speaker 701 is a system in which a system 702 of a transfer function H1 of a speaker without distortion, a system 703 of a second order transfer function H2 representing a second-order distortion of the speaker, and an output signal of these two systems are added by an adder 704 It can be expressed as [0006] The distortion removing device 705 converts the input signal x (t), which is an audio electrical signal from a CD player or the like, into digital form, and A / D converter 709 performs first onedimensional convolution calculation with a transfer function G1. A filter 706 and a second filter 707 for performing a two-dimensional convolution operation with a two-dimensional transfer function G2 are included. Furthermore, it has an adder 708 for adding the output signals of the first filter 706 and the second filter 707, and a D / A converter 709 for converting the output signal of the adder into an analog signal. [0007] In such a general speaker distortion removing device, the relationship between the input signal x (t) and the output signal y (t) is expressed by the following equation 7 when displaying in the 10-04-2019 2 frequency domain. [0008] The first term on the right side of (Equation 7) indicates a component through which the input signal x (t) passes through the first filter 706 and the system 702 of the transfer function H1 of the speaker. The second term in the braces of the second term of (Equation 7) indicates the component through which the input signal x (t) passes through the first filter 706 and the system 703 of the transfer function H2 representing the distortion of the speaker. [0009] However, the second term of Equation (7) is the sum of all combinations that satisfy m = m1 + m2 or | m1-m2 | for m representing the address of the digital signal converted to the frequency domain. The component of the input signal x (t) passing through the second filter 707 and the system 703 of the transfer function H2 representing the distortion of the speaker is negligible compared to the other terms and ignored. [0010] In order to remove the distortion component of the speaker 701, the two terms in the braces of the second term of Eq. 7 may be canceled out and the value may be zero. [0011] Next, a method of determining the transfer function G1 of the first filter 706 and the transfer function G2 of the second filter 707 will be described. [0012] First, the transfer function G1 of the first filter 706 may be determined such that the component represented by the first term of Equation 7 is equal to the desired output signal output from the 10-04-2019 3 speaker. For example, in order to make the output signal Y (m) equal to the audio electrical signal X (m), the equation 9 is derived from the equation 8. [0013] Further, when it is desired to make the audio electrical signal be a signal influenced by the transfer function H1 of the speaker with respect to the output signal Y (m), the transfer function of the first filter 706 according to (Equation 10) G1 may be set as (Equation 11). [0014] Next, the method of determining the transfer function G2 of the second filter 707 will be described. Since two terms in the braces of the second term of (Equation 7) may be canceled out, assuming that the parentheses are equal to 0, solving for G 2 results in (Equation 12). [0015] In order to determine the transfer function G 2, the transfer function H 1 of the first-order system 702 of the speaker 701, the transfer function H 2 of the second-order distortion of the second-order system 703 of the speaker 701, and The transfer function G1 of the first filter 706 determined by the above method may be substituted. [0016] However, in the conventional distortion removing device 705, when the input signal x (t) is an arbitrary audio signal which is very long in time and inputted from a CD player or the like, In the first filter 706 with M taps and the second filter 707 with M × M taps, it is necessary to process convolution operations as shown in equation 13 in real time. In this real-time processing, in particular, the amount of operation of the two-dimensional 10-04-2019 4 convolution operation by the second filter 707 represented by the second term is enormous, so the scale of the operation device is extremely large to realize this. There was a problem of becoming [0017] Equation (13) shows the relationship between the input signal x and the output signal w in the time domain. Also, n indicates the order of the discretized signals, which corresponds to time. From equation (7), the number of multiplications required in the first filter 706 is M times and the number of additions is M−1 for one sample signal of the signal x (n) obtained by converting the analog input signal x (t) into digital It is time. Similarly, according to (Equation 13), the number of multiplications required by the second filter 707 is M × M × 2 times and M × M−1 times. [0018] For example, when M is 128 (M = 128), the number of multiplications required in the second filter 706 is 32,768 and the number of additions is 16,368. That is, a very large amount of computation is required per sample of the audio signal x (n), and the device becomes very large to realize this device. [0019] The present invention has been made in view of such conventional problems, and eliminates harmonic distortion and intermodulation distortion even when general audio signals of infinite length are input to general speakers. It is an object of the present invention to realize a distortion removing device capable of [0020] According to the distortion removing apparatus of the present invention, an audio signal is divided into time domain signals of length N by frame division means, and the operation shown in equation 14 is performed in the frequency domain. After that, the signal is reconverted into a 10-04-2019 5 time domain signal, and a part of the signal is sequentially connected and output by the frame rigidity means. As a result, it is possible to significantly reduce the amount of calculation, and it is possible to make the distortion removing device extremely compact as compared with the prior art. Detailed means will be described below. [0021] First, an analog audio signal is input to A / D conversion means and converted into a digital signal. Next, the frame division means divides and takes in the digital signal outputted from the A / D conversion means with a length N while partially overlapping. The divided digital signal is subjected to fast Fourier transform by the FFT operation means to convert the time domain signal into the frequency domain signal. [0022] The first storage means stores N first coefficients in advance. Here, the first multiplier is connected to the output side of the FFT operation means and the first storage means. Therefore, the first multiplier multiplies the N first coefficients by the signal of length N that has been converted into the signal in the frequency domain by the FFT operation means. This operation is shown in the first term of the right side of (Equation 14). [0023] On the other hand, the second storage means stores N × N two-dimensional second coefficients in advance. Here, the multiplication and addition unit is connected to the output side of the FFT operation means and the second storage means. Therefore, the multiplication and addition unit performs multiplication and addition as shown in the second term of the right side of Equation 14 using the second coefficient and the signal converted into the frequency domain by the FFT operation means. [0024] 10-04-2019 6 The output signal of the first multiplier and the output signal of the multiply-add are respectively input to the adder and added. This addition is the addition of the first term and the second term on the right side of (Equation 14). [0025] The added signal is the left side of (Equation 14), which is inverse Fourier transformed by the IFFT operation means to convert the signal of the frequency domain into the signal of the time domain. [0026] A part of IFFT output signals converted to time domain signals is sequentially connected by a frame synthesis unit and output. Finally, the D / A conversion means converts the digital audio signal connected by the frame synthesis means into an analog signal and outputs it. [0027] BEST MODE FOR CARRYING OUT THE INVENTION The loudspeaker distortion removing device according to the present invention has a length N while partially overlapping an A / D conversion means to which an audio signal is input and an output signal of the A / D conversion means. Frame division means for dividing and taking in, FFT operation means for performing fast Fourier transform on signals in the time domain divided by the frame division means into signals in the frequency domain, N one-dimensional N-th one-dimensional elements in the frequency domain A first multiplication for performing multiplication of the first term of the right side of Eq. 14 using a first storage unit storing a coefficient of 1, the first coefficient, and the output signal of the FFT operation unit And the [0028] Where W (m) is the m component of the distortion removing device in the frequency domain, G 1 (m) is the first coefficient, and X (m) is the discrete signal of the input signal and then fast Fourier transform to the frequency domain M component of the converted signal, m, m1 and m2 are integer values representing the number of points on the frequency axis that are discretized, G2 10-04-2019 7 (m1, m2) is a second coefficient, and X (m1) is the input signal, The m1 component of the signal that has been discretized and then transformed to the frequency domain by Fourier transform, X (m2) represents the m2 component of the signal that has been discretized and then transformed to the frequency domain by fast Fourier transform. A second storage unit storing twodimensional N × N second coefficients, the second coefficient, and an output signal of the FFT operation unit, Multiplication to perform two-term multiplication and addition An adder, an adder for adding an output signal of the first multiplier and an output signal of the multiply-adder, and an IFFT for converting the output signal of the adder into a time domain signal by inverse fast Fourier transform Arithmetic means, frame combining means for sequentially connecting and outputting a part of output signals of the IFFT means, and D / A conversion means for converting the output signals of the frame combining means into analog audio signals Do. [0029] Also, the distortion removal apparatus for a speaker includes an A / D conversion unit to which an audio signal is input, and a frame division unit that divides an output signal of the A / D conversion unit by a length N while partially overlapping. And FFT operation means for performing fast Fourier transform on the time domain signal divided by the frame division means to convert it into a frequency domain signal, and group delay amount of linear first-order impulse response of a system for which distortion is desired to be removed Reading out the first storage means storing coefficients of a frequency domain obtained by Fourier transform of an impulse response of N-tap delayers having substantially equal delay amounts, and reading out the coefficients of the first storage means A first multiplier for multiplying the read out coefficient by the output signal of the fast Fourier transform means, and a second storage means for storing N first coefficients in the frequency domain , Said first coefficient, by using the output signal of said first multiplier, a second multiplier for multiplying the first term of equation (15), [0030] (Where W (m) is the m component of the output signal of the distortion removal apparatus in the frequency domain, G 1 (m) is the first coefficient, D 1 (m) is the m of the coefficient stored in the first storage means The component, X (m), is an integer value representing the number of m components of the signal obtained by discretizing the input signal and then transforming it into the frequency domain by fast Fourier transform, m, m1 and m2 are the number of points on the discretized frequency axis , G2 (m1, m2) is the second coefficient, D1 (m1) is the m1 component of the coefficient read from the first storage means, and X (m1) is the frequency by fast Fourier transform after discretizing the input signal The m1 component of the signal converted to the domain, D1 (m2) is the m2 component of the coefficient read out from the first storage means, and the X (m2) is the input signal discretized and converted to the frequency domain by fast Fourier transform Represents the m2 component of the signal, ) Using a second storage means for storing a second coefficient in a frequency domain of two-dimensional N × N, a second 10-04-2019 8 coefficient, and an output signal of the first multiplier, A multiplication / adder for performing multiplication and addition of the second term of the right side of 15), an adder for adding the output signal of the second multiplier and the output signal of the multiplication / adder, and an output signal of the adder IFFT operation means for performing inverse fast Fourier transform to convert into time domain signals, frame synthesis means for sequentially connecting and outputting a part of output signals of the IFFT operation means, and output signals of the frame synthesis means are analog And D / A conversion means for converting into an audio signal. [0031] The second storage device is a region defined by (Equation 16) and (Equation 17) and a region defined by (Equation 18) among the two-dimensional second coefficients represented in the frequency domain. Remember the coefficients of [0032] The multiplication and addition unit uses the second coefficient of only the defined region and the output signal of the FFT operation means to obtain [0033] The following operation is performed. [0034] An embodiment of the present invention will be described below with reference to FIGS. EXAMPLE 1 Example 1 of the present invention will be described with reference to FIG. The present embodiment provides a distortion removing apparatus which can reduce the amount of calculation significantly as compared with the prior art. The distortion removing apparatus 10 has an A / D conversion means 11 to which an audio signal is input, a relationship between N and N1 representing numerical values is N> N1, and the digital output signal of the A / D conversion means And N 1 + 1) frame dividing means 12 for dividing and taking in a length N. 10-04-2019 9 Further, FFT operation means 13 for performing fast Fourier transform on the time domain signal divided by the frame division means to convert it into a frequency domain signal, and a first for storing N first coefficients in the frequency domain. A storage unit 15 and a multiplier 14 which performs multiplication of the first term of Expression 20 using the first coefficient and the output signal of the FFT operation unit 13 are provided. [0035] Furthermore, using second storage means 17 for storing two-dimensional N × N second coefficients in the frequency domain, the second coefficients and the output signal of the FFT operation means 20) having a multiplication and addition unit 16 for performing multiplication and addition of the second term of 20) and an adder 18 for adding the output signal of the first multiplier 14 and the output signal of the multiplication and addition unit 16 . Furthermore, an IFFT operation means 19 for performing inverse fast Fourier transform on the output signal of the adder, a frame synthesis means 20 for sequentially connecting and outputting N1 to N data of the output signal of the IFFT operation means, and And D / A conversion means 21 for converting an output signal of the frame synthesis means into an analog audio signal. [0036] In such a configuration, when an analog audio signal x (t) is input from the CD player or the like to the distortion removing apparatus 10, the A / D conversion means 11 converts the audio signal x (t) into a digital signal x (n). Convert to The frame dividing means 12 outputs the output signal x (n) of a signal of length N previously inputted every time (N-N1 + 1) output signals x (n) of the A / D converting means are inputted. Divide into xi (n) and output. However, here, N> N1, and N1 corresponds to the number M of taps of the first filter 706 of the conventional distortion removing device 705 of FIG. The divided signal of length N is input to the 10-04-2019 10 fast Fourier transform means 13 and converted into a signal Xi (m) in the frequency domain. "M" is an integer value representing the number of points on the discretized frequency axis. [0037] The signal Xi (m) is subjected to the operation shown in equation 21 by the first multiplier 14, the first memory 15, the multiply-add 16, the second memory 17 and the adder 18, and the signal It becomes Wi (m). [0038] The operation of equation (21) will be described in detail. The first multiplier 14 multiplies the N first coefficients G1 (m) stored in the first storage device 15 by the output signal Xi (m) of the fast Fourier transform means 13 for each m. Match. This is the operation of the first term of (Equation 21). On the other hand, the multiplication and addition unit 16 uses the N × N second coefficients G2 (m1, m2) stored in the second storage device 17 and the signals Xi (m1) and Xi (m2). , And the second term of (Equation 21) is performed. “M1” and “m2” each represent a specific value of the integer value m. The adder 18 adds together the output signal of the multiplier 14 and the output signal of the multiplication / addition unit 16 for each m, and outputs a signal Wi (m). [0039] The output signal Wi (m) of the adder 18 is input to the IFFT operation means 19 which performs inverse fast Fourier transform, and is converted into the signal wi (n) in the time domain. The frame combining unit 20 cuts out the signal wi (N) from the signal wi (N1 + 1) from the signal wi (n), and w1 (N1 + 1),... W1 (N), w2 (N1 + 1),. (N), w3 (N1 + 1),... Here, “i” is the number of the frame divided by the frame dividing means 12. The D / A conversion means 21 converts the output signal of the frame synthesis means 20 into an analog signal w (t) and outputs it. w (t) is input to the speaker 22. [0040] 10-04-2019 11 The speaker 22 that generates harmonic distortion and intermodulation distortion with the input audio signal includes a system 23 of the linear transfer function H1 of the speaker, a system 24 of the nonlinear second-order transfer function H2 that generates distortion, and an addition And the symbol 25. The relationship between the input signal of the speaker 22 and the output sound pressure is expressed as in Expression 22 when displaying in the frequency domain. [0041] Here, when the relationship between the audio signal x (t) from a CD player or the like and the output sound pressure y (t) from the speaker is represented in the frequency domain, the signal W (m) of Eq. By substituting into (Equation 22) and erasing, it becomes (Equation 7). [0042] The first term of (Equation 7) shows a component through which the audio signal x (t) passes through the multiplier 14 and the system 23 of the transfer function H1 of the speaker 22. The first term in the braces of the second term of the right side of (Equation 7) indicates the component through which the audio signal x (t) passes through the multiplication / adder 16 and the system 23 of the transfer function H1 of the speaker 22. The second term in the braces of the second term on the right-hand side of (Equation 7) shows the component through which the audio signal x (t) passes the multiplier 14 and the system 24 of the transfer function H2 representing the distortion of the speaker 22. [0043] However, m in the second term of the right side of Equation 7 is the sum of all combinations that satisfy m = m1 + m2 or | m1-m2 |. The component through which the audio signal x (t) passes through the multiplication / adder 16 and the system 24 of the transfer function H2 representing the distortion of the speaker 22 is negligible compared to the other terms and ignored. [0044] In order to remove the distortion component of the speaker 22, it is sufficient if the two terms in the parentheses of the second term of (Equation 7) are offset and their value becomes zero. 10-04-2019 12 [0045] Next, a method of determining the coefficient G1 (m) of the first storage unit 15 and the coefficient G2 (m1, m2) of the second storage unit 17 will be described. [0046] First, the coefficient G1 (m) of the first storage unit 15 may be determined so that the component represented by the first term of (Equation 7) is equal to the desired output sound pressure output from the speaker . For example, in order to make the output signal Y (m) equal to the audio signal X (m), equation 9 is derived from equation 8. [0047] Alternatively, for the output signal Y (m), if it is desired that the audio signal be a signal affected by the transfer characteristic H1 of the linear system 23 of the speaker 22, according to (Equation 10), the first filter The transfer function G1 may be set as (Equation 11). [0048] Next, a method of determining the coefficient G2 (m1, m2) of the second storage unit 17 will be described. Since two terms in the braces on the right-hand second term of (Equation 7) should be canceled out, assuming that the braces are equal to 0, solving for G2 (m1, m2) results in (Equation 12) . [0049] To determine G 2 (m 1, m 2), the transfer function H 1 of the first-order system 23 of the speaker 22 and the transfer function H 2 of the second-order system 24 representing harmonic distortion and intermodulation distortion are given by And the coefficient G1 (m) of the first storage unit 15 determined by the above method may be substituted. 10-04-2019 13 [0050] As described above, if the coefficient G1 (m) of the first storage unit 15 and the coefficient G2 (m1, m2) of the second storage unit 17 are determined, the distortion removal device 10 can reduce harmonic distortion of the speaker 22 and Intermodulation distortion can be removed. [0051] The number of multiplications and the number of additions in the first multiplier 14 and the multiplication and addition unit 16 of the distortion removal apparatus 10 according to the first embodiment of the present invention will be described below. [0052] The number of multiplications required in the first multiplier 14 is the number shown in equation 23 per one of the audio signals x (n) converted into digital. [0053] Further, the number of multiplications required in the multiplication and addition unit 16 is the number shown in (Equation 24) per one of the audio signal x (n) converted into digital, and the number of additions is the number shown in (Equation 25) It becomes. [0054] When the required number of multiplications and additions in the system combining the first multiplier 14, the multiply-adder 16 and the adder 18 are summarized, the number of multiplications is the number shown in Eq. 26 and the number of additions is It becomes the number shown in (Equation 27). [0055] On the other hand, in the conventional method based on convolution operation in the time domain, the number of multiplications required in the convolution operation part is the number shown in (Equation 28), and the number of additions is the number shown in (Equation 29). [0056] 10-04-2019 14 For example, if N = 256 and N1 = 128 are substituted for the above (Equation 26), (Equation 27), (Equation 28) and (Equation 29), the number of multiplications is about 10 18 in the method according to the present invention. And the number of additions is 508. On the other hand, in the conventional method by convolution operation in the time domain, the number of multiplications is 32,768 and the number of additions is 16,511. As described above, according to the present invention, the number of multiplications and the number of additions become about 30 times smaller than those of the conventional method, and the effectiveness of the present invention can be understood. [0057] As mentioned above, in the distortion removal apparatus 10 of this invention, the amount of calculations is reduced significantly and the scale of an apparatus can be reduced. [0058] Embodiment 2 Embodiment 2 of the present invention will be described with reference to FIGS. 2 to 5. [0059] The configuration of the second embodiment differs from that described in the first embodiment in the following points. Of the N × N second coefficients G2 (m1, m2), the second storage means of FIG. 1 only needs to store the coefficients of the areas 201 and 202 indicated by hatching in FIG. The area 201 shown in FIG. 2 is an area defined by (Equation 30) and (Equation 31) when it is expressed by an equation. Further, when the area 202 is expressed by a mathematical expression, it becomes an area defined by (Equation 32). 10-04-2019 15 [0060] In the distortion removing apparatus 10 according to the first embodiment, the number of multiplications and additions of the second term of the right side of the equation (21) in the multiplication and addition unit 16 in the distortion removing apparatus 10 In the second embodiment, the amount of operations in the multiplication and addition unit 16 can be reduced, and the scale of the apparatus 10 can be reduced. [0061] The features of the second coefficient G2 (m1, m2) will be described. In the second coefficient G2 (m1, m2), the coefficients of the area 301 in the shaded area in FIG. 3 and the coefficients of the other areas 302 are in a line symmetry with m1 = m2 as the axis of symmetry. Also, the coefficients of the shaded area 401 in FIG. 4 and the coefficients of the non-shaded area 402 are in a conjugate relationship with (N / 2, N / 2) as the center of point symmetry. Further, the coefficients of the shaded region 403 in FIG. 4 and the coefficients of the nonshaded region 404 are in a conjugate relationship with (N / 2, N / 2) as the center of point symmetry. Further, the coefficients of the shaded area 405 in FIG. 4 and the coefficients of the non-hatched area 406 are in a conjugate relationship with (N / 2, N / 2) as the center of point symmetry. Further, considering the sampling theorem, in the product-sum operation of the second term of (Equation 21), the result of the product-sum operation is conjugate in the hatched areas 501 and 502 and the other areas 503 and 504 in FIG. In actual calculation, it is sufficient to consider only the shaded area. Therefore, taking the product set of FIG. 3, FIG. 4 and FIG. 5 results in regions 201 and 202 of 10-04-2019 16 hatched portions in FIG. 2, and when performing the calculation of the second term of the right side of Eq. The product-sum operation may be performed on the areas 201 and 202 in the hatched portion of FIG. [0062] Thus, the number of multiplications in the multiplication and addition unit 16 required for one sample of the digitally converted audio signal is the number shown in (Equation 34), and the number of additions is the number shown in (Equation 35) It becomes. [0063] For example, when N = 256 and N1 = 128 are substituted into the equations (34) and (35), the number of multiplications is about 161 and the number of additions is about 79. Thus, the size of the distortion removing device 10 can be reduced. [0064] EXAMPLE 3 Example 3 of the present invention will be described with reference to FIG. [0065] In the configuration of the third embodiment of FIG. 6, the difference from the first embodiment is that the second multiplier 46 is provided downstream of the fast Fourier transform means 33 and the first storage means 47 storing the coefficient D1 (m) is used. This is provided in the second multiplier 46. [0066] The impulse response h1 (t) of the speaker 42 generally has a characteristic including a group delay with respect to the input signal x (t). The transfer function H1 of the first-order system 43 of the speaker 42 is obtained by Fourier transform of the impulse response h1. However, when there is a group delay in the impulse response h1 and the first coefficient G1 (m) is determined by (Eq. 9), a general audio signal that changes randomly in time is input to the 10-04-2019 17 distortion removal device 30 In some cases, the causality of the input / output characteristics of the entire system including the distortion removing device 30 and the speaker 42 may be violated, and the distortion may not be removed as desired. [0067] In the present invention, in order to solve this problem, as shown in FIG. 6, the second multiplier 46 and the first storage means 47 are provided. [0068] In the above configuration, first, the first storage means 47 is a coefficient D1 (m) obtained by Fourier-transforming the impulse response characteristic of an N-tap delayer having a delay action substantially equal to the group delay amount τ of the impulse response h1. Store [0069] The second multiplier 46 multiplies the signal X (m) by the coefficient D1 (m) read from the first storage means 47 for m, and outputs the result. As a result, the component passing through the first multiplier 34 is subjected to a delay corresponding to the group delay amount of the transfer characteristic of the primary system 43 of the speaker 42. Further, for the component passing through the multiplication and addition unit 36, the second multiplier 46 multiplies the signal X (m1) by the coefficient D1 (m1) read from the first storage means 47, The coefficient D1 (m2) read from the first storage means 47 is multiplied with X (m2) and output. As a result, an output is obtained with a delay corresponding to the group delay amount of the harmonic distortion and intermodulation distortion of the speaker 42. As a result, the relationship between the signal X (m) and the signal W (m) by the first multiplier 34, the second multiplier 46, the multiplication and addition unit 36, and the adder 38 is expressed by Equation 36. 10-04-2019 18 [0070] By the above operation, even when a general audio signal having no periodicity is input to the distortion removing apparatus 30, it is possible to remove the distortion of the speaker. [0071] In addition, in order to acquire the effect similar to a present Example, it is possible also with the following structures. First, the first storage means 47 and the second multiplier 46 are not provided, and the configuration is the same as that of the first embodiment or the second embodiment shown in FIG. The second storage unit 35 includes the coefficient D1 (m) stored in the first storage unit 47 and the coefficient G1 (m) stored in the second storage unit 35 in the third embodiment. , And m are stored in advance as data G1 (m) D1 (m) as a result of multiplication. The third storage unit 37 includes the coefficient D1 (m1) stored in the first storage unit 47 and the coefficient G2 (m1, m2) stored in the third storage unit 37 in the third embodiment. Are stored for m1 and m2, respectively, and data G2 (m1, m2) D1 (m1) D1 (m2) as a result is stored in advance. [0072] With the above configuration, the first multiplier 34 multiplies the signal X (m) by the data G1 (m) D1 (m) read from the second storage means 35 for each m and outputs the result. Do. As a result, the component passing through the first multiplier 34 is subjected to a delay corresponding to the group delay amount of the impulse response h1 of the primary system 43 of the speaker. The multiplication and addition unit 36 multiplies the signal X (m) by the data G2 (m1, m2) D1 (m1) D1 (m2) read from the third storage means 37 for each of m1 and m2. Output. Here, the relationship between m, m1 and m2 satisfies m = m1 + m2 or m = | m1-m2 |, as in the first embodiment or the second embodiment. [0073] As a result, the relationship between the signal X (m) and the signal W (m) by the first multiplier 34, the multiplication and addition unit 36, and the adder 38 becomes as shown in Expression 36. 10-04-2019 19 [0074] By the above operation, even when a general audio signal having no periodicity is input to the distortion removing device 42, it is possible to remove the distortion of the speaker. [0075] According to the present invention, an A / D-converted audio signal is divided into predetermined lengths, subjected to fast Fourier transformation, and converted into a signal in the frequency domain. Then, by performing one-dimensional and two-dimensional convolution operations in the distortion removing device with the multiplier and the multiplier / adder in the frequency domain, calculation processing becomes possible with a small number of multiplications and additions, and the speaker of the speaker A distortion removing device can be realized. [0076] Further, in addition to the above effects, the impulse response characteristic of the delay device having a delay action substantially equivalent to the group delay amount of the impulse response of the speaker is provided by further providing a multiplier in front of the multiplier and the multiplier / adder respectively. By multiplying the converted coefficient D1 (m), a desired distortion removal effect can be obtained even when the transfer characteristic of the speaker includes a group delay amount. [0077] Furthermore, in the frequency domain, the number of operations can be further reduced by performing multiplication and addition using coefficients of a two-dimensional predetermined domain and the fast Fourier transform output. [0078] Brief description of the drawings [0079] 10-04-2019 20 1 is a block diagram showing the configuration of the distortion removing apparatus for the speaker and the speaker in the first embodiment of the present invention. [0080] In the second embodiment of the present invention, a diagram showing the area of the coefficient to be stored in the second storage device [0081] 3 is a diagram showing the symmetry of the coefficients to be stored in the second storage device in the second embodiment of the present invention. [0082] FIG. 4 is a diagram showing the conjugate of coefficients to be stored in the second storage device in the second embodiment of the present invention. [0083] In the second embodiment of the present invention, among the coefficients to be stored in the second storage device, a diagram showing the effective range considered by the sampling theorem [0084] 6 is a block diagram showing the configuration of the distortion removing device for the speaker and the speaker in the third embodiment of the present invention. [0085] Fig. 7 A block diagram showing the configuration of the conventional speaker distortion removing device and the speaker [0086] Explanation of sign [0087] 10, 30 distortion removing unit 11, 31 A / D converting means 12, 32 frame dividing means 13, 10-04-2019 21 33 fast Fourier transforming means 14, 34 first multiplier 15, 47 first memory means 16, 36 power adder 17 , 35 second storage means 18, 38 adder 19, 39 inverse fast Fourier transform means 20, 40 frame synthesizer 21, 41 D / A conversion means 22, 42 speakers 23, 43 linear transfer function H1 system 24, 44 system 25 45 second-order transfer function H 2 of nonlinear distortion Adder 46 first delay unit 37 third storage unit 201 third embodiment of the area of coefficients to be stored in the second storage unit 1202 of the second embodiment In 2, the area 2 of the coefficient to be stored in the second storage means 10-04-2019 22

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