Signal processing apparatus and method, and program

Abstract

The present technology relates to a signal processing device and method and a program for reducing Doppler distortion. The signal processing device is provided with: a displacement predicting unit for predicting the displacement of a speaker diaphragm on the basis of an audio signal, when reproducing sound with a single speaker on the basis of an audio signal in which a high-band signal and a low-band signal are mixed; and a correcting unit for performing temporal-direction correction on the audio signal by an interpolating process using three or more samples of the audio signal, on the basis of a correction time obtained on the basis of the predicted displacement and the speed of sound. This technology may be applied to audio reproduction systems.

Description

[Technical field]

[0001] The present technology relates to a signal processing device, a method, and a program, and in particular to a signal processing device, a method, and a program that are capable of reducing Doppler distortion. [Background technology]

[0002] For example, when playing music through speakers, low-frequency signals can affect high-frequency signals, causing the sound image to become blurred or to sound shaky.

[0003] One of the causes of this phenomenon is Doppler distortion, which occurs when a low-frequency signal causes the speaker's diaphragm to vibrate back and forth, changing the position of the sound source of the signal emitted from the diaphragm as the diaphragm moves back and forth. This is particularly noticeable in full-range speakers that output sound from a single diaphragm from low to high frequencies.

[0004] Therefore, a technique has been proposed in which a clock oscillator is controlled by a signal that has been integrated twice, and the delay time of the signal is changed by a variable delay device, thereby canceling out the Doppler distortion (see, for example, Patent Document 1).

[0005] Also, a technology has been proposed in which speaker displacement is linearly predicted using parameters for a displacement of 0 mm in digital signal processing to correct nonlinear speaker distortion (see, for example, Patent Document 2). In this technology, Doppler distortion is corrected using linear prediction of the displacement used to correct the speaker nonlinear distortion. [Prior art documents] [Patent documents]

[0006] [Patent Document 1] Patent No. 1556673 [Patent Document 2] U.S. Pat. No. 5,438,625 Summary of the Invention [Problem to be solved by the invention]

[0007] However, with the above-mentioned techniques, it is difficult to sufficiently reduce Doppler distortion.

[0008] For example, the technology described in Patent Document 1 performs a simple double integration as a method of calculating the movement (displacement) of a speaker's diaphragm, but the movement calculated by integration often differs from the actual displacement movement of the speaker, and can actually increase distortion.

[0009] In addition, in the technology described in Patent Document 2, phase modulation is performed by controlling the delay time as a method of correcting Doppler distortion, but linear interpolation is used to calculate data between sample intervals in controlling the delay time of the discrete signal.

[0010] In particular, Doppler distortion increases at 6 dB / Oct as the frequency of high-frequency signals increases, but linear interpolation can cause large errors, which can result in new distortion due to the errors.In addition, no consideration is given to time correction when the displacement of the speaker diaphragm is large and exceeds one sampling interval.

[0011] The present technology has been made in consideration of such circumstances, and makes it possible to reduce Doppler distortion. [Means for solving the problem]

[0012] A signal processing device according to one aspect of the present technology includes a displacement prediction unit that predicts, based on an audio signal, a displacement of a diaphragm of a speaker when sound is reproduced through one speaker based on the audio signal obtained by mixing a high-frequency signal and a low-frequency signal, and a correction unit that performs time-direction correction on the audio signal by interpolation processing using three or more samples of the audio signal, based on a correction time obtained based on the displacement obtained by the prediction and the speed of sound.

[0013] A signal processing method or program according to one aspect of the present technology includes a step of predicting, based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed, a displacement of a diaphragm of a speaker when sound is reproduced through one speaker based on the audio signal, and performing a time-direction correction on the audio signal by an interpolation process using three or more samples of the audio signal based on a correction time obtained based on the displacement obtained by the prediction and the speed of sound.

[0014] In one aspect of the present technology, when sound is reproduced through one speaker based on an audio signal in which high-frequency signals and low-frequency signals are mixed, the displacement of the diaphragm of the speaker is predicted based on the audio signal, and the audio signal is corrected in the time direction by an interpolation process using three or more samples of the audio signal based on the displacement obtained by the prediction and a correction time obtained based on the speed of sound. [Brief description of the drawings]

[0015] [Figure 1] FIG. 1 is a diagram illustrating Doppler distortion. [Diagram 2] FIG. 1 is a diagram illustrating Doppler distortion. [Diagram 3] FIG. 1 is a diagram illustrating Doppler distortion. [Figure 4] FIG. 1 is a diagram illustrating an example of the configuration of an audio reproduction system. [Diagram 5] FIG. 11 is a diagram illustrating a process flow during Doppler distortion correction. [Figure 6] FIG. 2 is a diagram illustrating an example of an equivalent circuit of a speaker. [Figure 7] FIG. 1 is a diagram illustrating an example of the configuration of a third-order IIR filter. [Figure 8] FIG. 13 is a diagram showing the characteristics of the force coefficient with respect to the displacement of a speaker. [Figure 9] FIG. 13 is a diagram showing the characteristics of mechanical compliance with respect to the displacement of a speaker. [Figure 10]FIG. 13 is a diagram showing the characteristics of inductance with respect to the displacement of a speaker. [Figure 11] FIG. 1 is a diagram illustrating an example of the configuration of a third-order IIR filter. [Figure 12] FIG. 13 is a diagram showing predicted displacement results and actual measured values ​​when nonlinear prediction is performed. [Figure 13] FIG. 13 is a diagram showing predicted results and actual measured values ​​of displacement when linear prediction is performed. [Figure 14] FIG. 11 is a diagram illustrating Doppler distortion correction. [Figure 15] 11A and 11B are diagrams illustrating the effect of Doppler distortion correction. [Figure 16] FIG. 4 is a diagram illustrating an example of the configuration of a Doppler distortion correction unit. [Figure 17] 11 is a flowchart illustrating a playback process. [Figure 18] FIG. 2 is a diagram illustrating an example of an equivalent circuit of a speaker. [Figure 19] FIG. 2 is a diagram illustrating an example of an equivalent circuit of a speaker. [Figure 20] FIG. 1 is a diagram illustrating an example of the configuration of an audio reproduction system. [Figure 21] FIG. 1 illustrates an example of the configuration of a computer. DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0016] Hereinafter, embodiments to which the present technology is applied will be described with reference to the drawings.

[0017] First Embodiment About this technology This technology reduces Doppler distortion by shifting the audio signal in the time direction through interpolation processing using a polynomial of degree 2 or higher. This technology also improves the prediction accuracy of the actual diaphragm movement by nonlinearly predicting the displacement of the speaker diaphragm, thereby further reducing Doppler distortion.

[0018] When playing music or other sounds through speakers, low-frequency signals can affect high-frequency signals, causing the sound image to become blurred or to sound shaky. One of the causes of this phenomenon is Doppler distortion.

[0019] Doppler distortion occurs when, as shown in FIG. 1, a low-frequency signal causes a diaphragm D11 of a speaker to vibrate back and forth, changing the sound source position of a signal radiated from the diaphragm D11.

[0020] Specifically, for example, when the diaphragm D11 moves forward, that is, toward the listening point P11, as indicated by the arrow Q11 in Fig. 1, the sound source position, that is, the position where the sound wave is generated, moves forward, and the phase of the sound (signal) output by the diaphragm D11 advances. As a result, the wavelength of the sound output from the diaphragm D11 becomes shorter.

[0021] Conversely, when the diaphragm D11 moves backward, that is, in the opposite direction to the listening point P11, as shown by the arrow Q12, the sound source position moves backward, and the phase of the sound (signal) output by the diaphragm D11 is delayed. As a result, the wavelength of the sound output from the diaphragm D11 becomes longer.

[0022] In this way, when a high-frequency signal (sound) is output from the diaphragm D11 in a state in which the diaphragm D11 moves back and forth due to a low-frequency signal, the wavelength of the sound changes.

[0023] This phenomenon is called Doppler distortion, and it occurs most noticeably in full-range speakers that use a single diaphragm to output sound from low to high frequencies.

[0024] From the usual 2-channel stereo playback to 5.1-channel surround, sound AR (Augmented Reality) and VR (Virtual Reality) using multiple speakers, and wave field synthesis often use full-range speakers because the speakers are treated as ideal point sound sources.

[0025] If Doppler distortion occurs in a speaker, it will affect the position and size of the intended sound source and the sound source that is actually reproduced.

[0026] Doppler distortion occurs, for example, when a low-frequency signal and a high-frequency signal are reproduced simultaneously, as shown in FIG.

[0027] That is, as mentioned above, the low-frequency signal causes the speaker diaphragm to vibrate back and forth, which changes the sound source position of the high-frequency signal, and this changes the arrival time of the sound to the listening point. This causes the wavelength of the high-frequency signal (sound) to become shorter or longer, resulting in signal distortion.

[0028] For example, when low-frequency and high-frequency signals are output simultaneously when Doppler distortion occurs, they are viewed on the frequency axis as shown in Figure 3. In Figure 3, the vertical axis indicates the amplitude of the signal, and the horizontal axis indicates the frequency.

[0029] In this example, the frequency f 1 The component of represents the low-frequency signal component, and the frequency f 2 The component indicates a high-frequency signal component. In particular, here, both the low-frequency signal and the high-frequency signal are sinusoidal signals.

[0030] In this example, low-frequency and high-frequency signals are simultaneously output from the speaker, causing Doppler distortion. 2 -f 1 ) and frequency (f 2 +f 1 ) are signal components generated by Doppler distortion.

[0031] As a method for reducing the Doppler distortion described above, a method is considered in which the forward and backward movement (displacement) of the speaker diaphragm is predicted, and the predicted displacement is used to control the delay time so that it is the opposite of the forward and backward movement of the speaker diaphragm. That is, the delay time is controlled by delaying the timing of the signal output (reproduction) by a time according to the predicted displacement of the diaphragm.

[0032] In this way, the time it takes for sound to reach the listening point, which changes due to the forward and backward movement of the speaker diaphragm, can be controlled to be uniform, thereby reducing Doppler distortion.

[0033] From the above, to cancel out Doppler distortion, the movement of the speaker's diaphragm is obtained by predicting or actually measuring it, and a time correction is performed on the signal in the opposite direction by the change in the arrival time of the sound (signal) due to that movement.

[0034] However, with currently proposed techniques, it is difficult to sufficiently reduce Doppler distortion.

[0035] Also, a method has been proposed to reduce Doppler distortion by modifying the shape of the speaker diaphragm. For example, a method has been proposed in which the shape of the diaphragm is made noncircular, such as an asymmetric ellipse, to radiate high-frequency signals nonuniformly from the diaphragm and disperse phase modulation to reduce Doppler distortion. However, even with such a method, the effect of improving Doppler distortion is small and cannot be considered sufficient.

[0036] Therefore, this technology uses nonlinear prediction to predict the movement (displacement) of the speaker diaphragm with higher accuracy, and also reduces Doppler distortion by time-correcting the audio signal through interpolation processing using polynomials of the second degree or higher.

[0037] For example, by performing nonlinear prediction, it is possible to predict the speaker displacement with higher accuracy than by performing linear prediction. Also, by performing interpolation processing with a polynomial of degree 2 or higher, it is possible to perform interpolation with higher accuracy than by performing linear interpolation at two points. Therefore, it is possible to further reduce Doppler distortion.

[0038] Example of an audio playback system configuration FIG. 4 is a diagram showing an example of the configuration of an embodiment of an audio reproduction system to which the present technology is applied.

[0039] The audio reproduction system shown in FIG. 4 includes a signal processing device 11, an amplifier section 12, and a speaker 13.

[0040] The signal processing device 11 performs correction on an audio signal of a content to be reproduced or the like in order to reduce Doppler distortion, and supplies the resulting corrected audio signal to the amplifier unit 12 .

[0041] Hereinafter, the audio signal input to the signal processing device 11, i.e., the source signal of the sound to be reproduced, will also be referred to as the input audio signal. In addition, hereinafter, the correction for reducing Doppler distortion will also be referred to as Doppler distortion correction.

[0042] The input audio signal input to the signal processing device 11 is an audio signal containing high-frequency components and low-frequency components, that is, an audio signal in which a high-frequency signal and a low-frequency signal are mixed.

[0043] The amplifier 12 amplifies the corrected audio signal supplied from the signal processing device 11 by an amplifier gain that is a predetermined output voltage, and supplies the amplified corrected audio signal to a speaker 13 to drive it.

[0044] The speaker 13 is, for example, a full-range speaker that outputs sounds in a frequency band from low to high. Note that since Doppler distortion occurs in speakers other than full-range speakers, the speaker 13 is not limited to a full-range speaker and may be any speaker.

[0045] The speaker 13 is driven based on the corrected audio signal supplied from the amplifier 12 to vibrate a diaphragm, thereby outputting a sound based on the corrected audio signal.

[0046] The signal processing device 11 also includes a speaker displacement prediction unit 21 and a Doppler distortion correction unit 22 .

[0047] The speaker displacement prediction unit 21 predicts the displacement of the speaker 13, which is the target for Doppler distortion correction, more specifically, the displacement of the diaphragm of the speaker 13, based on the supplied input audio signal, and supplies the prediction result to the Doppler distortion correction unit 22.

[0048] That is, in the speaker displacement prediction unit 21, the displacement of the diaphragm of the speaker 13 when sound is reproduced from one speaker 13 based on the input audio signal is obtained by nonlinear prediction based on the input audio signal. In particular, in the speaker displacement prediction unit 21, nonlinear prediction is performed using polynomial approximation (approximate polynomial) to obtain the displacement of the speaker 13.

[0049] The speaker displacement prediction unit 21 includes an amplifier unit 31 and a filter unit 32 .

[0050] The amplifier unit 31 amplifies the input audio signal supplied thereto by the output voltage (amplifier gain) of the amplifier unit 12 , and supplies the amplified signal to the filter unit 32 .

[0051] The filter unit 32 is composed of, for example, a third-order IIR (Infinite Impulse Response) filter, and performs nonlinear prediction by filtering the input audio signal supplied from the amplifier unit 31, and supplies the displacement obtained as the prediction result to the Doppler distortion correction unit 22.

[0052] The Doppler distortion correction unit 22 performs Doppler distortion correction on the input audio signal based on the prediction result supplied from the filter unit 32 of the speaker displacement prediction unit 21, and supplies the resulting corrected audio signal to the amplification unit 12.

[0053] In such a signal processing device 11, a corrected audio signal is generated by carrying out the process roughly shown in FIG.

[0054] That is, first, gain adjustment is performed by multiplying the input audio signal (source signal) by an amplifier gain in the amplifier unit 31. This amplifier gain is a gain value used for amplification in the amplifier unit 12, that is, for gain adjustment.

[0055] Next, in the filter unit 32, filtering is performed on the gain-adjusted input audio signal using a filter such as a third-order IIR filter.

[0056] This filtering process is a nonlinear displacement prediction process that predicts the displacement of the diaphragm of the speaker 13, and the prediction result obtained by such displacement prediction process is supplied to the Doppler distortion correction unit 22. For example, as the predicted result of the displacement of the diaphragm, a distance indicating the magnitude of the change in the position of the diaphragm, such as a displacement x [mm], is obtained.

[0057] In the Doppler distortion correction unit 22, the displacement x [mm] supplied as the prediction result is converted (transformed) into a correction time d = x / c [s] equivalent to the displacement x [mm] based on the sound speed c [m / s]. This correction time d indicates the delay time by which the input audio signal is delayed.

[0058] For example, when the diaphragm of the speaker 13 moves forward, that is, toward the listening point, the displacement x [mm] is a positive value. In such a case, the correction time d is increased (becomes a positive value) to delay the timing of the sound output from the speaker 13.

[0059] Conversely, when the diaphragm of the speaker 13 moves backward, that is, toward the opposite side of the listening point, the displacement x [mm] becomes a negative value. In such a case, the timing of the sound output from the speaker 13 is advanced, so the correction time d decreases (becomes a negative value).

[0060] In addition, the Doppler distortion correction unit 22 converts the correction time d [s] into a sample unit time equivalent to the displacement x [mm], that is, the number of correction samples d×Fs [samples], based on the sampling frequency Fs of the input audio signal.

[0061] The number of correction samples thus obtained indicates the amount of correction by which the output timing of the input audio signal is delayed or advanced in the time direction in order to correct the Doppler distortion. In particular, the number of correction samples here includes a value after the decimal point.

[0062] Furthermore, the Doppler distortion correction unit 22 performs an interpolation process based on the number of correction samples and the input audio signal to shift the input audio signal in the time direction by the number of correction samples (correction amount), i.e., a delay time correction process, and a corrected audio signal is generated.

[0063] At this time, as a delay time correction process for the input audio signal, instead of linear interpolation between two decimal points, interpolation processing is performed using a polynomial of second order or higher, such as Lagrange interpolation of second order or higher using at least three points, i.e., three or more samples of the input audio signal.

[0064] By such an interpolation process using a polynomial of degree 2 or higher, the sample values ​​of the input audio signal are corrected, and as a result, a delay time correction process is realized in which the input audio signal is shifted in the time direction by the correction sample number.

[0065] In the Doppler distortion correction unit 22 where such an interpolation process is performed, a delay time offset is prepared taking into consideration the amount of displacement of the diaphragm of the speaker 13 moving back and forth and the sampling frequency of the input audio signal. This offset is the number of delay samples that delays the output timing of the corrected audio signal as a whole, regardless of the amount of Doppler distortion correction.

[0066] In this manner, Doppler distortion correction is performed in the signal processing device 11. Such Doppler distortion correction corresponds to phase modulation of the input audio signal.

[0067] <Speaker Displacement Prediction> Next, the prediction of the displacement of the speaker 13 in the speaker displacement prediction unit 21 and the Doppler distortion correction in the Doppler distortion correction unit 22 will be described in more detail.

[0068] The filter unit 32 predicts the displacement of the speaker 13 when an input audio signal is input based on an equivalent model, i.e., an equivalent circuit, of the speaker 13. That is, by making the equivalent circuit of the speaker 13 into a digital filter, the displacement prediction of the speaker 13 is realized.

[0069] For example, when the speaker 13 is a sealed speaker, the equivalent circuit of the speaker 13 is as shown in FIG.

[0070] In the example of FIG. 6, the circuit on the left side of the drawing shows the equivalent circuit of the electrical system, and the circuit on the right side of the drawing shows the equivalent circuit of the mechanical system.

[0071] Moreover, each character in FIG. 6 indicates each parameter called a TS parameter.

[0072] That is, Re indicates the direct current resistance (DCR) of the voice coil, Le indicates the inductance of the voice coil, and BL indicates the force coefficient, i.e., the BL value. The force coefficient BL is calculated by multiplying the magnetic flux density in the voice coil and magnetic circuit by the coil length of the voice coil.

[0073] Moreover, Mms indicates the equivalent mass of the vibration system, which is the mass of the diaphragm and voice coil of the speaker 13.

[0074] Cms indicates the mechanical compliance, which is an index showing the softness of the unit's suspension, Rms indicates the mechanical resistance of the unit's suspension, and Cmb indicates the compliance provided by the suspension of speaker 13, i.e., the sealed box of a sealed speaker.

[0075] The displacement prediction of the speaker 13 using these TS parameters will be described below.

[0076] The velocity v(s) of the speaker diaphragm can be expressed by the following equation (1) using the above-mentioned TS parameters.

[0077]

number

[0078] In addition, the displacement X(s) of the speaker diaphragm is the integral of the velocity v(s), and can be expressed by the following equation (2).

[0079]

number

[0080] Therefore, from the above equations (1) and (2), the displacement X(s) can be expressed by the following equation (3) using the TS parameters.

[0081]

number

[0082] Such a displacement X(s) is an analog transfer function. This displacement X(s) is transformed by bilinear Z transformation (s=(1-Z -1 ) / (1+Z -1 )) or the like, and the coefficients of the digital filter are calculated, and the displacement X(s), i.e., the analog transfer function, can be expressed as a third-order IIR filter as shown in FIG.

[0083] In the example of FIG. 7, the third-order IIR filter has amplifiers 61-1 to 61-4, delays 62-1 to 62-3, an adder 63, delays 64-1 to 64-3, and amplifiers 65-1 to 65-3.

[0084] In this example, a signal to be processed is supplied to an amplifier 61-1 and a delay unit 62-1.

[0085] The amplifier 61-1 multiplies the supplied signal by a coefficient a0 to amplify the signal, and supplies the signal to the adder 63. In addition, the delay unit 62-1 delays the supplied signal, and supplies the delayed signal to the delay unit 62-2 and the amplifier 61-2.

[0086] The delay unit 62-2 delays the signal supplied from the delay unit 62-1 and supplies the delayed signal to the delay unit 62-3 and the amplifier unit 61-3, and the delay unit 62-3 delays the signal supplied from the delay unit 62-2 and supplies the delayed signal to the amplifier unit 61-4.

[0087] The amplifiers 61-2 to 61-4 multiply the signals supplied from the delay units 62-1 to 62-3 by coefficients a1 to a3 to amplify the signals, and supply the amplified signals to the adder 63.

[0088] In the following, when there is no need to particularly distinguish between the amplifiers 61-1 to 61-4, they will also be referred to simply as amplifiers 61. In the following, when there is no need to particularly distinguish between the delay units 62-1 to 62-3, they will also be referred to simply as delay units 62.

[0089] The adder 63 adds the signals supplied from the amplifiers 61-1 to 61-4 and the amplifiers 65-1 to 65-3, and supplies the signal obtained by the addition to the subsequent stage as the output of a third-order IIR filter, and also supplies the signal to the delay unit 64-1. The output of the adder 63 indicates the displacement of the speaker.

[0090] The delay unit 64-1 delays the signal supplied from the adder unit 63 and supplies the signal to the delay unit 64-2 and the amplifier unit 65-1, and the amplifier unit 65-1 amplifies the signal supplied from the delay unit 64-1 by multiplying the signal by a coefficient b1 and supplies the signal to the adder unit 63.

[0091] The delay unit 64-2 delays the signal supplied from the delay unit 64-1 and supplies the delayed signal to the delay unit 64-3 and the amplifier unit 65-2, and the delay unit 64-3 delays the signal supplied from the delay unit 64-2 and supplies the delayed signal to the amplifier unit 65-3.

[0092] The amplifiers 65-2 and 65-3 multiply the signals supplied from the delay units 64-2 and 64-3 by coefficients b2 and b3 to amplify the signals, and supply the amplified signals to the adder 63.

[0093] In the following, when there is no need to particularly distinguish between the delay units 64-1 to 64-3, they will also be referred to simply as delay units 64. In the following, when there is no need to particularly distinguish between the amplifier units 65-1 to 65-3, they will also be referred to simply as amplifier units 65.

[0094] For example, the coefficients a0 to a3 and the coefficients b1 to b3 used in the third-order IIR filter shown in Fig. 7 can be calculated by bilinear transformation. That is, these coefficients can be calculated based on the TS parameters.

[0095] Among the TS parameters of the equivalent circuit of the speaker 13, the force coefficient BL, mechanical compliance Cms, and inductance Le, which are parameters of the speaker unit, change nonlinearly with the displacement x of the speaker 13, as shown in, for example, FIGS.

[0096] Fig. 8 shows the characteristic of the force coefficient BL of the speaker unit with respect to the change in displacement x. That is, in Fig. 8, the vertical axis represents the force coefficient BL, and the horizontal axis represents the displacement x.

[0097] In this example, it can be seen that as the absolute value of the displacement x increases, the force coefficient BL decreases nonlinearly.

[0098] Fig. 9 also shows the characteristic of the mechanical compliance Cms of the speaker unit with respect to the change in displacement x. That is, in Fig. 9, the vertical axis represents the mechanical compliance Cms, and the horizontal axis represents the displacement x.

[0099] In this example as well, as in the case of FIG. 8, it can be seen that the value of the mechanical system compliance Cms changes nonlinearly with respect to the displacement x.

[0100] Fig. 10 shows the characteristics of the inductance Le of the speaker unit with respect to the change in displacement x. That is, in Fig. 10, the vertical axis represents the inductance Le, and the horizontal axis represents the displacement x.

[0101] In this example, it can be seen that as the value of the displacement x increases, the inductance Le decreases nonlinearly.

[0102] In this way, the force coefficient BL, the mechanical compliance Cms, and the inductance Le change nonlinearly.

[0103] Therefore, when predicting the displacement x that includes these nonlinear elements, the nonlinear parameters, the force coefficient BL, the mechanical compliance Cms, and the inductance Le, are calculated from the output displacement x, and the coefficients of the third-order IIR filter are updated using these calculated nonlinear parameters.

[0104] In such a case, for example, when the filter unit 32 is made up of a third-order IIR filter, the third-order IIR filter is configured as shown in Fig. 11. Note that in Fig. 11, parts corresponding to those in Fig. 7 are given the same reference numerals, and their explanation will be omitted as appropriate.

[0105] The third-order IIR filter shown in FIG. 11 includes amplifiers 61-1 to 61-4, delay units 62-1 to 62-3, an adder 63, delay units 64-1 to 64-3, amplifiers 65-1 to 65-3, and an update unit 91.

[0106] In the third-order IIR filter shown in FIG. 11, an input audio signal u[n] obtained by adjusting the gain of the input audio signal using an amplifier gain is supplied to an amplifier section 61-1 and a delay section 62-1 that constitute the third-order IIR filter.

[0107] In the input audio signal u[n], n indicates a sample, and in each of the delay units 62 and 64, the supplied signal is delayed by a time equivalent to one sample and output to the subsequent stage.

[0108] Based on the displacement x[n-1] supplied from the adder 63, the update unit 91 calculates the force coefficient BL[n], the mechanical compliance Cms[n], and the inductance Le[n] used to determine the displacement x[n] of the next sample.

[0109] For example, the force coefficient BL[n], the mechanical compliance Cms[n], and the inductance Le[n] can be calculated using a fourth-order approximation polynomial as shown in the following equation (4).

[0110]

number

[0111] In the formula (4), bl0 to bl4 respectively indicate the zeroth to fourth order terms of the approximation equation expressing the force coefficient BL. Similarly, cms0 to cms4 indicate the zeroth to fourth order terms of the approximation equation expressing the mechanical compliance Cms, and le0 to le4 indicate the zeroth to fourth order terms of the approximation equation expressing the inductance Le.

[0112] The update unit 91 performs the calculation of the formula (4) and updates the above-mentioned coefficients a0 to a3 and coefficients b1 to b3 based on the force coefficient BL[n], the mechanical compliance Cms[n], and the inductance Le[n] obtained as a result. Then, the update unit 91 supplies the updated coefficients to each of the amplifier units 61 and 65.

[0113] In this way, the update unit 91 calculates the force coefficient BL[n], the mechanical compliance Cms[n], and the inductance Le[n] based on the previous displacement x[n-1], thereby realizing nonlinear displacement prediction using an approximate polynomial, and making it possible to obtain a more accurate displacement x[n].

[0114] Here, with reference to Figs. 12 and 13, a comparison between the predicted results and the actual measured values ​​when linear prediction and nonlinear prediction are performed for the displacement of a given speaker 13 will be described.

[0115] 12 and 13, the vertical axis indicates the displacement x[n] of the speaker 13, and the horizontal axis indicates the frequency of the signal input to the speaker 13. In particular, on the vertical axis in these figures, a positive value of the displacement x[n] indicates the amount of displacement toward the listening point, that is, forward, and a negative value indicates the amount of displacement backward.

[0116] Fig. 12 shows the predicted results of displacement x[n] by nonlinear prediction and the actual measured values. In particular, in Fig. 12, the solid curve shows the predicted results by nonlinear prediction, and the dotted line shows the actual measured values. In this example, it can be seen that the difference (prediction error) between the predicted results and the actual measured values ​​is small at each frequency, regardless of the signal level, i.e., the amount of displacement of the speaker 13, and the displacement x[n] can be predicted with high accuracy.

[0117] In contrast, Fig. 13 shows the predicted results of displacement x[n] by linear prediction and the actual measured values. In particular, in Fig. 13, the solid curve shows the predicted results by linear prediction, and the dotted line shows the actual measured values. In this example, the force coefficient BL, mechanical compliance Cms, and inductance Le of the speaker 13 (speaker unit) have large nonlinearity, and it can be seen that as the signal level, i.e., the amount of displacement of the speaker 13, increases, the predicted results diverge from the actual measured values, and the prediction error increases.

[0118] From the above, it is clear that in such a speaker 13 (speaker unit), nonlinear prediction is required to reduce the prediction error of the displacement x[n].

[0119] When the speaker 13 is used in a range in which the force coefficient BL, the mechanical compliance Cms, and the inductance Le change little with respect to the change in the displacement x[n], the displacement x[n] may be obtained by linear prediction.

[0120] This can be the case, for example, when a high-pass filter that cuts the low frequencies of the input audio signal is placed before this displacement prediction process to attenuate the frequency band where the displacement becomes highly nonlinear, and the speaker 13 is used mainly in a frequency band that is close to linear.

[0121] In addition, the nonlinearity of the force coefficient BL, mechanical compliance Cms, and inductance Le with respect to changes in the displacement x[n] is small, and even in the case of a speaker 13 used in a linear region, the displacement x[n] may be predicted linearly.

[0122] <Doppler distortion correction> Next, the Doppler distortion correction, that is, the time correction for the input audio signal will be described.

[0123] For example, as shown on the left side of Fig. 14, when the diaphragm D11 of the speaker 13 moves forward (towards the listening point P11), the displacement x[n] becomes positive (plus). In this case, the time it takes for the sound (signal) output from the speaker 13 to reach the listening point P11 becomes shorter, so it is necessary to delay the output time of the sound by the plus amount of the displacement x[n]. Note that in Fig. 14, parts corresponding to those in Fig. 1 are given the same reference numerals, and their explanation will be omitted as appropriate.

[0124] On the other hand, when the diaphragm D11 of the speaker 13 moves backward, the displacement x[n] becomes negative. In this case, the time it takes for the sound (signal) output from the speaker 13 to reach the listening point P11 becomes longer, so it is necessary to advance the sound output time by the negative amount of the displacement x[n].

[0125] Therefore, to achieve Doppler distortion correction during playback, a delay offset is prepared by advancing the time of the input audio signal, and time correction is performed as Doppler distortion correction according to the amount of displacement of speaker 13 (displacement x[n]) centered on that offset.

[0126] Here, the time correction performed as the Doppler distortion correction is a process of obtaining a corrected audio signal by delaying or advancing the input audio signal in the time direction by an amount corresponding to the displacement x[n].

[0127] This process can be said to be a process of determining, by an interpolation process based on the sample values ​​of a plurality of samples of the input audio signal, the sample value of the sample to be processed in the signal when the input audio signal is delayed or advanced in the time direction by an amount corresponding to the displacement x[n]. In other words, the time correction performed as the Doppler distortion correction can be said to be a correction process of the amplitude value of the input audio signal.

[0128] The offset can be obtained by converting the maximum displacement of the diaphragm D11 of the speaker 13 from distance to time using the speed of sound, and further converting it into sample units using the sampling frequency.

[0129] Specifically, for example, it is assumed that the maximum displacement of the diaphragm D11 of the speaker 13 is ±10 [mm], and the sampling frequency Fs of the input audio signal is 48 [kHz].

[0130] In such a case, the maximum displacement of ±10 mm is converted to time using the speed of sound c = 340 m / s, which becomes ±29.4 μs. Furthermore, when ±29.4 μs is converted to sample units using a sampling frequency of 48 kHz, it becomes ±1.4118 samples.

[0131] Therefore, in this example, the number of samples by which the input audio signal is offset is set to two samples, and four delay units 121-1 to 121-4, which are delay circuits for a maximum of four samples, twice the offset, are provided as shown on the right side of the figure.

[0132] The delay unit 121-1 delays the supplied input audio signal by a time equivalent to one sample and supplies the delayed signal to the delay unit 121-2.

[0133] Furthermore, delay units 121-2 and 121-3 delay the input audio signals supplied from delay units 121-1 and 121-2 by a time equivalent to one sample and supply the delayed signals to delay units 121-3 and 121-4. Similarly, delay unit 121-4 delays the input audio signal supplied from delay unit 121-3 by a time equivalent to one sample and outputs the delayed signal to the subsequent stage.

[0134] In the following description, the delay units 121-1 to 121-4 will be simply referred to as delay unit 121 unless there is a need to distinguish between them.

[0135] In the example shown on the right side of Figure 14, by providing a delay circuit for four samples, it is possible to cover time changes from 0.5882 (= 2 - 1.4118) samples to 3.4118 (= 2 + 1.4118) samples, making it possible to perform time correction corresponding to changes in the displacement x[n] of the diaphragm D11 of the speaker 13.

[0136] To achieve this kind of time correction, an interpolation process can be used to obtain signals at time sample points that include decimal points, such as Lagrange interpolation, which is often used for interpolation of oversampling filters in DACs (Digital to Analog Converters) for CDs (Compact Discs).

[0137] Specifically, for example, Lagrange interpolation is used to perform interpolation using an (n-1)th degree polynomial over n points (e.g., n=3) or more, that is, n or more samples, that include an offset corresponding to a displacement of 0 [mm] of the speaker 13 and cover the maximum displacement of the speaker 13.

[0138] As an example, it is assumed that the maximum displacement of the diaphragm of the speaker 13 is ±10 [mm], and the sampling frequency Fs of the input audio signal is 48 [kHz].

[0139] In this case, for example, as shown in the following formula (5), an interpolation process is performed using a fourth-order interpolation polynomial at five points (five samples) from order n=0 to order n=4, and the input audio signal u[n] is delayed or advanced by a time corresponding to the displacement x[n] to generate a corrected audio signal u d It is possible to find [n].

[0140]

number

[0141] In addition, in formula (5), x indicates the number of correction samples, which is the correction time in sample units corresponding to the displacement x[n]. In addition, although an example in which Lagrange interpolation is used as the interpolation process will be described here, the present invention is not limited to this, and any interpolation process using a polynomial of degree 2 or higher, such as Newtonian interpolation or spline interpolation, may be used.

[0142] If sound is reproduced from the speaker 13 based on the corrected audio signal generated by the Lagrangian interpolation shown in the above equation (5), the Doppler distortion is cancelled out at the listening point P11, and high quality sound is observed.

[0143] For example, the frequency f 1 A low-frequency sinusoidal signal at frequency f 2 When a corrected audio signal is generated by the Doppler distortion correction of the present technology from an input audio signal consisting of a sine wave signal in the high frequency range and is played back on the speaker 13, the result is as shown in Fig. 15. In Fig. 15, the vertical axis indicates the amplitude of the signal, and the horizontal axis indicates the frequency.

[0144] Figure 15 shows the frequency components of an audio signal obtained by picking up (measuring) the sound reproduced by speaker 13 based on the corrected audio signal obtained by Doppler distortion correction of the present technology using a microphone at listening point P11.

[0145] In this example, as in Figure 3, the frequency f 1 and frequency f 2 and frequency f 2 The frequency (f 2 -f 1 ) and frequency (f 2 +f 1 ) ingredients.

[0146] In particular, in Fig. 15, the frequency (f 2 -f 1 ) and frequency (f 2 +f 1The dotted line in the component of (1) represents the Doppler distortion reduced by performing Doppler distortion correction. In other words, this dotted line represents the difference in Doppler distortion between the case where Doppler distortion correction is performed and the case where it is not performed (as in FIG. 3).

[0147] By performing Doppler distortion correction in this manner, Doppler distortion can be suppressed, and higher quality sound reproduction can be achieved.

[0148] <Configuration example of Doppler distortion correction unit> When the above-described Doppler distortion correction is performed, the Doppler distortion correction unit 22 of the signal processing device 11 is configured, for example, as shown in Fig. 16. In Fig. 16, the same reference numerals are used to designate parts corresponding to those in Fig. 14, and the description thereof will be omitted as appropriate.

[0149] In the example shown in FIG. 16, the Doppler distortion correction unit 22 includes delay units 121-1 to 121-4, a conversion unit 151, and an interpolation processing unit 152.

[0150] The conversion unit 151 converts the displacement x[n] supplied from the filter unit 32 of the speaker displacement prediction unit 21 into a correction sample number x in sample units corresponding to the displacement x[n], and supplies it to the interpolation processing unit 152.

[0151] The conversion unit 151 includes a delay unit 161 - 1 , a delay unit 161 - 2 , a multiplication unit 162 , a multiplication unit 163 , and an addition unit 164 .

[0152] The delay unit 161-1 delays the displacement x[n] supplied from the filter unit 32 by a time equivalent to one sample and supplies the delayed displacement x[n] to the delay unit 161-2. The delay unit 161-2 delays the displacement x[n] supplied from the delay unit 161-1 by a time equivalent to one sample and supplies the delayed displacement x[n] to the multiplication unit 162.

[0153] In the following description, when there is no need to particularly distinguish between the delay unit 161-1 and the delay unit 161-2, they will also be simply referred to as the delay unit 161.

[0154] The multiplication unit 162 multiplies the displacement x[n] supplied from the delay unit 161-2 by the inverse 1 / c of ​​the sound speed c=340 [m / s], and supplies the correction time corresponding to the displacement x[n] obtained as a result to the multiplication unit 163. That is, the multiplication unit 162 divides the displacement x[n] by the sound speed c to calculate the correction time.

[0155] The multiplication unit 163 multiplies the correction time supplied from the multiplication unit 162 by the sampling frequency Fs of the input audio signal, and supplies the resulting correction sample number, which is the correction time in sample units including decimal places, to the addition unit 164.

[0156] The adder 164 obtains a final corrected sample number x by adding the offset sample number to the corrected sample number supplied from the multiplier 163, and supplies the result to the interpolation processing unit 152. For example, in this example, the offset sample number "2" is added to the corrected sample number supplied from the multiplier 163 to obtain the corrected sample number x.

[0157] The interpolation processing unit 152 performs an interpolation process based on the input audio signal u[n], the input audio signals u[n-1] to u[n-4] supplied from the delay units 121, and the correction sample number x supplied from the adder unit 164, to generate a corrected audio signal u[n]. d Generate [n].

[0158] For example, the interpolation processing unit 152 performs Lagrange interpolation by calculating the above-mentioned equation (5). The interpolation processing unit 152 converts the corrected audio signal u d [n] is supplied to the amplifier section 12.

[0159] <Explanation of Regeneration Treatment> Next, the operation of the audio reproduction system shown in Fig. 4 will be described. That is, the reproduction process performed by the audio reproduction system will be described below with reference to the flowchart in Fig. 17. This reproduction process is started when an input audio signal, which is a source signal, is input and an instruction is given to reproduce sound of content or the like.

[0160] In step S11 , the amplification unit 31 multiplies the supplied input audio signal u[n] by the amplifier gain in the amplification unit 12 , and supplies the amplified input audio signal u[n] obtained as a result to the filter unit 32 .

[0161] In step S12, the filter unit 32 filters the input audio signal u[n] supplied from the amplifier unit 31 using a third-order IIR filter, and supplies the resulting displacement x[n] to the delay unit 161-1 of the converter 151.

[0162] For example, in the filter unit 32, as described with reference to FIG. 11, the update unit 91 calculates the above-mentioned equation (4) based on the displacement x[n-1] supplied from the adder unit 63, and calculates the force coefficient BL[n], the mechanical system compliance Cms[n], and the inductance Le[n].

[0163] In addition, the update unit 91 calculates coefficients a0 to a3 and coefficients b1 to b3 based on the TS parameters including the force coefficient BL[n], mechanical compliance Cms[n], and inductance Le[n], and supplies them to each amplifier unit 61 and amplifier unit 65.

[0164] Furthermore, each delay unit 62 and delay unit 64 delays the supplied signal by one sample time and outputs it to the subsequent stage, and amplification unit 61 and amplification unit 65 multiply the supplied signal by the coefficient supplied from update unit 91 and supply the obtained signal to addition unit 63.

[0165] The adder 63 adds the signals supplied from the amplifiers 61 and 65 to obtain a displacement x[n], and supplies the displacement x[n] to the updater 91 and delay unit 161-1.

[0166] Then, the delay unit 161-1 delays the displacement x[n] supplied from the addition unit 63 and supplies it to the delay unit 161-2, and the delay unit 161-2 delays the displacement x[n] supplied from the delay unit 161-1 and supplies it to the multiplication unit 162.

[0167] Such filtering by the filter unit 32 results in a nonlinear prediction of the displacement x[n].

[0168] In step S13, the multiplication unit 162 obtains a correction time by multiplying the displacement x[n] supplied from the delay unit 161-2 by the reciprocal 1 / c of ​​the sound speed c, and supplies the obtained correction time to the multiplication unit 163.

[0169] In step S14, the multiplication unit 163 obtains a corrected sample number by multiplying the correction time supplied from the multiplication unit 162 by the sampling frequency Fs, and supplies the corrected sample number to the addition unit 164. The addition unit 164 also obtains a final corrected sample number x by adding the offset sample number to the corrected sample number supplied from the multiplication unit 163, and supplies the final corrected sample number x to the interpolation processing unit 152.

[0170] Furthermore, each delay unit 121 delays the input audio signal supplied thereto and supplies the delayed signal to the subsequent delay unit 121 or the interpolation processing unit 152 .

[0171] In step S15, the interpolation processing unit 152 performs Lagrangian interpolation based on the input audio signal u[n], the input audio signals u[n-1] to u[n-4] supplied from each delay unit 121, and the number of correction samples x supplied from the addition unit 164.

[0172] That is, the interpolation processing unit 152 performs Lagrange interpolation by calculating the above-mentioned equation (5), and the resulting corrected audio signal ud [n] is supplied to the amplifier section 12.

[0173] In step S16, the amplifier 12 receives the corrected audio signal u d The gain adjustment is performed by multiplying [n] by the amplifier gain, and the corrected audio signal after the gain adjustment is u d [n] is supplied to speaker 13.

[0174] In step S17, the speaker 13 outputs the corrected audio signal u d The audio reproduction system outputs sound by driving the input audio signal based on [n], and the reproduction process ends. In the audio reproduction system, the above-described process is performed for each sample of the input audio signal.

[0175] In this manner, the audio reproduction system obtains the displacement x[n] by nonlinear prediction, and performs Lagrange interpolation using a polynomial of degree 2 or higher based on the number of correction samples x corresponding to the displacement x[n] to obtain a corrected audio signal u d [n] is calculated. By doing this, it is possible to reduce Doppler distortion and achieve high quality sound reproduction.

[0176] In the above, the speaker system, i.e., the speaker 13, has been described as an example of a sealed type, but the present technology is not limited to this and can be applied to any type of speaker, such as a bass reflex type or a passive radiator type.

[0177] For example, when the speaker 13 is a bass reflex type speaker, the equivalent circuit of the speaker 13 is as shown in FIG.

[0178] In the example of Fig. 18, the circuit on the left side shows the equivalent circuit of the electrical system, and the circuit on the right side shows the equivalent circuit of the mechanical system. Each character in Fig. 18 shows each parameter called TS parameter, and these TS parameters are the same as those in Fig. 6.

[0179] Furthermore, for example, if the speaker 13 is a passive radiator type, the equivalent circuit of the speaker 13 is as shown in FIG.

[0180] In the example of Fig. 19, the circuit on the left side shows the equivalent circuit of the electrical system, and the circuit on the right side shows the equivalent circuit of the mechanical system. Each character in Fig. 19 shows each parameter called TS parameter, and these TS parameters are the same as those in Fig. 6.

[0181] In the examples shown in Figures 18 and 19, if a filter for displacement prediction obtained by performing digital filtering based on the equivalent circuit of the speaker 13 is used, the displacement x[n] can be obtained by nonlinear prediction.

[0182] Second embodiment Example of an audio playback system configuration Furthermore, in the above description, an example has been described in which the input audio signal, which is the source signal, is input to the speaker displacement prediction unit 21 as shown in FIG. 4, but a corrected audio signal after Doppler distortion correction may be input.

[0183] In such a case, the audio reproduction system is configured as shown in Fig. 20. Note that in Fig. 20, parts corresponding to those in Fig. 4 are given the same reference numerals, and their explanation will be omitted as appropriate.

[0184] The audio reproduction system shown in FIG. 20 includes a signal processing device 11, an amplifier unit 12, and a speaker 13, and the signal processing device 11 includes a speaker displacement prediction unit 21 and a Doppler distortion correction unit 22.

[0185] Although not shown in the figure, the speaker displacement prediction unit 21 has an amplifier unit 31 and a filter unit 32, and the Doppler distortion correction unit 22 has delay units 121-1 to 121-4, a conversion unit 151, and an interpolation processing unit 152.

[0186] This audio playback system differs from the audio playback system shown in Figure 4 in that the corrected audio signal output from the Doppler distortion correction unit 22 is input to the speaker displacement prediction unit 21, but in other respects it is the same as the audio playback system of Figure 4.

[0187] Therefore, in the audio reproduction system shown in FIG. 20, the amplifier unit 31 of the speaker displacement prediction unit 21 amplifies the corrected audio signal supplied from the interpolation processing unit 152 of the Doppler distortion correction unit 22 by the amplifier gain in the amplifier unit 12, and supplies it to the filter unit 32.

[0188] The filter unit 32 performs nonlinear prediction by filtering the corrected audio signal supplied from the amplifier unit 31, and supplies the displacement obtained as a result of the prediction to the conversion unit 151 of the Doppler distortion correction unit 22, more specifically, to the delay unit 161-1 of the conversion unit 151.

[0189] In this way, even in the case of the configuration shown in FIG. 20, it is possible to reduce Doppler distortion and achieve high-quality sound reproduction, similarly to the case in FIG.

[0190] In the above first and second embodiments, the speaker 13 is a full-range speaker. However, the present technology can also be applied to a multi-way mid-speaker or a woofer.

[0191] For example, if the speaker 13 is a multi-way mid-speaker or woofer, and the band-splitting filter has gentle characteristics such as 12 dB / Oct, the high frequencies affected by Doppler distortion are also reproduced, although to a small extent. By applying this technology and correcting the Doppler distortion, the quality of the sound emitted from a multi-way speaker or the like is improved.

[0192] Example of computer configuration The above-mentioned series of processes can be executed by hardware or software. When the series of processes is executed by software, the program constituting the software is installed in a computer. Here, the computer includes a computer built into dedicated hardware, and a general-purpose personal computer, for example, capable of executing various functions by installing various programs.

[0193] FIG. 21 is a block diagram showing an example of the hardware configuration of a computer that executes the above-mentioned series of processes according to a program.

[0194] In the computer, a CPU (Central Processing Unit) 501, a ROM (Read Only Memory) 502, and a RAM (Random Access Memory) 503 are interconnected via a bus 504.

[0195] An input / output interface 505 is further connected to the bus 504. An input unit 506, an output unit 507, a recording unit 508, a communication unit 509, and a drive 510 are connected to the input / output interface 505.

[0196] The input unit 506 includes a keyboard, a mouse, a microphone, an image sensor, etc. The output unit 507 includes a display, a speaker, etc. The recording unit 508 includes a hard disk, a non-volatile memory, etc. The communication unit 509 includes a network interface, etc. The drive 510 drives a removable recording medium 511 such as a magnetic disk, an optical disk, a magneto-optical disk, or a semiconductor memory.

[0197] In a computer configured as described above, the CPU 501 loads, for example, a program recorded in the recording unit 508 into the RAM 503 via the input / output interface 505 and the bus 504, and executes the program, thereby performing the series of processes described above.

[0198] The program executed by the computer (CPU 501) can be provided by being recorded on a removable recording medium 511 such as a package medium, for example. The program can also be provided via a wired or wireless transmission medium such as a local area network, the Internet, or digital satellite broadcasting.

[0199] In the computer, the program can be installed in the recording unit 508 via the input / output interface 505 by mounting the removable recording medium 511 in the drive 510. The program can also be received by the communication unit 509 via a wired or wireless transmission medium and installed in the recording unit 508. Alternatively, the program can be installed in the ROM 502 or the recording unit 508 in advance.

[0200] In addition, the program executed by the computer may be a program in which processing is performed chronologically in the order described in this specification, or it may be a program in which processing is performed in parallel or at the required timing, such as when called.

[0201] Furthermore, the embodiments of the present technology are not limited to the above-described embodiments, and various modifications are possible without departing from the gist of the present technology.

[0202] For example, the present technology can be configured as cloud computing, in which a single function is shared and processed collaboratively by multiple devices via a network.

[0203] Furthermore, each step described in the above flow chart can be executed by one device, or can be shared and executed by a plurality of devices.

[0204] Furthermore, when a single step includes multiple processes, the multiple processes included in the single step can be executed by a single device, or can be shared and executed by multiple devices.

[0205] Furthermore, the present technology can also be configured as follows.

[0206] (1) a displacement prediction unit that predicts a displacement of a diaphragm of a speaker based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed and a sound is reproduced from the speaker based on the audio signal; a correction unit that performs a correction in a time direction on the audio signal by an interpolation process using three or more samples of the audio signal based on a correction time obtained based on the displacement obtained by the prediction and the sound speed; A signal processing device comprising: (2) The displacement prediction unit obtains the displacement by nonlinear prediction. A signal processing device as described in (1). (3) The displacement prediction unit performs the nonlinear prediction using polynomial approximation. A signal processing device as described in (2). (4) The correction time is a delay time of the audio signal, and when the diaphragm moves forward, the correction time increases, and when the diaphragm moves backward, the correction time decreases. A signal processing device according to any one of (1) to (3). (5) The correction unit calculates a number of samples of the correction time based on the displacement obtained by the prediction, the sound speed, and a sampling frequency of the audio signal, and performs the interpolation process based on the number of samples. A signal processing device according to any one of (1) to (4). (6) The correction unit calculates the number of samples including a decimal point value. A signal processing device according to (5). (7) The correction unit performs the correction in the time direction by correcting sample values ​​of the audio signal through the interpolation process. A signal processing device according to any one of (1) to (6). (8) The interpolation process is Lagrange interpolation, Newtonian interpolation, or spline interpolation. A signal processing device according to any one of (1) to (7). (9) The displacement prediction unit predicts the displacement based on the audio signal obtained by the interpolation process. A signal processing device according to any one of (1) to (8). (10) A signal processing device, predicting, based on an audio signal obtained by mixing a high-frequency signal and a low-frequency signal, a displacement of a diaphragm of a speaker when a sound is reproduced from the speaker based on the audio signal; Based on a correction time calculated based on the displacement and sound speed obtained by the prediction, the audio signal is corrected in the time direction by an interpolation process using three or more samples of the audio signal. Signal processing methods. (11) predicting, based on an audio signal obtained by mixing a high-frequency signal and a low-frequency signal, a displacement of a diaphragm of a speaker when a sound is reproduced from the speaker based on the audio signal; Based on a correction time calculated based on the displacement and sound speed obtained by the prediction, the audio signal is corrected in the time direction by an interpolation process using three or more samples of the audio signal. A program that causes a computer to execute a process including steps. [Explanation of symbols]

[0207] 11 signal processing device, 12 amplifier, 13 speaker, 21 speaker displacement prediction unit, 22 Doppler distortion correction unit, 31 amplifier, 32 filter unit, 151 conversion unit, 152 interpolation processing unit

Claims

1. a displacement prediction unit that predicts a displacement of a diaphragm of a speaker based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed and a sound is reproduced from the speaker based on the audio signal; a correction unit that performs a correction in a time direction on the audio signal by an interpolation process using three or more samples of the audio signal based on a correction time obtained based on the displacement obtained by the prediction and a sound speed; A signal processing device comprising:

2. The displacement prediction unit obtains the displacement by nonlinear prediction. The signal processing device according to claim 1 .

3. The displacement prediction unit performs the nonlinear prediction using polynomial approximation. The signal processing device according to claim 2 .

4. The correction time is a delay time of the audio signal, and when the diaphragm moves forward, the correction time increases, and when the diaphragm moves backward, the correction time decreases. The signal processing device according to claim 1 .

5. The correction unit calculates a number of samples of the correction time based on the displacement obtained by the prediction, the sound speed, and a sampling frequency of the audio signal, and performs the interpolation process based on the number of samples. The signal processing device according to claim 1 .

6. The correction unit calculates the number of samples including a decimal point value. The signal processing device according to claim 5 .

7. The correction unit performs the correction in the time direction by correcting sample values ​​of the audio signal through the interpolation process. The signal processing device according to claim 1 .

8. The interpolation process is Lagrange interpolation, Newtonian interpolation, or spline interpolation. The signal processing device according to claim 1 .

9. The displacement prediction unit predicts the displacement based on the audio signal obtained by the interpolation process. The signal processing device according to claim 1 .

10. A signal processing device, predicting, based on an audio signal obtained by mixing a high-frequency signal and a low-frequency signal, a displacement of a diaphragm of a speaker when a sound is reproduced by the speaker based on the audio signal; Based on a correction time calculated based on the displacement and sound speed obtained by the prediction, the audio signal is corrected in the time direction by an interpolation process using three or more samples of the audio signal. Signal processing methods.

11. predicting, based on an audio signal obtained by mixing a high-frequency signal and a low-frequency signal, a displacement of a diaphragm of a speaker when a sound is reproduced by the speaker based on the audio signal; Based on a correction time calculated based on the displacement and sound speed obtained by the prediction, the audio signal is corrected in the time direction by an interpolation process using three or more samples of the audio signal. A program that causes a computer to execute a process including steps.

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