Signal processing apparatus and method and program

Abstract

The present technology relates to a signal processing apparatus and method and program capable of reducing Doppler distortion. The signal processing apparatus includes: a displacement prediction unit that, in a case where a speaker reproduces sound based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed, predicts a displacement of a diaphragm of the speaker based on the audio signal; and a correction unit that, based on the displacement obtained from the prediction and a correction time obtained based on a speed of sound, performs time direction correction on the audio signal by performing interpolation processing using at least three samples of the audio signal. The present technology is applicable in an audio reproduction system.

Description

Technical Field

[0001] This technology relates to signal processing apparatus, methods, and procedures, and more particularly to signal processing apparatus, methods, and procedures capable of reducing Doppler distortion. Background Technology

[0002] In music playback using speakers, for example, a phenomenon may occur where high-frequency signals are affected by low-frequency signals, resulting in blurred sound image positioning or sound vibration.

[0003] Doppler distortion can be cited as a contributing factor to this phenomenon, in which the loudspeaker diaphragm vibrates back and forth due to low-frequency signals, and the sound source position of the signal radiated from the diaphragm changes due to the diaphragm's back-and-forth movement. This is particularly noticeable in loudspeakers that output a full range of frequencies from low to high from a single diaphragm.

[0004] Therefore, a technique is proposed to counteract Doppler distortion by using a quadratic integral signal to control a clock oscillator and using a variable delay device to change the delay time of the signal (e.g., see PTL1).

[0005] A technique is also proposed to correct nonlinear distortion of a loudspeaker by linearly predicting its displacement using a parameter with a displacement of 0 [mm] in digital signal processing (e.g., see PTL2). This technique uses a linear prediction of the displacement used to correct nonlinear distortion in the loudspeaker to correct Doppler distortion.

[0006] [List of Citations]

[0007] [Patent Literature]

[0008] [PTL1]

[0009] JP 1556673 B

[0010] [PTL2]

[0011] U.S. Patent No. 5,438,625. Summary of the Invention

[0012] [Technical Issues]

[0013] However, the above techniques are insufficient to adequately reduce Doppler distortion.

[0014] For example, in the technique described in PTL1, only twice the integration is performed as in the method used to obtain the movement (displacement) of the loudspeaker diaphragm. However, the movement obtained by integration is often different from the actual movement of the loudspeaker displacement, which has the opposite effect of increasing distortion.

[0015] Furthermore, using the techniques described in PTL2, phase modulation is performed by controlling the delay time as a method for correcting Doppler distortion, and linear interpolation is used to calculate the data between sampling intervals in the control of the delay time of the discrete signal.

[0016] Specifically, as the frequency of the high-frequency signal increases, Doppler distortion increases at 6 dB / Oct, but linear interpolation produces large errors, and new distortions caused by these errors will appear in this case. Furthermore, time correction is not considered when the displacement in the speaker diaphragm is large and exceeds a single sampling interval.

[0017] Given this situation, this technology has been implemented, and it is able to reduce Doppler distortion.

[0018] [Solution to the problem]

[0019] A signal processing apparatus according to one aspect of the present technology includes: a displacement prediction unit that predicts the displacement of a loudspeaker diaphragm based on an audio signal in which the loudspeaker reproduces sound based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed; and a correction unit that performs time direction correction on the audio signal by performing interpolation processing using at least three samples of the audio signal based on the displacement obtained from the prediction and a correction time obtained based on the speed of sound.

[0020] A signal processing method or program according to one aspect of the present technology includes a signal processing apparatus that performs the following steps: predicting the displacement of the speaker diaphragm based on the audio signal when the speaker reproduces sound based on an audio signal in which a high-frequency signal and a low-frequency signal are mixed; and performing time direction correction on the audio signal by performing interpolation processing using at least three samples of the audio signal based on the displacement obtained from the prediction and a correction time obtained based on the speed of sound.

[0021] In one aspect of this technology, when a loudspeaker reproduces sound based on an audio signal in which high-frequency and low-frequency signals are mixed, the displacement of the loudspeaker diaphragm is predicted based on the audio signal; and time direction correction is performed on the audio signal by performing interpolation processing using at least three samples of the audio signal based on the displacement obtained from the prediction and a correction time obtained based on the speed of sound. Attached Figure Description

[0022] Figure 1 This is a diagram illustrating Doppler distortion.

[0023] Figure 2 It shows a diagram illustrating Doppler distortion.

[0024] Figure 3 It shows a diagram illustrating Doppler distortion.

[0025] Figure 4This is a diagram illustrating an example of the configuration of an audio playback system.

[0026] Figure 5 This is a diagram illustrating the processing flow when correcting Doppler distortion.

[0027] Figure 6 This is a diagram showing an example of the equivalent circuit of a loudspeaker.

[0028] Figure 7 This is a diagram illustrating an example of the configuration of a third-order IIR filter.

[0029] Figure 8 It is a diagram showing the characteristics of the force coefficient relative to the speaker displacement.

[0030] Figure 9 This is a PTL diagram showing the characteristics of the mechanical system's compliance with respect to the speaker's displacement.

[0031] Figure 10 This is a diagram showing the inductance characteristics relative to the speaker displacement.

[0032] Figure 11 This is a diagram illustrating an example of the configuration of a third-order IIR filter.

[0033] Figure 12 This is a PTL plot showing the predicted results and the actual values ​​of the displacement when performing nonlinear prediction.

[0034] Figure 13 This is a PTL plot showing the predicted results and the actual values ​​of the displacement when performing linear prediction.

[0035] Figure 14 This is a diagram showing Doppler distortion correction.

[0036] Figure 15 This is a diagram illustrating the effect of Doppler distortion correction.

[0037] Figure 16 This is a diagram illustrating an example of the configuration of a Doppler distortion correction unit.

[0038] Figure 17 This is a flowchart illustrating the playback process.

[0039] Figure 18 This is a diagram showing an example of the equivalent circuit of a loudspeaker.

[0040] Figure 19 This is a diagram showing an example of the equivalent circuit of a loudspeaker.

[0041] Figure 20 This is a diagram illustrating an example of the configuration of an audio playback system.

[0042] Figure 21 This is a diagram illustrating an example of a computer configuration. Detailed Implementation

[0043] In the following description, embodiments of the application of this technology will be described with reference to the accompanying drawings.

[0044] <First Implementation Method>

[0045] <This technology>

[0046] This technique reduces Doppler distortion by performing time-shifted audio signal correction through interpolation using second-order or higher polynomial expressions. It further improves the accuracy of predicting actual movement in the speaker diaphragm by performing nonlinear predictions of displacement within the diaphragm, thereby reducing Doppler distortion.

[0047] When using a speaker to reproduce sounds such as music, a phenomenon may occur where high-frequency signals are affected by low-frequency signals, resulting in blurred sound image localization or sound jitter, and Doppler distortion is one of the factors that causes this phenomenon.

[0048] like Figure 1 As shown, for example, Doppler distortion occurs due to the change in the sound source position of the signal radiated from the speaker diaphragm D11, which is caused by the diaphragm D11 vibrating back and forth due to the low-frequency signal.

[0049] Specifically, for example, if the vibrating diaphragm D11 is as follows: Figure 1 As indicated by arrow Q11, the sound source position (i.e., the position where the sound wave is generated) moves forward in the direction of the listening point P11, and the phase of the sound (signal) output by the diaphragm D11 moves forward. As a result, the wavelength of the sound output from the diaphragm D11 becomes shorter.

[0050] Conversely, if the diaphragm D11 moves backward as indicated by arrow Q12, that is, in the opposite direction to the listening point P11, 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.

[0051] In this way, when a high-frequency signal (sound) is output from the diaphragm D11 while the diaphragm D11 moves back and forth due to a low-frequency signal, the wavelength of the sound changes.

[0052] This phenomenon is called "Doppler distortion," and it is particularly noticeable in loudspeakers that output low to high frequencies from a single diaphragm.

[0053] Full-range loudspeakers are frequently used in so-called normal two-channel stereo playback, 5.1-channel surround sound, sound augmented reality (AR) and virtual reality (VR) using multiple loudspeakers, and wavefront synthesis, so that the loudspeakers can be regarded as ideal point sound sources.

[0054] Doppler distortion in loudspeakers affects the location and volume of the expected sound source and the actual sound source being reproduced.

[0055] For example, such as Figure 2 As shown, Doppler distortion occurs when low-frequency and high-frequency signals are reproduced simultaneously.

[0056] In other words, as described above, low-frequency signals cause the speaker's diaphragm to vibrate back and forth, which changes the location of the high-frequency signal's sound source, and in turn, changes the time it takes for the sound to reach the listening point. This shortens or lengthens the wavelength of the high-frequency signal (sound), resulting in signal distortion.

[0057] For example, when viewing low-frequency and high-frequency signals simultaneously output when Doppler distortion occurs on the frequency axis, the situation is as follows: Figure 3 As shown. It is important to note that in Figure 3 In the diagram, the vertical axis represents the amplitude of the signal, and the horizontal axis represents the frequency.

[0058] In this example, the component with frequency f1 is a low-frequency signal component, while the component with frequency f2 is a high-frequency signal component. Specifically, both the low-frequency and high-frequency signals are considered sinusoidal signals here.

[0059] In this example, the simultaneous output of low-frequency and high-frequency signals from the speaker results in Doppler distortion. In other words, the frequency (f2-f1) and frequency (f2+f1) components, which are frequency components in the sideband, are signal components generated by Doppler distortion.

[0060] As a method to reduce the aforementioned Doppler distortion, it is conceivable to predict the back-and-forth movement (displacement) of the loudspeaker diaphragm and use the predicted displacement to control the delay time to reverse the forward and backward movement of the loudspeaker diaphragm. That is, as a delay time control, control is performed so that the timing delay of the signal output (reproduction) corresponds to the time of the predicted diaphragm displacement.

[0061] In this way, the arrival time of the sound, which varies due to the back-and-forth movement of the speaker diaphragm, at the listening point is controlled to be uniform, which makes it possible to reduce Doppler distortion.

[0062] Based on the above, in order to eliminate Doppler distortion, the movement of the loudspeaker diaphragm can be obtained by prediction or actual measurement, and the signal can be time-corrected in the opposite direction by an amount equivalent to the change in the arrival time of the sound (signal) caused by the movement.

[0063] However, it is difficult to adequately reduce Doppler distortion using current and proposed techniques.

[0064] Another method for reducing Doppler distortion has been proposed by modifying the shape of the loudspeaker diaphragm. For example, a method has been proposed to reduce Doppler distortion by shaping the diaphragm into a non-circular shape, such as an asymmetric ellipse, so that higher frequency signals are radiated non-uniformly from the diaphragm and phase modulation is dispersed. However, even with this method, the improvement in Doppler distortion is small and cannot be considered sufficient.

[0065] Therefore, by using this technology, the movement (displacement) of the speaker diaphragm can be predicted with higher accuracy by performing nonlinear prediction and the audio signal can be corrected by interpolation processing using polynomial expressions of second order or higher.

[0066] For example, speaker displacement can be predicted more accurately by performing nonlinear prediction. Furthermore, if interpolation is performed using polynomial expressions of second order or higher, the interpolation can be performed more accurately compared to performing linear interpolation at two points. This allows for further reduction of Doppler distortion.

[0067] <Configuration Example of an Audio Playback System>

[0068] Figure 4 This is a diagram illustrating a configuration example of an implementation of an audio playback system using this technology.

[0069] Figure 4 The audio playback system shown includes a signal processing unit 11, an amplifier unit 12, and a speaker 13.

[0070] The signal processing device 11 performs correction on the audio signal of the content to be reproduced to reduce Doppler distortion, and provides the obtained corrected audio signal as a result to the amplifier unit 12.

[0071] In the following text, the audio signal input to the signal processing device 11 (i.e., the source signal of the sound to be reproduced) will also be specifically referred to as the "input audio signal". In addition, the correction used to reduce Doppler distortion will also be referred to as "Doppler distortion correction" in the following text.

[0072] The input audio signal to the signal processing device 11 is an audio signal containing high-frequency components and low-frequency components, that is, a mixed audio signal containing high-frequency signals and low-frequency signals.

[0073] The amplifier unit 12 amplifies the corrected audio signal provided from the signal processing device 11 by means of an amplifier gain that is a predetermined output voltage, and then provides the amplified corrected audio signal to the speaker 13 to drive the speaker 13.

[0074] The loudspeaker 13 is, for example, a full-range loudspeaker that outputs sound in the frequency band from low to high frequencies. Note that because Doppler distortion occurs in other loudspeakers besides full-range loudspeakers, the loudspeaker 13 is not limited to a full-range loudspeaker and can be any loudspeaker.

[0075] The loudspeaker 13 vibrates the diaphragm by driving the diaphragm based on the corrected audio signal provided from the amplifier unit 12, and outputs sound based on the corrected audio signal.

[0076] The signal processing device 11 also includes a loudspeaker displacement prediction unit 21 and a Doppler distortion correction unit 22.

[0077] The loudspeaker displacement prediction unit 21 predicts the displacement of the loudspeaker 13 based on the provided input audio signal. More specifically, it predicts the displacement of the diaphragm of the loudspeaker 13, which is the target for correcting Doppler distortion, and provides the prediction result to the Doppler distortion correction unit 22.

[0078] In other words, in the speaker displacement prediction unit 21, the displacement of the diaphragm of the speaker 13 when sound is reproduced through the speaker 13 based on the input audio signal is obtained through nonlinear prediction based on the input audio signal. Specifically, in the speaker displacement prediction unit 21, nonlinear prediction is performed using a polynomial approximation (approximate polynomial), and the displacement of the speaker 13 is obtained.

[0079] The loudspeaker displacement prediction unit 21 includes an amplifier unit 31 and a filter unit 32.

[0080] Amplifier unit 31 amplifies the input audio signal provided by amplifier unit 12 through output voltage (amplifier gain) and provides the amplified signal to filter unit 32.

[0081] The filter unit 32 is composed of, for example, a third-order infinite impulse response (IIR) filter, which performs nonlinear prediction by filtering the input audio signal provided from the amplifier unit 31, and provides the displacement obtained as a prediction result to the Doppler distortion correction unit 22.

[0082] The Doppler distortion correction unit 22 performs Doppler distortion correction on the input audio signal based on the prediction result provided by the filter unit 32 of the speaker displacement prediction unit 21, and provides the obtained corrected audio signal as a result to the amplifier unit 12.

[0083] In the signal processing device 11, by means of, Figure 5 The process is roughly as shown in the diagram to generate the corrected audio signal.

[0084] In other words, firstly, gain adjustment is performed in amplifier unit 31 by multiplying the input audio signal (source signal) by the amplifier gain. This amplifier gain is the gain value used for amplification (i.e., gain adjustment) in amplifier unit 12.

[0085] Next, in filter unit 32, for example, a filter such as a third-order IIR filter is used to filter the gain-adjusted input audio signal.

[0086] The filtering process is a nonlinear displacement prediction process that predicts the displacement of the diaphragm of the loudspeaker 13. The prediction result obtained by this displacement prediction process is provided to the Doppler distortion correction unit 22. For example, as the prediction result of the diaphragm displacement, a distance (such as displacement x [mm]) representing the magnitude of the change in the position of the diaphragm is obtained.

[0087] In the Doppler distortion correction unit 22, the displacement x [mm] provided based on the prediction result is converted (transformed) into a correction time d = x / c [s] corresponding to the displacement x [mm]. The correction time d indicates the delay time of the input audio signal.

[0088] For example, when the diaphragm of speaker 13 moves forward, i.e., toward the listening point, the displacement x [mm] is positive. In this case, the correction time d is increased (taken as a positive value) to delay the timing of the sound output by speaker 13.

[0089] Conversely, when the diaphragm of speaker 13 moves backward, i.e. in the opposite direction to the listening point, the displacement x [mm] becomes negative. In this case, the correction time d decreases (becomes negative), causing the sound output timing of speaker 13 to advance.

[0090] Furthermore, in the Doppler distortion correction unit 22, based on the sampling frequency Fs of the input audio signal, the correction time d[s] is transformed (converted) into a time in units of samples corresponding to the displacement x[mm], that is, the number of correction samples d×Fs[samples].

[0091] The number of corrected samples obtained in this way represents the amount of correction used to delay or advance the output timing of the input audio signal in the time direction to correct Doppler distortion. Specifically, the number of corrected samples also includes values ​​below the decimal point.

[0092] Furthermore, in the Doppler distortion correction unit 22, a corrected audio signal is generated by performing an interpolation process based on the number of corrected samples (correction amount) and the input audio signal (i.e., by performing a time delay correction process) to offset the input audio signal in the time direction by the number of corrected samples.

[0093] In this case, instead of linear interpolation between two points of the sampled sample below the decimal point, as a delay time correction process for the input audio signal, interpolation is performed using a polynomial expression of second order or higher, such as second order or higher Lagrange interpolation, using at least three points, i.e., more than three samples of the input audio signal.

[0094] This interpolation process, which uses polynomial expressions of second order or higher, corrects the sampled values ​​of the input audio signal. This results in a time delay correction process that shifts the input audio signal by the number of corrected samples in the time direction.

[0095] In the Doppler distortion correction unit 22 that performs this interpolation, a time offset is prepared, taking into account the displacement of the diaphragm of the speaker 13 and the sampling frequency of the input audio signal. This offset is a delay sample number that delays the output timing of the corrected audio signal as a whole, regardless of the amount of Doppler distortion correction.

[0096] In the signal processing device 11, Doppler distortion correction is performed as described above. This Doppler distortion correction corresponds to phase modulation on the input audio signal.

[0097] <Speaker Displacement Prediction>

[0098] 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.

[0099] In filter unit 32, the displacement of speaker 13 when an audio signal is input is predicted based on the equivalent model (i.e., equivalent circuit) of speaker 13. That is, the displacement of speaker 13 is predicted by performing digital filtering on the equivalent circuit of speaker 13.

[0100] For example, if speaker 13 is a sealed speaker, then the equivalent circuit of speaker 13 is as follows: Figure 6 As shown.

[0101] exist Figure 6 In the example, the circuit on the left side of the figure represents the equivalent circuit of the electrical system, and the circuit on the right side of the figure represents the equivalent circuit of the mechanical system.

[0102] and, Figure 6 Each letter in the t symbol represents a parameter called a TS parameter.

[0103] In other words, "Re" represents the DC resistance (DCR) of the voice coil, "Le" represents the inductance of the voice coil, and "BL" represents the force coefficient, i.e., the BL value. The force coefficient BL is calculated by multiplying the magnetic flux density of the voice coil and magnetic circuit components by the length of the voice coil coil.

[0104] “Mms” represents the equivalent mass of the vibration system, and the equivalent mass of the vibration system Mms is the mass of the diaphragm and voice coil of the loudspeaker 13.

[0105] “Cms” indicates mechanical system compliance, which is an indication of the softness of the unit’s suspension; “Rms” indicates the mechanical resistance of the unit’s suspension; and “Cmb” indicates compliance due to the sealed suspension of speaker 13 (i.e., the sealed speaker).

[0106] The following explains how to predict the displacement of speaker 13 using these TS parameters.

[0107] Using the above TS parameters, the velocity v(s) of the loudspeaker diaphragm can be represented by the following equation (1).

[0108] [Mathematical Expression 1]

[0109]

[0110] The displacement X(s) of the loudspeaker diaphragm is obtained by integrating over the velocity v(s), and can therefore be expressed by the following equation (2).

[0111] [Mathematical Expression 2]

[0112]

[0113] Therefore, according to the above equations (1) and (2), using the TS parameter, the displacement X(s) can be expressed by the following equation (3).

[0114] [Mathematical Expression 3]

[0115]

[0116] This displacement X(s) is an analog transfer function. Using the bilinear Z-transform (s = (1-Z...),... -1 ) / (1+Z -1 The displacement X(s) is digitally filtered, and by obtaining the coefficients of the digital filter, the displacement X(s) (i.e., the analog transfer function) can be derived from... Figure 7 The representation of a third-order IIR filter is shown in the figure.

[0117] exist Figure 7In the example, the third-order IIR filter includes amplifier units 61-1 to 61-4, delay units 62-1 to 62-3, adder unit 63, delay units 64-1 to 64-3, and amplifier units 65-1 to 65-3.

[0118] In this example, the signal to be processed is provided to amplifier unit 61-1 and delay unit 62-1.

[0119] Amplifier unit 61-1 amplifies the provided signal by multiplying it by a coefficient a0 and provides the resulting signal to adder unit 63. Furthermore, delay unit 62-1 delays the provided signal and provides the delayed signal to delay unit 62-2 and amplifier unit 61-2.

[0120] Delay unit 62-2 delays the signal provided from delay unit 62-1 and provides the resulting signal to delay unit 62-3 and amplifier unit 61-3, and delay unit 62-3 delays the signal provided from delay unit 62-2 and provides the resulting signal to amplifier unit 61-4.

[0121] Amplifier units 61-2 to 61-4 amplify the signals provided from delay units 62-1 to 62-3 by multiplying the signals by coefficients a1 to a3, and provide the resulting signals to adder unit 63.

[0122] It should be noted that when there is no need to specifically distinguish between amplifier units 61-1 to 61-4, amplifier units 61-1 to 61-4 may be simply referred to as "amplifier unit 61" in the following text. Similarly, when there is no need to distinguish between delay units 62-1 to 62-3, delay units 62-1 to 62-3 may be simply referred to as "delay unit 62" in the following text.

[0123] Adder 63 adds the signals provided by amplifier units 61-1 to 61-4 and amplifier units 65-1 to 65-3, and provides the summed signal as the output of a third-order IIR filter to the subsequent stage and also to delay unit 64-1. The output of adder 63 represents the displacement of the loudspeaker.

[0124] Delay unit 64-1 delays the signal provided from adder unit 63 and provides the resulting signal to delay unit 64-2 and amplifier unit 65-1. Amplifier unit 65-1 amplifies the signal provided from delay unit 64-1 by multiplying the signal by coefficient b1 and provides the amplified signal to adder unit 63.

[0125] Delay unit 64-2 delays the signal provided from delay unit 64-1 and provides the resulting signal to delay unit 64-3 and amplifier unit 65-2, and delay unit 64-3 delays the signal provided from delay unit 64-2 and provides the resulting signal to amplifier unit 65-3.

[0126] Amplifier units 65-2 and 65-3 amplify the signals provided from delay units 64-2 and 64-3 by multiplying the signals by coefficients b2 and b3, and provide the resulting signals to adder unit 63.

[0127] It should be noted that when there is no need to distinguish between delay units 64-1 to 64-3, delay units 64-1 to 64-3 may also be referred to as "delay unit 64" in the following text. In addition, when there is no need to distinguish between amplifier units 65-1 to 65-3, amplifier units 65-1 to 65-3 may also be referred to as "amplifier unit 65" in the following text.

[0128] For example, the bilinear transformation can be used to calculate... Figure 7 The coefficients a0 to a3 and coefficients b1 to b3 used in the third-order IIR filter shown are, in other words, calculated based on the TS parameters.

[0129] Incidentally, in the TS parameters of the equivalent circuit of speaker 13, the parameters of the speaker unit (i.e., force coefficient BL, mechanical system compliance Cms, and inductance Le) vary non-linearly with respect to the displacement x of speaker 13, for example as... Figures 8 to 10 As shown.

[0130] Figure 8 The characteristics of the force coefficient BL of the loudspeaker unit as a function of displacement x are shown. That is, in Figure 8 In the diagram, the vertical axis represents the force coefficient BL, and the horizontal axis represents the displacement x.

[0131] In this example, it can be seen that the force coefficient BL decreases nonlinearly as the absolute value of the displacement x increases.

[0132] also, Figure 9 The characteristics of the mechanical system compliance Cms of the loudspeaker unit as a function of displacement x are shown. That is, in Figure 9 In the diagram, the vertical axis represents the mechanical system compliance Cms, and the horizontal axis represents the displacement x.

[0133] In this example, similar to Figure 8 It can be seen that the value of the mechanical system compliance Cms changes nonlinearly with respect to the displacement x.

[0134] Figure 10 The characteristic of the inductance Le of the loudspeaker unit changing with respect to displacement x is shown. That is, in Figure 10 In the diagram, the vertical axis represents the inductance Le, and the horizontal axis represents the displacement x.

[0135] In this example, it can be seen that as the value of displacement x increases, the inductance Le decreases nonlinearly.

[0136] In this way, the force coefficient BL, the mechanical system compliance Cms, and the inductance Le change nonlinearly.

[0137] Therefore, when predicting the displacement x that includes these nonlinear elements, the nonlinear parameters (i.e., the force coefficient BL, the mechanical system compliance Cms, and the inductance Le) can be obtained from the output displacement x. These obtained nonlinear parameters can then be used to update the coefficients of the third-order IIR filter.

[0138] In this case, for example, if filter unit 32 is composed of a third-order IIR filter, then the third-order IIR filter is configured as follows: Figure 11 As shown. It should be noted that, in Figure 11 In, corresponding to Figure 7 The components in the reference numerals are indicated by the same reference numerals, and the descriptions of those components will be omitted as appropriate.

[0139] Figure 11 The third-order IIR filter shown includes amplifier units 61-1 to 61-4, delay units 62-1 to 62-3, adder unit 63, delay units 64-1 to 64-3, amplifier units 65-1 to 65-3, and update unit 91.

[0140] exist Figure 11 In the third-order IIR filter shown, the input audio signal u[n] obtained by performing gain adjustment on the input audio signal through amplifier gain is provided to the amplifier unit 61-1 and the delay unit 62-1 that constitute the third-order IIR filter.

[0141] It should be noted that “n” in the input audio signal u[n] represents sampling, and in each of delay units 62 and 64, the provided signal is delayed for a time equivalent to one sample and then output to the subsequent stage.

[0142] The update unit 91 calculates the force coefficient BL[n], mechanical system compliance Cms[n], and inductance Le[n] for obtaining the next sampled displacement x[n] based on the displacement x[n-1] provided by the adder unit 63.

[0143] For example, the force coefficient BL[n], the mechanical system compliance Cms[n], and the inductance Le[n] can be obtained by the fourth-order approximate polynomial shown in the following equation (4).

[0144] [Mathematical Expression 4]

[0145] Bl[n] = bl4*x[n-1] 4 +bl3*x[n-1] 3 +bl2*x[n-1] 2 +bl1*x[n-1]+bl0

[0146] Cms[n] = cms4 * x[n-1] 4 +cms3*x[n-1] 3 +cms2*x[n-1] 2 +cms1*x[n-1]+cms0

[0147] Le[n] = Ie4*x[n-1] 4 +Ie3*x[n-1] 3 +Ie2*x[n-1] 2 +Ie1*x[n-1]+Ie0

[0148] ···(4)

[0149] Note that in equation (4), b10 to b14 represent zero-order to fourth-order terms in the approximate expression for the force coefficient BL, respectively. Similarly, cms0 to cms4 represent zero-order to fourth-order terms in the approximate expression for the mechanical system compliance Cms, respectively, and le0 to le4 represent zero-order to fourth-order terms in the approximate expression for the inductance Le, respectively.

[0150] The updating unit 91 performs the calculation shown in equation (4) and updates the coefficients a0 to a3 and coefficients b1 to b3 based on the force coefficient BL[n], mechanical system compliance Cms[n], and inductance Le[n] obtained as a result. Then, the updating unit 91 provides these updated coefficients to the amplifier unit 61 and the amplifier unit 65.

[0151] In this way, by having the update unit 91 calculate the force coefficient BL[n], mechanical system compliance Cms[n], and inductance Le[n] based on the immediately preceding displacement x[n-1], nonlinear displacement prediction using approximate polynomials is achieved, which makes it possible to obtain more accurate displacement x[n].

[0152] Here, refer to Figure 12 and Figure 13 This section explains the comparison between the predicted results and the actual values ​​when performing linear and nonlinear predictions of the displacement in a predetermined loudspeaker 13.

[0153] Note that in Figure 12 and Figure 13In the diagram, the vertical axis represents the displacement x[n] of the speaker 13, and the horizontal axis represents the frequency of the signal input to the speaker 13. Specifically, on the vertical axis of these diagrams, positive values ​​of displacement x[n] represent the amount of displacement towards the listening point (i.e., in the forward direction), and negative values ​​represent the amount of displacement towards the rear.

[0154] Figure 12 The predicted and actual values ​​of displacement x[n], discovered through nonlinear prediction, are shown. Specifically, in Figure 12 In the diagram, the solid curve represents the prediction result found through nonlinear prediction, and the dashed line represents the actual value. In this example, the difference between the prediction result and the actual value (prediction error) is small and independent of the signal level (i.e., the displacement of speaker 13 at each frequency), indicating that displacement x[n] can be predicted with high accuracy.

[0155] on the contrary, Figure 13 The diagram shows the predicted and actual values ​​of the displacement x[n] obtained through linear prediction. Specifically, in Figure 13 In the diagram, the solid curve represents the prediction result obtained through linear prediction, and the dashed line represents the actual value. In this example, it can be seen that the force coefficient BL, mechanical system compliance Cms, and inductance Le of loudspeaker 13 (loudspeaker unit) are highly nonlinear, and as the signal level (i.e., the displacement of loudspeaker 13) increases, the prediction result and the actual value diverge, leading to an increase in prediction error.

[0156] As can be seen from the above, for this type of loudspeaker 13 (loudspeaker unit), nonlinear prediction is required to reduce the prediction error of displacement x[n].

[0157] Note that if the loudspeaker 13 is used within a range where the force coefficient BL, mechanical system compliance Cms, and inductance Le change almost invariably with respect to displacement x[n], displacement x[n] can be obtained through linear prediction.

[0158] This corresponds to the following situation: for example, setting a low-frequency high-pass filter that truncates the input audio signal in the early stages of the displacement prediction process to attenuate the frequency band where the nonlinearity of the displacement increases, and the speaker 13 is mainly used in the almost linear frequency band.

[0159] The displacement x[n] can be predicted linearly even when the force coefficient BL, mechanical system compliance Cms, and inductance Le have low nonlinearity relative to the displacement x[n], and the loudspeaker 13 is used in the linear region.

[0160] Doppler distortion correction

[0161] Next, we will describe the Doppler distortion correction (i.e., time correction) of the input audio signal.

[0162] For example, such as Figure 14 As shown on the left, when the diaphragm D11 of the loudspeaker 13 moves forward (towards the listening point P11), the displacement x[n] is positive. In this case, the arrival time of the sound (signal) output from the loudspeaker 13 to the listening point P11 is shortened, therefore a positive displacement x[n] is required to delay the sound output time. It should be noted that in Figure 14 In, and in Figure 1 The corresponding parts in the text are represented by the same reference symbols, and the descriptions of those parts are omitted appropriately.

[0163] On the other hand, when the diaphragm D11 of the loudspeaker 13 moves backward, the displacement x[n] is negative. In this case, the arrival time of the sound (signal) from the loudspeaker 13 to the listening point P11 becomes longer, so it is necessary to advance the sound output time by a negative displacement x[n].

[0164] Therefore, in order to achieve Doppler distortion correction during playback, a delay offset can be prepared for the amount of time advance of the input audio signal, and as Doppler distortion correction, time correction can be performed centered on the offset based on the displacement amount (displacement x[n]) of the speaker 13.

[0165] Here, the time correction performed as Doppler distortion correction is the processing of a signal that is delayed or advanced in the time direction by a quantity corresponding to the displacement x[n] of the input audio signal as the corrected audio signal.

[0166] This processing can be considered as a process for obtaining the sample values ​​of the sample to be processed in the signal by performing interpolation based on the sample values ​​of multiple samples of the input audio signal, wherein the signal is generated by delaying or advancing the input audio signal in the time direction by an amount corresponding to the displacement x[n]. In other words, the time correction performed as Doppler distortion correction can be considered as a correction process for the amplitude value of the input audio signal.

[0167] The offset can be obtained by converting the maximum displacement of the diaphragm D11 of the loudspeaker 13 from distance to time using the speed of sound, and then converting it to sampling units using the sampling frequency.

[0168] Specifically, for example, assume 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].

[0169] In this case, when converted to time in terms of the speed of sound c = 340 m / s, the maximum displacement of ±10 mm becomes ±29.4 μs, and when ±29.4 μs is converted to sampling units at a sampling frequency of 48 kHz, the maximum displacement becomes ±1.4118 samples.

[0170] Therefore, in this example, the number of samples of the compensated input audio signal is two samples, and as shown on the right side of the figure, the delay circuit consisting of four delay units 121-1 to 121-4 can prepare for up to four samples (i.e., twice the offset).

[0171] Delay unit 121-1 delays the provided input audio signal by a time equivalent to one sample and provides the resulting signal to delay unit 121-2.

[0172] Furthermore, delay units 121-2 and 121-3 delay the input audio signals provided by delay units 121-1 and 121-2 by an amount equivalent to one sample time, and provide the resulting signals to delay units 121-3 and 121-4, respectively. Similarly, delay unit 121-4 delays the input audio signal provided by delay unit 121-3 by an amount equivalent to one sample time and outputs the resulting signal to the subsequent stage.

[0173] It should be noted that when there is no need to distinguish between delay units 121-1 to 121-4, delay units 121-1 to 121-4 may also be referred to as "delay unit 121" in the following text.

[0174] exist Figure 14 In the example shown on the right, a delay circuit for four samples is provided so that the time variation from 0.5882 (=2-1.4118) samples to 3.4118 (=2+1.4118) samples can be covered, and time correction corresponding to the change in displacement x[n] of the diaphragm D11 of the loudspeaker 13 is enabled.

[0175] As an interpolation process used to obtain signals with time-sampled points including values ​​below the decimal point, it achieves such time correction. For example, Lagrange interpolation can be used, which is widely used in the interpolation of oversampling filters in digital-to-analog converters (DACs) such as compact discs (CDs).

[0176] Specifically, for example, Lagrange interpolation is used to include the offset corresponding to the displacement of 0 [mm] of the loudspeaker 13, and interpolation is performed using an (n-1) order polynomial expression, where more than n points cover the maximum displacement of the loudspeaker 13 (e.g., n = 3), that is, more than n samples.

[0177] As an example, assume that the maximum displacement of the diaphragm of speaker 13 is ±10 [mm] and the sampling frequency Fs of the input audio signal is 48 [kHz].

[0178] In this case, for example, as shown in equation (5) below, interpolation can be performed using a fourth-order interpolation polynomial expression to obtain a corrected audio signal u[n] that is delayed or advanced by the time corresponding to the displacement x[n]. d [n], the fourth-order interpolation polynomial expression has five points (five samples) from order n=0 to order n=4.

[0179] [Mathematical Expression 5]

[0180]

[0181] It should be noted that in equation (5), x represents the number of correction samples, which is the correction time per sample unit corresponding to the displacement x[n]. Although an example of using Lagrange interpolation as an interpolation process is described here, the interpolation process is not limited to this, and any interpolation process can be used as long as it uses a polynomial of second order or higher, such as Newton interpolation or spline interpolation.

[0182] When the sound is reproduced by the speaker 13 according to the corrected audio signal generated by the Lagrange interpolation shown in the above equation (5), Doppler distortion is eliminated at the listening point P11, and high-quality sound is observed.

[0183] For example, as referenced Figure 3 The method described above involves generating a corrected audio signal from an input audio signal composed of a low-frequency sine wave signal with frequency f1 and a high-frequency sine wave signal with frequency f2, through Doppler distortion correction using this technology, and then reproducing the generated signal through speaker 13. Figure 15 The situation is shown in the image. It is important to note that in... Figure 15 In the diagram, the vertical axis represents the amplitude of the signal, and the horizontal axis represents the frequency.

[0184] Figure 15 The corrected audio signal obtained at the listening point P11 based on Doppler distortion correction from this technology is shown, with each frequency component of the audio signal obtained by collecting (measuring) the sound reproduced by the speaker 13 using a microphone.

[0185] In this example, similar to Figure 3 This includes the components of frequency f1 and frequency f2 contained in the original input audio signal, as well as the components of frequency (f2-f1) and frequency (f2+f1) as sidebands of frequency f2.

[0186] Specifically, in Figure 15In the diagram, the dashed portions of the frequency components (f2-f1) and (f2+f1) indicate the reduction in Doppler distortion achieved by performing Doppler distortion correction. In other words, these dashed portions represent the difference between when Doppler distortion correction is performed and when it is not. Figure 3 The difference in Doppler distortion between the cases shown in the figure.

[0187] Performing Doppler distortion correction in this way allows for the suppression of Doppler distortion and the achievement of higher quality sound reproduction.

[0188] <Example of Doppler distortion correction unit configuration>

[0189] When performing the Doppler distortion correction described above, the Doppler distortion correction unit 22 of the signal processing device 11 is configured, for example, as follows: Figure 16 As shown. It is important to note that in Figure 16 In, and in Figure 14 The corresponding parts in the text are represented by the same reference symbols, and the descriptions of those parts are omitted appropriately.

[0190] exist Figure 16 In the example shown, the Doppler distortion correction unit 22 includes delay units 121-1 to 121-4, a conversion unit 151, and an interpolation processing unit 152.

[0191] The conversion unit 151 converts the displacement x[n] provided by the filter unit 32 of the speaker displacement prediction unit 21 into the number of corrected samples x in the sampling unit corresponding to the displacement x[n], and provides the number of corrected samples x to the interpolation processing unit 152.

[0192] 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.

[0193] Delay unit 161-1 delays the displacement x[n] provided by filter unit 32 by an amount equivalent to one sample time, and provides the resulting displacement to delay unit 161-2. Delay unit 161-2 delays the displacement x[n] provided by delay unit 161-1 by an amount equivalent to one sample time, and provides the resulting displacement to multiplication unit 162.

[0194] It should be noted that when there is no need to distinguish between delay unit 161-1 and delay unit 161-2, these delay units may also be referred to as "delay unit 161" in the following text.

[0195] Multiplication unit 162 multiplies the displacement x[n] provided by delay unit 161-2 by the inverse 1 / c of ​​the speed of sound c = 340 [m / s], and provides the correction time corresponding to the displacement x[n] obtained as a result to multiplication unit 163. In other words, in multiplication unit 162, the correction time is calculated by dividing the displacement x[n] by the speed of sound c.

[0196] The multiplication unit 163 multiplies the correction time provided by the multiplication unit 162 by the sampling frequency Fs of the input audio signal and provides the correction sample number to the addition unit 164. The correction sample number is the correction time in units including the values ​​below the decimal point.

[0197] The addition unit 164 obtains the final corrected sample number x by adding the offset sample number to the corrected sample number provided from the multiplication unit 163, and provides the result to the interpolation processing unit 152. For example, in this instance, sample number 2 is added as an offset to the corrected sample number provided from the multiplication unit 163, and the result is regarded as the corrected sample number x.

[0198] The interpolation processing unit 152 performs interpolation processing based on the input audio signal u[n], the input audio signals u[n-1] to u[n-4] provided by each delay unit 121, and the correction sample number x provided by the addition unit 164, and generates a correction audio signal u. d [n].

[0199] For example, the interpolation processing unit 152 performs Lagrange interpolation by calculation as indicated by equation (5) above. The interpolation processing unit 152 then processes the corrected audio signal u obtained through the interpolation process. d [n] is provided to amplifier unit 12.

[0200] <Replay Processing>

[0201] The following will describe Figure 4 The operation of the audio playback system is shown below. In other words, reference will be made to the following text. Figure 17 The flowchart in the diagram describes the playback process performed by the audio playback system. This playback process begins when an input audio signal, serving as the source signal, is input, along with sounds indicating the content to be played back.

[0202] In step S11, amplifier unit 31 multiplies the provided input audio signal u[n] by the amplifier gain in amplifier unit 12, and provides the resulting amplified input audio signal u[n] to filter unit 32.

[0203] In step S12, the filter unit 32 uses a third-order IIR filter to filter the input audio signal u[n] provided from the amplifier unit 31, and provides the resulting shift x[n] to the delay unit 161-1 of the conversion unit 151.

[0204] For example, in filter unit 32, as referenced Figure 11 The update unit 91 calculates the above equation (4) based on the displacement x[n-1] provided by the adder unit 63, and calculates the force coefficient BL[n], mechanical system compliance Cms[n] and inductance Le[n].

[0205] Furthermore, based on the TS parameters including the obtained force coefficient BL[n], mechanical system compliance Cms[n], and inductance Le[n], update unit 91 calculates coefficients a0 to a3 and coefficients b1 to b3, and provides these coefficients to amplifier unit 61 and amplifier unit 65 respectively.

[0206] In addition, each of the delay units 62 and 64 delays the provided signal by an amount equivalent to a sampling time and outputs the resulting signal to the subsequent stage. The amplifier units 61 and 65 multiply the provided signal by a coefficient provided from the update unit 91 and provide the resulting signal to the adder unit 63.

[0207] Adding unit 63 adds the signals provided by amplifier unit 61 and amplifier unit 65 and uses the result as a displacement x[n], and provides the displacement x[n] to update unit 91 and delay unit 161-1.

[0208] In doing so, delay unit 161-1 delays the displacement x[n] provided from addition unit 63 and provides the displacement x[n] to delay unit 161-2, delay unit 161-2 delays the displacement x[n] provided from delay unit 161-1 and provides the displacement x[n] to multiplication unit 162.

[0209] This filtering in filter unit 32 results in a nonlinear prediction of the displacement x[n].

[0210] In step S13, the multiplication unit 162 obtains the correction time by multiplying the displacement x[n] provided from the delay unit 161-2 by the reciprocal of the speed of sound c, 1 / c, and provides the obtained correction time to the multiplication unit 163.

[0211] In step S14, multiplication unit 163 obtains the number of corrected samples by multiplying the correction time provided from multiplication unit 162 by the sampling frequency Fs, and provides the number of corrected samples to addition unit 164. Furthermore, addition unit 164 obtains the final number of corrected samples x by adding the offset number of samples to the number of corrected samples provided from multiplication unit 163, and provides the result to interpolation processing unit 152.

[0212] Furthermore, each delay in delay unit 121 provides the input audio signal and provides the resulting signal to delay unit 121, interpolation processing unit 152, etc. in subsequent stages.

[0213] In step S15, the interpolation processing unit 152 performs Lagrange interpolation based on the input audio signal u[n], the input audio signals u[n-1] to u[n-4] provided by each delay unit 121, and the correction sample number x provided by the addition unit 164.

[0214] In other words, the interpolation processing unit 152 performs Lagrange interpolation by calculating the above equation (5), and obtains the corrected audio signal u as a result. d [n] is provided to amplifier unit 12.

[0215] In step S16, the amplifier unit 12 transmits the corrected audio signal u provided by the interpolation processing unit 152. d [n] is multiplied by the amplifier gain to perform gain adjustment, and the adjusted corrected audio signal u is then used. d [n] is provided to speaker 13.

[0216] In step S17, the speaker 13 transmits a corrected audio signal u based on the audio signal provided from the amplifier unit 12. d [n] is driven to output sound, after which the playback process ends. In an audio playback system, the above processing is performed on each sample of the input audio signal.

[0217] In this way, the audio playback system obtains the displacement x[n] through nonlinear prediction, and obtains the corrected audio signal u by performing Lagrange interpolation using a polynomial of second order or higher based on the number of corrected samples x corresponding to the displacement x[n]. d [n]. As a result, Doppler distortion can be reduced further, enabling high-quality sound reproduction.

[0218] Although the speaker system (i.e., speaker 13) described above is an example of a sealed type, the type is not limited to this, and this technology can be applied to any speaker, such as bass-reflex type, passive radiator type, etc.

[0219] For example, if speaker 13 is a bass-reflex speaker, then the equivalent circuit of speaker 13 is as follows: Figure 18 As shown.

[0220] exist Figure 18 In the example, the circuit on the left side of the figure represents the equivalent circuit of the electrical system, and the circuit on the right side of the figure represents the equivalent circuit of the mechanical system. Figure 18 Each letter in the alphabet represents a parameter called a "TS parameter", and these TS parameters are related to... Figure 6 The parameters shown are similar.

[0221] Furthermore, for example, if the loudspeaker 13 is a passive radiator loudspeaker, then the equivalent circuit of the loudspeaker 13 is as follows: Figure 19 As shown in the image.

[0222] exist Figure 19 In the example, the circuit on the left side of the figure represents the equivalent circuit of the electrical system, and the circuit on the right side of the figure represents the equivalent circuit of the mechanical system. Figure 19 Each letter in the alphabet represents a parameter called a "TS parameter", and these TS parameters are related to... Figure 6 The parameters shown are similar.

[0223] exist Figure 18 and Figure 19 In the example shown, if a filter for displacement prediction is used, obtained by performing digital filtering through an equivalent circuit based on speaker 13, the displacement x[n] can be obtained through nonlinear prediction.

[0224] <Second Implementation Method>

[0225] <Configuration Example of an Audio Playback System>

[0226] Furthermore, although the preceding description indicates that the input audio signal (which is the source signal) is input to, for example... Figure 4 The example of speaker displacement prediction unit 21 shown is shown, but the corrected audio signal after Doppler distortion correction can be input instead.

[0227] In this case, such as Figure 20 The audio playback system is configured as shown. It should be noted that... Figure 20 In, and in Figure 4 The corresponding parts in the text are represented by the same reference symbols, and the descriptions of those parts are omitted appropriately.

[0228] exist Figure 20 The audio playback system shown 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.

[0229] Furthermore, although not shown, the loudspeaker displacement prediction unit 21 includes an amplifier unit 31 and a filter unit 32, and the Doppler distortion correction unit 22 includes delay units 121-1 to 121-4, a conversion unit 151, and an interpolation processing unit 152.

[0230] The audio playback system and Figure 4 The difference in the audio playback system shown is that the corrected audio signal output from the Doppler distortion correction unit 22 is input to the speaker displacement prediction unit 21, and in other respects... Figure 4 The audio playback system shown is the same.

[0231] Therefore, utilizing Figure 20 In the audio playback system shown, the amplifier unit 31 of the speaker displacement prediction unit 21 amplifies the corrected audio signal provided by the interpolation processing unit 152 of the Doppler distortion correction unit 22 using the amplifier gain in the amplifier unit 12, and provides the resulting signal to the filter unit 32.

[0232] The filter unit 32 performs nonlinear prediction by filtering the corrected audio signal provided from the amplifier unit 31, and provides the displacement obtained as a result of the prediction to the conversion unit 151 of the Doppler distortion correction unit 22, and more specifically, to the delay unit 161-1 of the conversion unit 151.

[0233] In this way, even when using Figure 20 The configuration shown can also reduce Doppler distortion and achieve high-quality sound reproduction, similar to... Figure 4 The situation is shown in the figure.

[0234] Although the first and second embodiments described above are examples of loudspeaker 13 being a full-range loudspeaker, this technology can also be applied to multi-channel mid-range loudspeakers, woofers, etc.

[0235] For example, when speaker 13 is a multi-channel mid-range speaker, a woofer, etc., and the bandwidth distribution filter has moderate characteristics such as 12dB / Oct, high frequencies affected by Doppler distortion are reproduced to a lesser extent. Therefore, by applying this technique and performing Doppler distortion correction, the quality of sound radiated from multi-channel speakers, etc., can be improved.

[0236] <Computer Configuration Examples>

[0237] Incidentally, the above series of processes can also be performed by hardware or software. When the series of processes is performed by software, the program constituting the software is installed in the computer. Here, the computer includes, for example, a computer integrated into dedicated hardware, a general-purpose personal computer with various programs installed to enable the computer to perform various functions, etc.

[0238] Figure 21 This is a block diagram illustrating an example of the hardware configuration of a computer that uses a program to perform the above series of processes.

[0239] In a computer, the central processing unit (CPU) 501, read-only memory (ROM) 502, and random access memory (RAM) 503 are connected to each other via a bus 504.

[0240] The input / output interface 505 is further connected to the bus 504. The input unit 506, output unit 507, storage unit 508, communication unit 509, and driver 510 are connected to the input / output interface 505.

[0241] Input unit 506 includes a keyboard, mouse, microphone, image sensor, etc. Output unit 507 includes a display, speaker, etc. Storage unit 508 consists of a hard disk, non-volatile memory, etc. Communication unit 509 includes a network interface, etc. Driver 510 drives removable storage media 511 such as disks, optical disks, magneto-optical disks, semiconductor memories, etc.

[0242] In a computer configured as described above, for example, the CPU 501 loads a program stored in the storage unit 508 into the RAM 503 via the input / output interface 505 and the bus 504 and executes the program to perform the series of processes described above.

[0243] The program executed by the computer (CPU 501) can be stored on, for example, a removable storage medium 511 as a packaging medium and provided in this state. The program can also be provided via wired or wireless transmission media (such as a local area network, the Internet, or digital satellite broadcasting).

[0244] In a computer, by installing the removable storage medium 511 in the drive 510, a program can be installed in the storage unit 508 via the input / output interface 505. Furthermore, the program can be received by the communication unit 509 via a wired or wireless transmission medium and installed in the storage unit 508. Additionally, the program can be pre-installed in the ROM 502 or the storage unit 508.

[0245] It should be noted that a program executed by a computer may be a program that processes data in the order described in this specification and executes it sequentially in time, or it may be a program that processes data in parallel or executes data at necessary time intervals, such as when it is called.

[0246] Furthermore, the implementation of this technology is not limited to the above-described implementation, and various modifications can be made without departing from the basic spirit of this technology.

[0247] For example, this technology can be configured as cloud computing, where multiple devices share and collaborate over a network to process a function.

[0248] Furthermore, each step described with reference to the aforementioned flowchart can be performed by a single device or by multiple devices in a shared manner.

[0249] Furthermore, when a single step includes multiple processes, the multiple processes included in the single step can be executed by a single device or by multiple devices in a shared manner.

[0250] In addition, this technology can also be configured as follows.

[0251] (1) A signal processing apparatus, comprising:

[0252] The displacement prediction unit predicts the displacement of the speaker diaphragm based on an audio signal that is a mixture of high-frequency and low-frequency signals when the speaker reproduces sound.

[0253] The correction unit performs time direction correction on the audio signal by performing interpolation processing using at least three samples of the audio signal, based on the displacement obtained from the prediction and the correction time obtained based on the speed of sound.

[0254] (2) According to the signal processing device in (1),

[0255] The displacement prediction unit obtains the displacement through nonlinear prediction.

[0256] (3) According to the signal processing device in (2),

[0257] The displacement prediction unit uses a polynomial approximation to perform nonlinear prediction.

[0258] (4) A signal processing device according to any one of (1) to (3),

[0259] The correction time is the delay time of the audio signal. The correction time increases when the diaphragm moves forward and decreases when the diaphragm moves backward.

[0260] (5) A signal processing device according to any one of (1) to (4),

[0261] The correction unit calculates the number of samples for the correction time based on the sampling frequency of the displacement, sound velocity, and audio signals obtained from the prediction, and performs interpolation processing based on the number of samples.

[0262] (6) According to the signal processing device in (5),

[0263] The correction unit calculates the number of samples, including values ​​below the decimal point.

[0264] (7) A signal processing device according to any one of (1) to (6),

[0265] The correction unit performs time direction correction by interpolating the sampled values ​​of the audio signal.

[0266] (8) A signal processing device according to any one of (1) to (7),

[0267] The interpolation process includes Lagrange interpolation, Newton interpolation, or spline interpolation.

[0268] (9) A signal processing device according to any one of (1) to (8),

[0269] The displacement prediction unit predicts displacement based on the audio signal obtained through interpolation.

[0270] (10) A signal processing method, comprising:

[0271] The signal processing device performs the following operations:

[0272] In the case where a loudspeaker reproduces sound based on an audio signal that is a mixture of high-frequency and low-frequency signals, the displacement of the loudspeaker diaphragm is predicted based on the audio signal; and

[0273] Based on the displacement obtained from the prediction and the correction time obtained from the sound speed, the audio signal is subjected to time direction correction by performing interpolation processing using at least three samples of the audio signal.

[0274] (11) A program that causes a computer to perform processing, the processing comprising the following steps:

[0275] In the case where a loudspeaker reproduces sound based on an audio signal that is a mixture of high-frequency and low-frequency signals, the displacement of the loudspeaker diaphragm is predicted based on the audio signal; and

[0276] Based on the displacement obtained from the prediction and the correction time obtained from the sound speed, the audio signal is subjected to time direction correction by performing interpolation processing using at least three samples of the audio signal.

[0277] [List of Reference Numbers]

[0278] 11. Signal Processing Device

[0279] 12 Amplifier Units

[0280] 13 speakers

[0281] 21 Loudspeaker displacement prediction unit

[0282] 22 Doppler distortion correction units

[0283] 31 Amplifier Unit

[0284] 32 filter units

[0285] 151 Conversion Unit

[0286] 152 Interpolation processing unit.

Claims

1. A signal processing apparatus comprising: a displacement prediction unit that predicts a displacement of a diaphragm of a speaker based on an audio signal in a case where the speaker reproduces sound based on the audio signal in which a high frequency signal and a low frequency signal are mixed, wherein the displacement prediction unit obtains the displacement by nonlinear prediction; and a correction unit that performs time direction correction on the audio signal by interpolation processing using at least three samples of the audio signal based on the displacement obtained from the prediction and a correction time obtained based on a speed of sound.

2. The signal processing apparatus according to claim 1, wherein the displacement prediction unit performs the nonlinear prediction using a polynomial approximation.

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

4. The signal processing apparatus according to claim 1, wherein the correction unit calculates a number of samples of the correction time based on the displacement obtained from the prediction, a speed of sound, and a sampling frequency of the audio signal, and performs the interpolation processing based on the number of samples.

5. The signal processing apparatus according to claim 4, wherein, the correction unit calculates the number of samples including a value after a decimal point.

6. The signal processing apparatus according to claim 1, wherein the correction unit performs the time direction correction by correcting a sample value of the audio signal by the interpolation processing.

7. The signal processing apparatus according to claim 1, wherein the interpolation processing is Lagrange interpolation, Newton interpolation, or spline interpolation.

8. The signal processing apparatus according to claim 1, wherein the displacement prediction unit predicts the displacement based on an audio signal obtained by the interpolation processing.

9. A signal processing method comprising: causing a signal processing apparatus to perform the following: predicting a displacement of a diaphragm of a speaker based on an audio signal in a case where the speaker reproduces sound based on the audio signal in which a high frequency signal and a low frequency signal are mixed, wherein the displacement is obtained by nonlinear prediction; and performing time direction correction on the audio signal by interpolation processing using at least three samples of the audio signal based on the displacement obtained from the prediction and a correction time obtained based on a speed of sound.

10. A computer-readable storage medium storing a program that, when executed, causes a computer to perform processing comprising the steps of: predicting a displacement of a diaphragm of a speaker based on an audio signal in a case where the speaker reproduces sound based on the audio signal in which a high frequency signal and a low frequency signal are mixed, wherein the displacement is obtained by nonlinear prediction; and performing time direction correction on the audio signal by interpolation processing using at least three samples of the audio signal based on the displacement obtained from the prediction and a correction time obtained based on a speed of sound.

Images