JP2010050875A - Equalizer, frequency characteristic adding method, frequency characteristic adding program and acoustic playback apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To appropriately add frequency characteristics corresponding to a user instruction to an input audio signal without deteriorating amplitude characteristics and phase characteristics. <P>SOLUTION: Frequency characteristic information in response to an instruction input accepted through a parameter input section 14 is formed by an infinite impulse response (IIR) filter 15 and on the basis of the frequency characteristic information, an FIR filter coefficient calculation section 16 calculates an FIR filter coefficient having the same amplitude characteristics as those of the frequency characteristic information, so that the phase characteristics become linear characteristics. While using the FIR filter coefficient, frequency characteristics in response to the frequency characteristic information corresponding to the user instruction input are added to an input audio signal by the FIR filter 12. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to an apparatus, a method, and a program for adding frequency characteristics to an input audio signal, which are used in various acoustic apparatuses such as a headphone system and a speaker system.

  In various apparatuses that process digital audio signals, finite impulse response filters (hereinafter referred to as FIR (Finite Impulse Response) filters) and infinite impulse response filters (hereinafter referred to as IIR (Infinite Impulse Response) filters) are used. Yes.

  The characteristics of the FIR filter are simply shown. The impulse response which is a response characteristic of the filter is “finite”, that is, it is stable to zero with a finite number of samples. For example, an Nth-order FIR filter lasts up to N + 1 samples for an impulse.

  In contrast, an IIR filter has an impulse response function that returns a non-zero value over an infinite length of time. That is, the IIR filter has internal feedback as will be described later, and may continue to respond indefinitely.

  As a technique using an FIR filter, Patent Document 1 and Patent Document 2 described later relate to a signal processing apparatus and an audio reproduction apparatus that can perform good signal processing even if an FIR filter with a short tap length is used. Technology is disclosed.

  Further, as a technique using an IIR filter, Patent Document 3 described later discloses a technique related to a sound field processing apparatus that filters an output signal of a microphone with a IIR filter in a time axis direction and a direction reverse to the time axis. Yes.

Patent Documents 1 to 3 described above are as follows.
JP 2006-276883 A JP 2006-323395 A JP 2007-043442 A

  By the way, in an equalizer device that imparts frequency characteristics to a digital audio signal, for example, an IIR filter as shown in FIG. 21 is widely used.

  The IIR filter shown in FIG. 21 includes delay circuits D1 and D2 before the adder circuit AD1 and a multiplier circuit that multiplies the variables a0, a1, and a2. Further, delay circuits D3 and D4 are provided at the subsequent stage of the adder circuit AD1, and a multiplication circuit for multiplying the respective variables −b1 and −b2 is provided.

  As shown in FIG. 21, the IIR filter incorporates feedback and can obtain a desired filter characteristic with fewer taps. The IIR filter having the configuration shown in FIG. 21 is obtained by replacing the transfer function of the filter by the analog circuit with the digital processing as it is by the SZ conversion, and realizing an equalizer curve having a stable characteristic by the digital signal processing. It has the feature of being able to do it.

  However, when such an IIR filter is used, a highly accurate and stable characteristic can be realized with respect to the amplitude frequency characteristic. However. When the IIR filter is used, there is a problem that the phase frequency characteristic changes from the vicinity of the change frequency as in the analog filter, and waveform transmission distortion occurs.

  For example, FIG. 22 shows amplitude frequency characteristics (upper side) and group delay characteristics (lower side) when a peak filter having a center frequency of 1 kHz is realized by an IIR filter. As shown in FIG. 22, it can be seen that the group delay characteristic is delayed around the center frequency (1 kHz). Further, when the impulse response at this time is shown, it becomes as shown in FIG. 23, and the impulse response does not become steep.

  FIG. 24 shows a state where a sweep sign signal is input to the IIR filter having the characteristics as shown in FIGS. FIG. 24A shows a sweep sine wave input to the IIR filter, and FIG. 24B shows an output signal from the IIR filter.

  As can be seen from FIG. 24, in the case of the IIR filter, the input sine wave (FIG. 24A) generates waveform transmission distortion near the peak frequency, and as a result, as shown in FIG. A waveform with a delayed phase is output.

  Here, an example in which a sweep sine wave is used as an input signal is shown. However, even when actual musical sound data is processed, if signals including various spectrums are input to the IIR filter at the same time, the spectrum of the band in which the group delay is generated by the IIR filter will be delayed, and the sound quality and sound image will be delayed. This will cause the localization to deteriorate.

  As described above, as a technique for improving the waveform transmission distortion due to the phase rotation in the equalizer process when the IIR filter is used, there is a method using an FIR filter having a configuration as shown in FIG.

  The FIR filter shown in FIG. 24 includes a delay circuit d1, d2,... Dn-1, dn, a multiplier that multiplies each variable h0, h1, h2,..., Hn-1, hn, and an adder ad1, , adn-1, adn.

  By using the FIR filter having such a configuration, it is possible to set the phase characteristic as well as the amplitude characteristic to the linear phase characteristic, and it is possible to avoid waveform transmission distortion as in the case of using the IIR filter. .

However, when such an FIR filter is used, there are the following problems. (1) Calculation of filter coefficients is not easy.
(2) It is difficult to achieve consistency with the frequency characteristics of a conventional analog circuit compatible equalizer used in a general graphic equalizer.
(3) In order to generate a constant group delay over the entire band, a time delay is generated in the output signal.
It is a problem.

  In particular, the problem (3) also causes a problem that the synchronization between video and audio is impaired when used for an audio signal accompanied by a video signal, such as movie content.

  As described above, in the equalizing method in the digital signal processing, in the equalizer processing using the IIR filter, phase rotation or group delay is generated in the vicinity of the change frequency. For this reason, there is a problem in that waveform transmission distortion occurs by generating a time delay by a frequency in a specific band.

  In addition, when performing equalizer processing using an FIR filter, it is not easy to determine a coefficient for ensuring consistency with a graphic equalizer using a conventional analog circuit, or time delay increases due to group delay occurring in the entire band. There is a problem such as.

  In view of the above, the present invention eliminates the above problems and appropriately adds (provides) frequency characteristics according to user instructions to the input audio signal without degrading amplitude characteristics and phase characteristics. The purpose is to be able to.

In order to solve the above-mentioned problem, an equalizer device according to a first aspect of the present invention provides:
An accepting means for accepting an instruction input from the user regarding the frequency characteristics;
Characteristic information forming means for forming frequency characteristic information in response to the instruction input from the receiving means;
Based on the frequency characteristic information from the characteristic information forming means, adjustment information calculation means for calculating adjustment information having the same amplitude characteristic as the frequency characteristic information and controlling the phase characteristic to a linear phase;
Frequency characteristic adding means for adding a frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming means to the input audio signal using the adjustment information from the adjustment information calculating means.

  According to the equalizer apparatus of the first aspect of the present invention, the frequency characteristic information corresponding to the instruction input received through the receiving unit is formed by the characteristic information forming unit. Based on this frequency characteristic information, the adjustment information calculating means calculates adjustment information that has the same amplitude characteristic as the frequency characteristic information and that makes the phase characteristic a linear characteristic.

  Then, using the calculated adjustment information, a frequency characteristic corresponding to the frequency characteristic information corresponding to the user's instruction input is added to the input audio signal by the frequency characteristic adding means.

  In this way, when providing frequency characteristics to the input audio signal, adjustment information that has the same amplitude characteristics as the frequency characteristics information according to the user's instruction input and has been adjusted so that the phase characteristics are linear characteristics Is calculated. Using this adjustment information, a frequency characteristic corresponding to a user instruction is added to the input audio signal. Thus, it is possible to appropriately add a frequency characteristic according to a user instruction to the input audio signal without degrading the amplitude characteristic and the phase characteristic.

An equalizer device according to a second aspect of the invention is the equalizer device according to the first aspect,
The frequency characteristic adding means is a finite impulse response filter that uses the adjustment information to add a frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming means by a convolution operation.

  According to the equalizer device of the second aspect, the frequency characteristic imparting means is configured using a finite impulse response filter.

  Thereby, adjustment information (filter coefficient) can be set appropriately for the finite impulse response filter, and the adjustment information is appropriately adjusted in both amplitude characteristics and phase characteristics.

  Therefore, even when the adjustment information is used, a phase delay is not generated for the input audio signal. That is, the frequency characteristic according to the user instruction can be appropriately added to the input voice signal without degrading the amplitude characteristic and the phase characteristic.

An equalizer device according to a third aspect of the invention is the equalizer device according to the first aspect,
Characteristic orthogonal transforming means for transforming the frequency characteristic information from the characteristic information forming means into frequency characteristic information of a frequency axis region;
An input signal orthogonal transforming means for converting the input voice signal in the time axis domain into an input voice signal in the frequency axis domain;
An inverse orthogonal transforming means for transforming an output voice signal in the frequency axis region from the frequency characteristic adding means into a signal in the time axis region;
The adjustment information calculation unit calculates the adjustment information of the frequency axis region based on the frequency characteristic information of the frequency axis region from the characteristic orthogonal transform unit,
The frequency characteristic adding unit multiplies the input audio signal in the frequency axis region from the input signal orthogonal transform unit by the adjustment information in the frequency axis region from the adjustment information calculation unit, thereby forming the characteristic information Multiplication means for adding a frequency characteristic corresponding to the frequency characteristic information formed by the means.

  According to the equalizer apparatus of the third aspect of the present invention, both the adjustment information calculating means and the frequency characteristic adding means perform signal processing in the frequency axis region. For this purpose, characteristic orthogonal transforming means and input signal orthogonal transforming means are provided, so that the input voice signal and the adjustment information, which are frequency axis region information, can be supplied to the frequency characteristic adding means. To be.

  The frequency information adding unit is configured as a multiplying unit, and the input audio signal is multiplied by the adjustment information, whereby a frequency characteristic corresponding to the user instruction is given to the input audio signal. Further, an output audio signal (a signal to which a frequency characteristic according to a user instruction is added) from the frequency characteristic adding means which is a frequency axis domain signal is converted into a time axis domain signal by an inverse orthogonal means.

  As a result, in the frequency axis region, it is possible to add a frequency characteristic corresponding to a user instruction to the input audio signal. Also in this case, the frequency characteristic according to the user instruction can be appropriately added to the input voice signal without degrading the amplitude characteristic and the phase characteristic.

An equalizer device according to a fourth aspect of the invention is the equalizer device according to the third aspect,
Impulse generating means for generating an impulse signal at a constant level per unit time,
The characteristic information forming unit adds the amplitude information and phase information corresponding to the frequency characteristic information to the impulse signal from the generating unit and outputs the impulse.

  According to the equalizer apparatus of the fourth aspect of the present invention, the impulse signal generated by the impulse generator is supplied to the characteristic information forming means, and the frequency characteristic according to the user instruction is given to the impulse signal. Is output.

  As a result, an impulse signal to which a characteristic according to the frequency characteristic information formed by the characteristic information forming unit is given can be output and supplied to the adjustment information calculating unit. That is, the frequency characteristic information formed by the characteristic information forming unit by a simple method can be supplied to the adjustment information calculating unit. Accordingly, it is possible to appropriately add a frequency characteristic according to a user instruction to the input audio signal without degrading the amplitude characteristic and the phase characteristic.

  According to the present invention, it is possible to appropriately add a frequency characteristic according to a user instruction to an input voice signal without degrading the amplitude characteristic and the phase characteristic.

  Hereinafter, an embodiment of the present invention will be described with reference to the drawings. In the embodiment described below, a case where a digitized audio signal (digital audio signal) is processed will be described. However, even when analog audio signals are to be processed, it is possible to perform exactly the same processing by first performing A / D (Analog / Digital) conversion processing.

<First Embodiment>
FIG. 1 is a block diagram for explaining an equalizer apparatus 1 according to a first embodiment to which an embodiment of the apparatus, method and program of the present invention is applied. As shown in FIG. 1, the equalizer device 1 includes a digital audio signal input end 11, an FIR filter 12, and a digital audio signal output end 13 as a main signal processing system.

  Further, the equalizer device 1 includes a parameter input unit 14, an IIR filter 15, and an FIR filter coefficient calculation unit 16 as a filter coefficient processing system for setting filter coefficients for the FIR filter.

  In the equalizer device 1, the parameter input unit 14 realizes a function as an accepting unit, and the IIR filter 15 realizes a function as a characteristic information forming unit. Further, the FIR filter coefficient calculation unit 16 realizes a function as adjustment information calculation means, and the FIR filter 12 realizes a function as frequency characteristic addition means.

  The parameter input unit 14 receives an instruction input related to frequency characteristics to be given to the digital audio signal from the user, and supplies the instruction input to the IIR filter 15 as an electric signal.

  Here, the parameter input unit 14 is like a so-called graphic equalizer that specifies the gain and Q value for each frequency band. However, the present invention is not limited to this, and one that selects a desired characteristic from a plurality of preset equalizer curves (information on frequency characteristics), or only one equalizer curve that is preset in advance without providing a selection function. It may be something that only gives.

  The IIR filter 15 forms frequency characteristic information (equalizer curve) to be given to the digital audio signal in response to an instruction input regarding the frequency characteristic from the parameter input unit 14, and supplies this to the FIR filter coefficient calculation unit 16.

  Specifically, the IIR filter 15 is configured as a filter having amplitude characteristics designated by overall by one or a plurality of IIR filters. For example, the IIR filter 15 is obtained by digitizing the transfer function of an analog circuit by bilinear conversion, and has a characteristic close to that of an analog circuit.

  In the case where the IIR filter as the characteristic information forming unit is configured using a plurality of IIR filters, for example, it can be realized by cascading secondary IIR filters as shown in FIG. In this case, equalizers with different characteristics can be realized simultaneously with a plurality of IIR filters, so that the amplitude accuracy in the portion where the equalizer characteristics overlap is also faithfully reproduced.

  For example, in the method in which the frequency characteristic information from each IIR filter is linearly phased and then the amplitude value is added, the change in level due to the phase around is not taken into account in the part where the envelopes of the frequency characteristic information overlap. The amplitude frequency characteristics of

  However, in the equalizer device 1 according to the first embodiment to which the present invention is applied, the IIR filter 15 has a configuration in which the IIR filters are connected in cascade, so that the total frequency characteristic information can be linearly phased. it can. Thereby, even in the part where the envelopes as described above of the frequency characteristic information overlap, the frequency characteristic does not shift.

  Based on the frequency characteristic information from the IIR filter 15, the FIR filter coefficient calculation unit 16 performs an amplitude equivalent process and a phase equivalent process to form a filter coefficient (adjustment information) to be supplied to the FIR filter 12. The filter 12 is supplied.

  Specifically, the FIR filter coefficient calculation unit 16 includes an amplitude equivalent processing unit 161 and a phase equivalent coefficient generator 162 as shown in FIG.

  Based on the frequency characteristic information from the IIR filter 15, the amplitude equivalent processing unit 161 calculates an amplitude equivalent coefficient that causes the amplitude characteristic to be the same as the frequency characteristic information from the IIR filter 15.

  The phase equivalent coefficient generator 162 performs linear phase processing of the phase characteristic on the frequency characteristic information from the IIR filter 15 to control the phase characteristic of the digital audio signal to which the frequency characteristic is given to a linear phase. Is calculated.

  The amplitude equivalent coefficient calculated by the amplitude equivalent processing unit 161 and the phase equivalent coefficient calculated by the phase equivalent coefficient generator 162 are supplied to the FIR filter 12 as filter coefficients.

  The FIR filter 12 convolves the digital audio signal supplied through the input end 11 with the filter coefficient from the FIR filter coefficient calculation unit 16. As a result, it is possible to add frequency characteristics corresponding to frequency characteristic information (equalizer curve) corresponding to a user instruction that has been linearized to the digital audio signal from the input end 11.

  The digital audio signal to which the frequency characteristic according to the user instruction is added in the FIR filter 12 is output through the output terminal 13.

  FIG. 2 is a diagram illustrating the characteristics of the FIR filter 12 when the equalizer device 1 has 1 kHz as the center frequency. In FIG. 2, the upper graph shows the amplitude frequency characteristic, and the lower graph shows the group delay characteristic. FIG. 3 is a diagram showing the state of the impulse response of the FIR filter 12 at this time.

  As can be seen from the graph of the amplitude frequency characteristic shown in the upper part of FIG. 2, it can be seen that the amplitude characteristic can be satisfactorily expressed at the center frequency. Further, as can be seen from the graph of the group delay characteristic shown in the lower part of FIG. 2, the group delay characteristic sticks to 0 milliseconds (23 milliseconds in this example), and the disturbance of the group delay characteristic is It can be seen that no waveform transmission distortion occurs.

  Further, as can be seen from the state of the impulse response in FIG. 3, the impulse response from the FIR filter 12 of the equalizer device 1 also has a steep characteristic, and it is understood that the phase is not disturbed. The

  As described above, the equalizer device 1 according to the first embodiment uses the filter coefficients from the FIR filter coefficient calculation unit 16 to appropriately input parameters to the input audio signal without degrading the amplitude characteristics and the phase characteristics. The frequency characteristic according to the user instruction received through the unit 14 can be given.

[Modification 1 of the first embodiment]
Whereas the equalizer device 1 shown in FIG. 1 performs signal processing in the time axis region, the equalizer device 2 of the first modification performs signal processing in the frequency axis region.

  FIG. 4 is a block diagram for explaining a configuration of the equalizer device 2 according to the first modification of the first embodiment. As shown in FIG. 4, the equalizer device 2 includes a digital audio signal input end 11, a digital audio signal output end 13, a parameter input unit 14, and an IIR filter 15, as shown in FIG. 1. This is the same as in the case of the equalizer device 1.

  In the case of the equalizer device 2 according to the first modification, as shown in FIG. 4, an orthogonal transform unit 21, a multiplier 22, an inverse orthogonal transform unit 23, an orthogonal transform unit 24, and an adjustment information calculation unit. 25.

  Therefore, also in the equalizer device 2, the parameter input unit 14 realizes a function as an accepting unit, and the IIR filter 15 realizes a function as a characteristic information forming unit.

  In the equalizer device 2 shown in FIG. 4, the adjustment information calculation unit 25 realizes a function as an adjustment information calculation unit that calculates the adjustment information in the frequency axis region. Further, in the equalizer device 2, the multiplier 22 realizes a function as frequency characteristic adding means for giving a frequency characteristic according to a user instruction to the digital audio signal in the frequency axis region.

  Further, the orthogonal transform unit 21 realizes a function as an input signal orthogonal transform unit, the orthogonal transform unit 24 realizes a characteristic orthogonal transform unit, and the inverse orthogonal transform unit 23 realizes an inverse orthogonal transform unit.

  In the equalizer device 2 shown in FIG. 4, the digital audio signal input through the input terminal 11 is converted into a signal in the frequency axis region by the orthogonal transform unit 21 and supplied to the multiplication unit 22.

  On the other hand, the frequency characteristic information corresponding to the user instruction from the IIR filter 15 is converted into a signal in the frequency axis region by the orthogonal transform unit 24 and supplied to the adjustment information calculation unit 25.

  The adjustment information calculation unit 25 performs an amplitude equivalent process and a phase equivalent process on the frequency characteristic information that is a signal in the frequency axis region, and responds to a user instruction with respect to the digital audio signal that is a signal in the frequency axis region. Adjustment information for adding frequency characteristics is calculated.

  Specifically, the adjustment information calculation unit 25 includes an amplitude equivalent processing unit 251 and a phase equivalent coefficient generator 252 as shown in FIG.

  The amplitude equivalence processing unit 251 performs amplitude equivalence processing based on the frequency characteristic information that is the signal in the frequency axis region from the orthogonal transform unit 24 so that the amplitude characteristic becomes the same as the frequency characteristic information from the IIR filter 15. An amplitude equivalent coefficient to be calculated is calculated.

  The phase equivalent coefficient generator 252 performs a linear phase conversion process (phase equivalent process) of the phase characteristic on the frequency characteristic information that is a signal in the frequency axis region from the orthogonal transform unit 24, and the digital to which the frequency characteristic is given. A phase equivalent coefficient for controlling the phase characteristic of the audio signal to a linear phase is calculated.

  The amplitude equivalent coefficient calculated by the amplitude equivalent processing unit 251 and the phase equivalent coefficient calculated by the phase equivalent coefficient generator 252, which are signals in the frequency axis region, are supplied to the multiplier 22 as adjustment information. .

  As described above, the multiplier 22 is supplied with the digital audio signal converted into the signal in the frequency axis region by the orthogonal transform unit 21. The adjustment information from the adjustment information calculation unit 25 is supplied to the digital audio signal. Is multiplied. As a result, the multiplier 22 forms a digital audio signal in the frequency axis region to which the frequency characteristic according to the user instruction is added.

  The digital audio signal in the frequency axis region from the multiplier 22 is converted into a digital audio signal in the time axis region by being subjected to inverse orthogonal transform processing in the inverse orthogonal transform unit 23, and this is output through the output terminal 13.

  As described above, in the equalizer device 2 according to the first modification of the first embodiment shown in FIG. 4, the frequency characteristic corresponding to the user instruction is added to the digital audio signal in the frequency axis region. Can be output after returning to the time domain signal.

  In the equalizer device 2 according to the first modification of the first embodiment, the adjustment information from the adjustment information calculation unit 25 is used to appropriately adjust the input audio signal without degrading the amplitude characteristic and the phase characteristic. The frequency characteristic according to the user instruction received through the parameter input unit 14 can be given.

[Modification 2 of the first embodiment]
The equalizer device 3 according to the second modification of the first embodiment is capable of acquiring frequency characteristic information formed in the IIR filter 15 by a simple method and supplying it to a subsequent processing unit.

  FIG. 5 is a block diagram for explaining a configuration of the equalizer device 3 according to the second modification of the first embodiment. As can be seen from a comparison between FIG. 5 and FIG. 4, the equalizer device 3 of the second modification is the modification shown in FIG. 4 except that the equalizer 3 includes an impulse generator 31 and a window function processing unit 32. 2 is configured in the same manner as the equalizer device 2 of FIG.

  For this reason, in the equalizer device 3 shown in FIG. 5, the same reference numerals are given to the same components as those of the equalizer device 2 shown in FIG. I will do it.

  The impulse generator 31 of the equalizer device 3 generates an impulse signal at a constant level per unit time. That is, the impulse generator 31 realizes a function as impulse generating means.

  The impulse signal generated in the impulse generator 31 is supplied to the IIR filter 15. As described above, the IIR filter 15 forms frequency characteristic information in accordance with a user instruction input from the parameter input unit 14.

  For this reason, the IIR filter 15 forms an impulse signal (impulse response) including amplitude information and phase information of frequency characteristics according to a user's instruction, and this is supplied to the orthogonal transform unit 24 through the window function processing unit 32. .

  The window function processing unit 32 sets the impulse signal from the IIR filter 15 according to the size of the operation frame in the multiplier 22. The orthogonal transform unit 24 converts an impulse signal including amplitude information and phase information corresponding to a user instruction supplied thereto into a signal in the frequency axis region, and supplies the signal to the adjustment information calculation unit 25.

  As described above, the adjustment information calculation unit 25 calculates the amplitude equivalent coefficient calculated by the amplitude equivalent processing unit 251 and the phase equivalent coefficient calculated by the phase equivalent coefficient generator 252 which are signals in the frequency axis region. The adjustment information is supplied to the multiplier 22.

  As a result, the multiplier 22 multiplies the digital audio signal in the frequency axis region by the adjustment information, adds a frequency characteristic according to the user instruction to the digital audio signal, and supplies this to the inverse orthogonal transform unit 23. Supply.

  As described above, the inverse orthogonal transform unit 23 converts the digital audio signal in the frequency axis domain supplied thereto into a digital audio signal in the time axis domain, and outputs this.

  In the equalizer device 3 having such a configuration, a case where FFT (Fast Fourier Transform) is used as the orthogonal transform in the orthogonal transform units 21 and 24 will be considered.

Here, the n-point complex data obtained by FFT is
xr [m] + jxi [m] (m = 0, 1,..., n−1)
In the amplitude equivalent processing unit 251 of the adjustment information calculating unit 25,
abs [m] = √ (xr [m] ² + xi [m] ² ) (Formula 1)
An amplitude value (amplitude characteristic) abs [m] is calculated.

  Thus, the amplitude value abs [m] is obtained as the square root of the sum of the square of xr [m] and the square of xi [m].

Further, in the phase equivalent coefficient generator 252 of the adjustment information calculation unit 25,
xr [m] = abs [m] * cos (−PI * m) (Formula 2)
xi [m] = abs [m] * sin (−PI * m) (Formula 3)
Real part and imaginary part data are calculated. In (Expression 2) and (Expression 3), the character “PI” means “π” in the arc degree method, and the character “*” means multiplication.

  The adjustment information obtained in this way is used as adjustment information for giving a frequency characteristic of a user instruction to the digital audio signal in the multiplier 22. Thereby, the equalizer device 3 having the same amplitude characteristic as the IIR filter and having the linear phase characteristic can be realized with a simple configuration.

[Other Modifications of First Embodiment]
It is also possible to configure an equalizer device in which an impulse generator 31 is provided in front of the IIR filter 15 of the equalizer device 1 shown in FIG. 1 or the equalizer device 2 shown in FIG.

  Further, in the equalizer device 1 shown in FIG. 1, the FIR filter coefficient calculation unit 16 can be replaced with the adjustment information calculation unit 25 described with reference to FIG. In this case, an orthogonal transform unit may be provided between the IIR filter 15 and the adjustment information calculation unit 25, and an inverse orthogonal transform unit may be provided between the adjustment information calculation unit 25 and the FIR filter 12.

  Further, in the equalizer apparatus having the configuration shown in FIGS. 1, 4, and 5, the functions realized by the respective units are realized by a program, and the computer mounted on the audio apparatus is caused to execute the program. It is also possible to configure the equalizer device.

<Second Embodiment>
Next, an equalizer apparatus according to a second embodiment to which an embodiment of the apparatus, method, and program of the present invention is applied will be described. The equalizer device according to the second embodiment, which will be described in detail below, can more easily perform the process in the adjustment information calculation unit.

  FIG. 6 is a block diagram for explaining the equalizer device 4 according to the second embodiment. The equalizer device 4 shown in FIG. 6 is obtained by changing the multiplier 22 to the FIR filter 12 in the equalizer device 3 described as the modification 2 of the first embodiment shown in FIG.

  Further, as apparent from comparison between FIG. 6 and FIG. 4, in the equalizer device 4 of the second embodiment shown in FIG. 6, the orthogonal transform unit 21 and the inverse orthogonal transform unit 23 are not used. ing.

  Further, the equalizer device 4 illustrated in FIG. 6 includes an adjustment information calculation unit 41 that is different in processing content from the adjustment information calculation unit 25 of the first embodiment. Further, the equalizer device 4 illustrated in FIG. 6 includes an inverse orthogonal transform unit 42 and a data transposition unit 43 between the adjustment information calculation unit 41 and the FIR filter 12.

  For this reason, in the equalizer device 4 shown in FIG. 6, the same reference numerals are given to the parts configured in the same manner as the equalizer device 3 shown in FIG. To do.

  That is, in the equalizer device 4 of the second embodiment, the orthogonal transform unit 24 realizes a function as an orthogonal transform unit, and the inverse orthogonal transform unit 42 realizes a function as an adjustment information inverse orthogonal transform unit, and data The transposition unit 43 realizes a function as transposition means.

  The adjustment information calculation unit 41 realizes a function as adjustment information calculation means, and the FIR filter 12 realizes a function as frequency characteristic addition means.

  Further, the parameter input unit 14 realizes a function as an accepting unit, the IIR filter 15 realizes a function as a characteristic information forming unit, and the orthogonal transformation unit 24 realizes a function as a characteristic orthogonal transformation unit. Further, the impulse generator 31 realizes a function as impulse generating means.

  In the equalizer device 4 of the second embodiment shown in FIG. 6, the adjustment information calculation method in the adjustment information calculation unit 41 is the adjustment information calculation unit of the equalizer devices 2 and 3 of the first embodiment. It is simpler than 25.

  The adjustment information calculation unit 41 of the equalizer device 4 according to the second embodiment also includes an amplitude equivalent processing unit 411 and a phase equivalent coefficient generator 412.

The amplitude equivalent processing unit 411 of the adjustment information calculating unit 41 calculates an amplitude value equivalent to the frequency characteristic information from the IIR filter 15 as in the case of the amplitude equivalent processing unit 251 of the equalizer device 3 described with reference to FIG. This is used as real part data. That is, in the amplitude equivalent processing unit 411,
xr [m] = (xr [m] ² + xi [m] ² ) (Formula 4)
And an amplitude value (amplitude coefficient) is obtained.

On the other hand, in the phase equivalent coefficient generator 412 of the adjustment information calculation unit 41,
xi [m] = 0 (Formula 5)
And this is assumed to be imaginary part data. These real part (xr [m]) and imaginary part (xi [m]) are used as adjustment information.

  The adjustment information calculated in this way is converted into an impulse response in the time axis region by the inverse orthogonal transform unit 42. However, at this time, the impulse response is in a state where the first half and the second half are transposed by the processing of (Equation 4) and (Equation 5) described above, and does not have a desired linear phase characteristic.

  Therefore, the impulse response from the inverse orthogonal transform unit 42 transposes (replaces) the first half and the second half in the time axis direction in the data transposition unit 43. The filter coefficients in the form of impulse responses thus calculated are supplied to the FIR filter 12.

  Thereby, in the FIR filter 12, the filter coefficient is convoluted with the digital audio signal from the input end 11, and the frequency characteristic according to the user's instruction is added. That is, the FIR filter 12 can be used as a linear phase filter in the time axis region.

  Therefore, also in the case of this equalizer device 4, an equalizer device having the same amplitude characteristic as that of the IIR filter and having a linear phase characteristic can be realized with a simple configuration.

  FIG. 7 is a diagram for explaining processing of the data transposing unit 43 of the equalizer device 4 shown in FIG. Assume that the state of the impulse response shown in FIG. 7A is the original adjustment information in the time axis region. That is, in FIG. 7, the horizontal axis is the time axis and the vertical axis is the amplitude.

  However, as shown in (Equation 4) and (Equation 5), in the adjustment information calculation unit 41 of the equalizer device 4, the imaginary part is set to 0 (zero) in the phase equivalent coefficient generator 412. Not properly considered.

  For this reason, in the equalizer device 4 shown in FIG. 6, the adjustment information (impulse response), which is a time-axis domain signal from the inverse orthogonal transform unit 42, has an original shape ( In FIG. 7A, the first half and the second half are reversed in the time axis direction.

  Therefore, in the equalizer device 4 shown in FIG. 6, as shown in FIG. 7C, the adjustment information from the inverse orthogonal transform unit 42 is replaced (transposed) between the first half and the second half in the time axis direction. Process). As a result, the phase is also correctly taken into consideration, and a desired linear phase characteristic is realized.

  As shown in FIGS. 7B and 7C, the original second half indicated by reference sign B (the first half in FIG. 7B and the original first half indicated by reference sign A (FIG. 7). Replacing the latter part in (B) is called “transposition” in this specification.

  As described above, also in the equalizer device 4 of the second embodiment, while the processing in the adjustment information calculation unit 41 is simplified, an appropriate amplitude characteristic is given according to an instruction from the user, and the linear phase characteristic is also provided. The equalizer device 4 having the above can be realized with a simple configuration.

[Modification of Second Embodiment]
Whereas the equalizer device 4 shown in FIG. 6 performs signal processing in the time axis region, the equalizer device 5 according to this modification performs signal processing in the frequency axis region.

  FIG. 8 is a block diagram for explaining a configuration of an equalizer device 5 according to a modification of the second embodiment. As shown in FIG. 8, the equalizer device 5 has the same configuration as the equalizer device 4 shown in FIG. 6 except that the equalizer device 5 includes an orthogonal transform unit 51, a multiplication unit 52, an inverse orthogonal transform unit 53, and an orthogonal transform unit 54. It has been done.

  For this reason, in the equalizer device 5 shown in FIG. 8, the same reference numerals are given to the parts configured in the same manner as the equalizer device 4 shown in FIG.

  In FIG. 8, each of the orthogonal transform unit 51, the multiplying unit 52, and the inverse orthogonal transform unit 53 is an orthogonal transform unit in the equalizer device 2 of Modification 1 of the first embodiment described with reference to FIG. 4. 21, the multiplier 22, and the inverse orthogonal transform unit 23.

  Therefore, in the equalizer device 5 of the modified example of the second embodiment, the inverse orthogonal transform unit 42 realizes the function of the adjustment information inverse orthogonal transform unit, the data transposing unit 43 realizes the function of the transposing unit, The orthogonal transform unit 54 realizes a function as adjustment information orthogonal transform means.

  The adjustment information calculation unit 41 realizes a function as adjustment information calculation means, and the multiplier 52 realizes a function as frequency characteristic addition means.

  In the equalizer device 5 shown in FIG. 8, the digital audio signal input through the input terminal 11 is converted into a signal in the frequency axis domain by the orthogonal transform unit 511 and supplied to the multiplier unit 52.

  As in the case of the equalizer device 4 shown in FIG. 6, the adjustment information calculated by the adjustment information calculation unit 41 is subjected to inverse orthogonal transformation by the inverse orthogonal transformation unit 42 to be adjustment information in the time axis region.

  As described with reference to FIG. 7, the time axis region adjustment information from the inverse orthogonal transform unit 42 is transposed (replaced) in the first half and the second half in the time axis direction as described with reference to FIG. 7. The adjustment information after processing is supplied to the orthogonal transform unit 54.

  The orthogonal transform unit 43 converts the time axis domain adjustment information supplied thereto again into frequency axis domain adjustment information, which is supplied to the multiplier 52.

  As described above, the multiplier 52 is supplied with the digital audio signal converted into the signal in the frequency axis region by the orthogonal transform unit 51, and the adjustment information from the adjustment information calculation unit 41 is supplied to the digital audio signal. Is multiplied. Thereby, the multiplier 52 forms a digital audio signal in the frequency axis region to which the frequency characteristic according to the user instruction is added.

  The digital audio signal in the frequency domain from the multiplier 52 is subjected to inverse orthogonal transform processing in the inverse orthogonal transform unit 53 to be converted into a digital audio signal in the time axis domain, which is output through the output terminal 13.

  As described above, in the equalizer device 5 according to the modification of the second embodiment shown in FIG. 8, the frequency characteristic corresponding to the user instruction is added to the digital audio signal in the frequency axis region. It can be output after returning to the signal in the time domain.

  And also in the equalizer device 5 of the modified example of the second embodiment, the adjustment information from the adjustment information calculation means 41 is appropriately applied to the input audio signal without degrading the amplitude characteristic and the phase characteristic. A frequency characteristic corresponding to a user instruction received through the parameter input unit 14 can be given.

  As described above, in the equalizer device 5 according to the modification of the second embodiment, while the processing in the adjustment information calculation unit 41 is simplified, an appropriate amplitude characteristic is given according to an instruction from the user, The equalizer device 5 having the linear phase characteristic can be realized with a simple configuration.

  In the case of the equalizer devices 4 and 5 of the second embodiment shown in FIGS. 6 and 8, the equalizer devices 1 and 2 of the first embodiment shown in FIGS. 1, 4, and 5 are used. 3 is 0 (zero) in the frequency domain of the phase term in the phase equivalent coefficient generator.

  For this reason, the calculation which calculates | requires the coefficient in the frequency area | region which leads to (Formula 1)-(Formula 3) mentioned above becomes easy, and the load of the process in the adjustment information calculation part 41 can be reduced.

  Further, in the equalizer device having the configuration shown in FIGS. 6 and 8, the function realized by each unit is realized by a program, and the computer mounted on the audio device is caused to execute the program, thereby the equalizer device of the present invention. Can also be configured.

<Third Embodiment>
Next, an equalizer apparatus according to a third embodiment to which the apparatus, method and program of the present invention are applied will be described. In the equalizer device of the third embodiment, as in the case of the equalizer devices 4 and 5 of the second embodiment described above, the adjustment information is simplified by setting the imaginary term (phase term) to 0 (zero). Is what you want.

  Furthermore, the equalizer device according to the third embodiment appropriately grasps the amplitude characteristic according to the instruction input from the user for each frequency band divided in advance, and based on this, for the digital audio signal, A frequency characteristic corresponding to a user instruction is added.

  FIG. 9 is a block diagram for explaining the equalizer device 6 according to the third embodiment. 9 and FIG. 6, the equalizer device 6 shown in FIG. 9 is different from the equalizer device 4 of the second embodiment shown in FIG. The configuration is different.

  That is, the equalizer device 6 of FIG. 9 includes a digital audio signal input end 11, an FIR filter 12, a digital audio signal output end 13, an adjustment information calculation unit 41, an inverse orthogonal transform unit 42, and a data transposition unit 43. This is the same as the equalizer device 4 shown in FIG.

  For this reason, in the equalizer device 6 shown in FIG. 9, the same reference numerals are given to the parts configured in the same manner as the equalizer device 4 shown in FIG. .

  In the equalizer device 6 according to the third embodiment shown in FIG. 9, the parameter input unit 14 realizes a function as an accepting unit. And in the equalizer apparatus 6, the parameter input part 14 can receive the input of the information regarding the target frequency characteristic for every frequency band divided | segmented previously.

  Further, as will be described in detail later, a portion including the IIR filter unit 61, the spectrum amplitude calculation unit 62, and the adder 63 realizes a function as characteristic information forming means.

  As shown in FIG. 9, the IIR filter unit 61 is provided with a plurality of IIR filters for each frequency band that is divided in advance. Specifically, the IIR filter unit 61 includes three IIR filters 611, 612, and 613.

  The spectrum amplitude calculation unit 62 is also provided with a plurality of spectrum amplitude transmitters for each frequency band that is divided in advance. Specifically, the spectrum amplitude calculator 62 includes three spectrum amplitude calculators 621, 622, and 623.

  In the equalizer device 6 according to the third embodiment, each of the IIR filters 611, 612, and 613 of the IIR filter unit 61 is a secondary IIR filter having the configuration shown in FIG.

  In this case, in each of the IIR filters 611, 612, 613, based on the instruction input from the user received through the parameter input unit 14, the variables a0, a1, a2, -b1, Change -b2. Thereby, the frequency characteristic according to the user instruction can be expressed in each frequency band.

  Then, each of the spectrum amplitude calculators 621, 622, and 623 calculates an amplitude characteristic in each frequency band based on output signals (frequency characteristic information) from the IIR filters 611, 612, and 613.

That is, in the case of the second-order IIR filter having the configuration shown in FIG. 21, the amplitude characteristic M (ω) can be obtained by (Equation 6) next. That means
M (ω) = √ (P / R) (Formula 6)
It becomes.

here,
P = (a0 + a1 cos ωT + a2 cos 2ωT) ² + (a1 sin ωT + a2 sin 2ωT) ² (Expression 7)
R = (1 + b1 cos ωT + b2 cos 2ωT) ² + (b1 sin ωT + b2 sin 2ωT) ² (Equation 8)
ω = 2πf
T = 1 / fs
fs = sampling frequency
It is.

  In (Expression 7) and (Expression 8), each of characters a0, a1, a2, b1, and b2 is a variable of each multiplier circuit of the secondary IIR filter having the configuration shown in FIG.

  Accordingly, in each of the spectrum amplitude calculators 621, 622, and 623, the variables a0, a1, a2, b1, and b2 are provided in each multiplier circuit from the corresponding IIR filter of the IIR filters 611, 612, and 613. receive.

  In each of the spectrum amplitude calculators 621, 622, and 623, calculations according to the above-described (Expression 6), (Expression 7), and (Expression 8) are performed. Thereby, in each of the spectrum amplitude calculators 621, 622, and 623, the amplitude gains in the corresponding IIR filters 611, 612, and 613 are calculated.

  In this way, the gain calculated in each of the spectrum amplitude calculators 621, 622, and 623 is supplied to the adder 63. The adder 63 can calculate the total gain for each frequency band (frequency point) by adding the gains from the spectrum amplitude calculators 621, 622, and 623 supplied thereto in decibel conversion.

  The total gain from the adder 63 is supplied to the adjustment information calculation unit 41 as frequency characteristic information.

  The adjustment information calculation unit 41 calculates the adjustment information with the phase term (imaginary number term) set to 0 (zero) as described in the equalizer devices 4 and 5 of the second embodiment described above. The adjustment information calculated here is converted into a signal in the time axis domain by the inverse orthogonal transform unit 42, and then the transposition processing described with reference to FIG. 7 is performed in the data transposition unit 43 and supplied to the FIR filter 12. The

  Thereby, also in the equalizer device 5 of the modified example of the third embodiment, it is possible to give an appropriate amplitude characteristic in accordance with an instruction from the user while simplifying the processing in the adjustment information calculation unit 41. Also, the equalizer device 6 having the linear phase characteristic can be realized with a simple configuration.

[Modification of Third Embodiment]
Whereas the equalizer device 6 shown in FIG. 9 performs signal processing in the time axis region, the equalizer device 7 of this modified example performs signal processing in the frequency axis region.

  FIG. 10 is a block diagram for explaining a configuration of an equalizer device 7 according to a modification of the third embodiment. As is apparent from a comparison between FIG. 10 and FIGS. 8 and 9 described above, the equalizer device 7 shown in FIG. 10 is connected to the equalizer device shown in FIG. 9 before the adjustment information calculation unit 41 of the equalizer device 5 shown in FIG. 9 parameter input unit 14, IIR filter unit 61, spectrum amplitude calculation unit 62, and adder 63 are provided.

  For this reason, in FIG. 10, the same reference numerals are given to portions configured in the same manner as in FIG. 8 or FIG. 9, and detailed descriptions thereof are omitted because they are duplicated.

  Therefore, in the equalizer device 7 according to the modification of the third embodiment, the orthogonal transform unit 51 realizes a function as an input signal orthogonal transform unit, and the inverse orthogonal transform unit 53 functions as an inverse orthogonal transform unit. Is realized. The orthogonal transform unit 54 realizes a function as adjustment information orthogonal transform means, and the multiplier 52 realizes a function as frequency characteristic adding means.

  In the case of the equalizer device 7 according to the modification of the third embodiment shown in FIG. 10, the adjustment information transposed in the data transposition unit 43 is converted into adjustment information in the frequency axis region in the orthogonal transformation unit 54. It is configured to supply to the multiplier 52. As a result, frequency characteristics according to user instructions can be added to the input digital audio signal in the frequency axis region.

  Also in this case, an appropriate amplitude characteristic can be given in accordance with an instruction from the user while simplifying the process in the adjustment information calculation unit 41. Further, the equalizer device 7 having the linear phase characteristic can be realized with a simple configuration.

[Other Modifications of Third Embodiment]
In the third embodiment described above, the adjustment information calculation unit 41 calculates the adjustment information by setting the phase term to 0 (zero), as in the case of the equalizer devices 4 and 5 of the second embodiment. I did it. However, it is not limited to this.

  In the equalizer devices 6 and 7 of the third embodiment, instead of the adjustment information calculation unit 41, the adjustment information calculation unit 25 used in the equalizer devices 2 and 3 of the first embodiment is used. You can also

  That is, by using the total gain from the adder 63 as the amplitude value abs (m) and calculating the phase term in accordance with the above-described (Expression 2) and (Expression 3), adjustment information having a linear phase (equalizer coefficient) In the frequency axis region.

  Using the thus obtained characteristic information, it is possible to add a frequency characteristic corresponding to a user instruction to the digital audio signal. In other words, in the equalizer devices 2 and 3 shown in FIGS. 4 and 5, the configuration preceding the adjustment information calculation unit 25 is the same as the equalizer devices 6 and 7 shown in FIGS. 9 and 10. You can also

  Further, in the equalizer device having the configuration shown in FIGS. 9 and 1, the function realized by each unit is realized by a program, and the computer mounted on the audio device is caused to execute the program, thereby the equalizer device of the present invention. Can also be configured.

<Fourth embodiment>
Next, an equalizer apparatus according to a fourth embodiment to which an embodiment of the apparatus, method, and program of the present invention is applied will be described. In the first to third embodiments described above, the amplitude characteristic processing unit of each equalizer apparatus obtains the amplitude characteristic by performing the calculation shown in (Expression 1) or (Expression 6). In the amplitude characteristic calculation of these (Expression 1) and (Expression 6), it is necessary to obtain the square root.

  However, when a signal processing apparatus such as an amplitude equivalent processing unit is particularly of a fixed-point processing and does not have a square root process as a function, the square root calculation needs to be an approximate calculation. In this case, the calculation takes time and causes an increase in processing time.

  Therefore, in the equalizer device according to the fourth embodiment, the set value of the IIR filter is set to “1/2 in gain notation of a desired gain in decibels” and “Q value is higher than desired Q value”. “To be smaller” is set in advance. By doing so, it is possible to perform approximate calculation in which the square root calculation is omitted in the amplitude equivalence processing unit.

  For this reason, the configuration of the equalizer device itself has the same configuration as that of the equalizer devices 1 to 7 of the first to third embodiments described with reference to FIGS. However, among the set values for the IIR filter 14, the gain and the Q value are set under the above-described conditions.

  This will be specifically described. For example, FIG. 11A shows a target amplitude frequency characteristic. Here, a case of a so-called peak filter in which the center frequency is “1 kHz”, the gain is “10 dB”, and the Q value is “2” is taken as an example.

  On the other hand, the filter characteristics in the IIR filter 14 are set as shown in FIG. As can be seen from comparison between FIG. 11A and FIG. 11B, in the case of FIG. 11B, the gain is ½ (2 minutes) in decibel notation as compared to FIG. 1) shows a case where the Q value is set to about 4/5 (4/5).

In this way, by setting the gain to ½ in advance, the square root is removed in the above (Equation 1),
abs (m) = (Xm ² + Ym ² ) (Formula 9)
It is sufficient to perform the following calculation.

Also, in the case of (Equation 6) described above, the square root is removed,
M (ω) = (P / R) (Equation 10)
It is sufficient to perform the following calculation.

  Thus, by changing the settings of the gain and Q value in the IIR filter 14, it is possible to omit the calculation by the square root in the calculations in the amplitude equalization processing units 161, 251 and 411.

  This simplifies the calculations in the amplitude equivalent processing units 161, 251 and 411, prevents the calculation from taking time, and prevents the processing time from increasing.

  Also in the case of the fourth embodiment, the function realized by each unit is realized by a program, and the equalizer device of the present invention is configured by causing the computer mounted on the audio device to execute the program. You can also

<Fifth embodiment>
Next, a specific example of a headphone system to which the equalizer device according to the present invention is applied will be described. The headphone system according to the fifth embodiment is capable of localizing a sound image at an arbitrary position around a listener (user) as will be described in detail below, and is called a so-called virtual surround headphone system. Is.

  The virtual surround headphone system can handle any number of input signals in principle. However, in the following, in order to simplify the description, a case where the present invention is applied to a headphone system of two left and right channels will be described.

  FIG. 12 is a diagram for explaining a basic configuration of the headphone system according to the fifth embodiment. FIG. 13 is a conceptual diagram of the virtual sound image localization process applied to the headphone system of FIG. FIG. 14 is a diagram illustrating an example of an impulse response having a transfer characteristic which will be described later. FIG. 15 is a block diagram for explaining a specific example of the signal processing apparatus 102 shown in FIG.

  As shown in FIG. 12, the headphone system of the fifth embodiment includes a left channel audio input terminal 101L and a right channel audio input terminal 101R.

  Following these audio input terminals 101L and 101R are a signal processing device 102, a D / A converter 103L for the left channel, a D / A converter 103R for the right channel, an amplifier 104L for the left channel, an amplifier 104R for the right channel, and headphones. 105 is provided.

  The digital audio signals input through the audio input terminals 101L and 101R are supplied to the signal processing device 102, where signal processing (sound image localization processing) for localizing the sound image created by each audio signal to an arbitrary position is performed. Applied.

  The left and right digital audio signals that have been subjected to the sound image localization processing in the signal processing device 102 are converted into analog audio signals in the D / A converters 103L and 103R. The left and right audio signals converted into analog audio signals are amplified by the amplifiers 104L and 104R, and then supplied to the corresponding speakers of the headphones 105. As a result, sound corresponding to the sound signals of the left and right channels subjected to the sound image localization processing is emitted from the corresponding speaker of the headphones.

  Then, in the signal processing device 102 shown in FIG. 12, the transfer functions HLL, HLR, and the two speakers SL, SR placed in front of the listener M to both ears of the listener M, as shown in FIG. Transfer characteristics corresponding to HRR and HRL are provided.

  Here, the transfer characteristic HLL is a transfer characteristic from the speaker SL to the left ear YL of the listener M. The transfer characteristic HLR is a transfer characteristic from the speaker SL to the right ear YR of the listener M. The transfer characteristic HRR is a transfer characteristic from the speaker SR to the right ear YR of the listener M. The transfer characteristic HRL is a transfer characteristic from the speaker SR to the left ear YL of the listener M.

  These transfer functions HLL, HLR, HRR, and HRL can be obtained as impulse responses as shown in FIG. 14 on the time axis. By realizing this impulse response in the signal processing device 102 of FIG. 12, even when the reproduced sound is heard through the headphones, the speakers SL and SR placed in front of the listener M as shown in FIG. It is possible to reproduce a sound image equivalent to that produced by.

  As described above, the process of adding the transfer functions HLL, HLR, HRR, and HRL to the audio signal to be processed is realized by the FIR filter provided in the signal processing apparatus 102 of the headphone system shown in FIG. Is done.

  Specifically, the signal processing apparatus 102 shown in FIG. 12 is configured as shown in FIG. That is, an FIR filter 1021 that realizes the transfer function HLL and an FIR filter 1022 that realizes the transfer function HLR are provided for the audio signal input through the audio input terminal 101L of the left channel.

  In addition, an FIR filter 1023 that realizes a transfer function HRL and an FIR filter 1024 that realizes a transfer function HRR are provided for an audio signal input through the audio input terminal 101R of the right channel.

  The output signal from the FIR filter 1021 and the output signal from the FIR filter 1023 are added by the adder 1025 and supplied to the left speaker of the headphone 105. The output signal from the FIR filter 1024 and the output signal from the FIR filter 1022 are added by the adder 1026 and supplied to the right speaker of the headphone 105.

  With the signal processing apparatus 102 having such a configuration, transfer functions HLL and HLR can be added to the audio signal of the left channel, and transfer functions HRL and HRR can be added to the audio signal of the right channel. .

  The left channel audio signal to which the transfer function HLL is added and the right channel audio signal to which the transfer function HRL is added are added by the adder 1025 and supplied to the left speaker. On the other hand, the right channel audio signal to which the transfer function HRR is added and the left channel audio signal to which the transfer function HLR is added are added by the adder 1026 and supplied to the right speaker.

  As a result, the reproduced sound that can be heard as if the sound is being emitted from the speakers SL and SR arranged in front of the listener, as shown in FIG. You can listen. Therefore, you can listen to the sound as if it is being emitted from the front speaker, rather than receiving the feeling that the reproduced sound is in the listener's head even though you are using headphones. Can do.

  Although the speaker system to which the sound image localization processing is applied is realized as described above, there are cases where the transfer functions HLL, HLR, HRR, and HRL captured as data do not necessarily match the listener's auditory characteristics.

  This is because the characteristics of the microphone, commonly called a dummy head microphone, that measures the transfer function, do not match the characteristics of the ears of the listener that listens to the sound through the headphones, and the characteristics vary depending on the type of headphones and variations in wearing, etc. The effect of

  In order to reduce such influence, it is often performed to provide an equalizer for adjusting sound quality separately from such transfer function data so as to match the listener's audibility.

  However, when such an equalizer process is to be configured by a general analog compatible IIR filter filter, the sound image may not be localized at a target position.

  In other words, when the IIR filter is used, as described above, the phase and time relationship for sound image localization possessed by the original transfer characteristic is shifted due to the rotation of the phase and the disorder of the group delay characteristic. Sound image localization is lost.

  Therefore, it is considered to apply the equalizer device according to the present invention to such a headphone system. FIG. 16 is a block diagram for explaining an example in which the equalizer apparatus according to the present invention is applied to the signal processing apparatus 102 shown in FIG.

  As shown in FIG. 16, in the signal processing device 102 described with reference to FIG. 15, an equalizer device 1027 is provided after the adder 1025, and an equalizer device 1028 is provided after the adder 1026. It has a configuration.

  In FIG. 16, any of the equalizer devices of the first to fourth embodiments described with reference to FIGS. 1 to 11 is applied to each of the equalizer devices 1027 and 1028. As described above, the equalizer devices of the first to fourth embodiments all have linear phase characteristics.

  Therefore, even if an arbitrary amplitude frequency characteristic is added by performing the equalizer process using the equalizer device of the first to fourth embodiments described above, the phase characteristic and group that the original transfer characteristic has are added. The delay characteristic is not affected.

  Therefore, as shown in FIG. 16, even if an equalizer device is added, a disturbance of sound image localization due to the addition of the equalizer does not occur, and a headphone system (virtual surround headphone system) having a good sound image localization can be configured. Is possible.

  As can be seen from the above description, as shown in FIG. 16, the portion including the FIR filters 1021, 1922, 1023, and 1024 and the adders 1025 and 1026 is a signal processing unit that performs sound image localization processing. The processing units 1027 and 1028 are equalizer units having the configuration of the present invention.

[Modification of Fifth Embodiment]
The signal processing device 102 shown in FIG. 16 is provided with an equalizer device according to the present invention at the subsequent stage of the adders 1025 and 1026. On the other hand, in the modification of the fifth embodiment, the equalizer device of the present invention is provided before the adders 1025 and 1026.

  FIG. 17 is a block diagram for explaining a signal processing device 102X according to a modification of the fifth embodiment. That is, the signal processing device 102X of this modification is also used as the signal processing device 102 of the headphone system shown in FIG.

  Then, the signal processing device 102X according to this modification performs a convolution operation processing of the linear phase-equalized equalizer coefficient EQ obtained by the equalizer device according to the present invention and the original transfer function, and the result is used as a new filter coefficient. It is configured.

  That is, as shown in FIG. 17, FIR filters 1021S and 1022S are provided in the left channel, and FIR filters 1023S and 1024S are provided in the right channel.

  For the FIR filter 1021S, a new HLL which is a new transfer function obtained by convolution of the linear phase-equalized equalizer coefficient (adjustment information) EQ obtained by the equalizer device of the present invention and the original transfer function HLL. Set.

  For the FIR filter 1022S, a new HLR which is a new transfer function obtained by convolution of the linear phase-equalized equalizer coefficient (adjustment information) EQ obtained by the equalizer device of the present invention and the original transfer function HLR. Set.

  For the FIR filter 1023S, a new HRL which is a new transfer function obtained by convolution of the linear phased equalizer coefficient (adjustment information) EQ obtained by the equalizer device of the present invention and the original transfer function HRL. Set.

  For the FIR filter 1024S, a new HRR which is a new transfer function obtained by convolution of the linear phase-equalized equalizer coefficient (adjustment information) EQ obtained by the equalizer device of the present invention and the original transfer function HRR. Set.

  As a result, before the adders 1025 and 1026, the frequency characteristics corresponding to the user's instruction input can be added to the audio signal, and the target virtual sound image localization can also be added and output.

  FIG. 18 is a block diagram for explaining a specific configuration of a portion using the transfer function HLL located at the top of the four portions surrounded by a dotted line in FIG. The example of FIG. 18 shows a case where the equalizer device 1 according to the first embodiment described with reference to FIG. 1 is applied.

  18, parts that are configured in the same manner as the equalizer device 1 according to the first embodiment shown in FIG. That is, the parameter input unit 14, the IIR filter 15, and the FIR filter coefficient calculation unit 16 are portions to which the equalizer device 1 according to the first embodiment is applied.

  As described in the first embodiment, the FIR filter coefficient calculation unit 16 forms the FIR filter coefficient (equalizer coefficient) EQ that is linearly phased from the frequency characteristic information according to the instruction input from the user. The

  The equalizer coefficient EQ and the transfer function HLL obtained in advance are subjected to a convolution calculation process in the convolution calculator 95, whereby a new HLL which is a new transfer function is formed.

  The new HLL is supplied to the FIR filter 1021S and is convoluted with the audio signal to be processed, whereby a frequency characteristic according to an instruction from the user can be added to the audio signal, and a desired frequency characteristic can be added. Sound image localization processing can also be performed.

  The filter portion having the configuration shown in FIG. 18 is configured using a transfer function corresponding to each of the four portions indicated by dotted lines in FIG.

  Although the equalizer device 1 according to the first embodiment is applied here, the present invention is not limited to this. Any of the other equalizer devices 2, 3, 4, 5, 6, and 7 described above can be applied. It is of course possible to apply the fourth embodiment.

  Also, when using the equalizer devices 2, 3, 5, and 7 described above, the transfer function for sound image localization is also orthogonally transformed into a signal in the frequency axis region, and multiplied by the equalizer coefficient EQ. A new transfer function may be obtained. In this case, the portion that applies the user-designated frequency characteristics and sound image localization to the audio signal by a transfer function may be configured as a multiplier.

  Thus, the equalizer device of the present invention can be mounted on a headphone system that performs sound image localization processing. Accordingly, it is possible to realize a headphone system that can add a frequency characteristic according to a user instruction and appropriately perform a sound image localization process without causing a phase delay or the like.

  In other words, in the case of the headphone system according to the fifth embodiment, it is possible to avoid deterioration of the sound image due to phase delay or group delay regardless of which equalizer curve is set, and free sound quality adjustment. Can do.

  In the modification of the fifth embodiment, the parameter input unit 14 realizes the function of the accepting unit, the IIR filter 15 realizes the function of the characteristic information forming unit, and the FIR filter coefficient calculation unit 16 adjusts the adjustment information. The function of the calculation means is realized.

  The convolution calculator 95L realizes a function as coefficient forming means, and the FIR filter 1021S realizes a function as characteristic adding means.

<Sixth Embodiment>
Next, a specific example of a speaker system to which the equalizer device according to the present invention is applied will be described. As the speaker system, there are various so-called multi-channel speaker systems having multiple channels. However, in order to simplify the description, a case of a left and right two-channel speaker system will be described as an example.

  FIG. 19 is a block diagram for explaining a configuration example of a general speaker system. As shown in FIG. 19, the left channel includes an audio input terminal 201L, a signal processing unit 202L, a D / A (digital / Analog) converter 203L, an amplifier 204L, and a speaker 205L.

  The signal processing unit 202L includes an SP inverse filter 202L1 and an equalizer unit 202L2. The signal processing unit 202L is configured by, for example, a DSP (Digital signal Processor).

  The SP inverse filter 202L1 is for canceling the characteristic that the left channel system of the speaker system gives to the audio signal. The equalizer unit 202L2 adds a frequency characteristic corresponding to an instruction from the user to the audio signal.

  Similarly, the right channel includes an audio input terminal 201R, a signal processing unit 202R, a D / A (digital / Analog) converter 203R, an amplifier 204R, and a speaker 205R.

  The signal processing unit 202R includes an SP inverse filter 202R1 and an equalizer unit 202R2. The signal processing unit 202R is configured by, for example, a DSP (Digital signal Processor).

  The SP inverse filter 202R1 is for canceling the characteristic that the left channel system of the speaker system gives to the audio signal. The equalizer unit 202R2 adds a frequency characteristic corresponding to an instruction from the user to the audio signal.

  The linear phase loudspeaker system as shown in FIG. 19 is characterized by excellent sound image localization because the phase characteristic is a straight line. It is conceivable to provide a conventional IIR type equalizer or analog equalizer as the equalizer unit for adjusting the sound quality according to the installation state or the user's preference. However, in the case of a conventional IIR type equalizer or analog equalizer, it is considered that the phase characteristics are disturbed and the sound image localization may be adversely affected.

  Therefore, in the speaker system according to the sixth embodiment, in FIG. 19, the signal processing unit 202L including the SP inverse filter 201L1 and the equalizer unit 201L2 is configured so that the SP inverse filter and the equalizer unit are integrated. To do.

  By doing so, the SP inverse filter coefficient and the equalizer coefficient can be convolved with the input audio signal as one coefficient, and a linear phase speaker system without sound image localization deterioration is realized.

  The right channel signal processing unit 202R has the same configuration, but the configuration of the left channel signal processing unit 202L will be specifically described here for the sake of simplicity.

  FIG. 20 shows an example of the configuration of the signal processing unit 202L in which the SP inverse filter coefficient and the equalizer coefficient can be convolved with the input audio signal as one coefficient by integrating the SP inverse filter and the equalizer unit. It is a block diagram for demonstrating. The example of FIG. 20 shows a case where the equalizer device 1 of the first embodiment described with reference to FIG. 1 is applied.

  20, parts that are configured in the same way as the equalizer device 1 according to the first embodiment shown in FIG. That is, the parameter input unit 14, the IIR filter 15, and the FIR filter coefficient calculation unit 16 are portions to which the equalizer device 1 according to the first embodiment is applied.

  As described in the first embodiment, the FIR filter coefficient calculation unit 16 forms the FIR filter coefficient (equalizer coefficient) EQ that is linearly phased from the frequency characteristic information according to the instruction input from the user. The

  The equalizer coefficient EQ and the SP inverse filter coefficient RSP obtained in advance are subjected to a convolution calculation process in the convolution calculator 96, whereby a new combined coefficient MX is formed.

  This new coefficient MX is supplied to the FIR filter 97L, which is convolved with the audio signal to be processed. As a result, it is possible to add a frequency characteristic corresponding to an instruction from the user to the audio signal and to cancel the speaker characteristic.

  The signal processing unit having the configuration shown in FIG. 20 is also applied to the right channel signal processing unit 202R.

  In addition, although the equalizer apparatus 1 of 1st Embodiment was applied here, it is not restricted to this. Any of the other equalizer devices 2, 3, 4, 5, 6, and 7 described above can be applied. It is of course possible to apply the fourth embodiment.

  In addition, when the equalizer devices 2, 3, 5, and 7 described above are used, the SP inverse filter coefficient RSP is also orthogonally transformed into a signal in the frequency axis region, and multiplied by the equalizer coefficient EQ. A simple coefficient MX may be obtained. In this case, a portion that cancels the frequency characteristics and speaker characteristics specified by the user with respect to the audio signal may be configured as a multiplier.

  Thus, the equalizer device of the present invention can be mounted on the speaker system. As a result, it is possible to realize a speaker system excellent in linear phase characteristics by adding frequency characteristics according to user instructions and canceling speaker transmission characteristics without causing phase delay or the like.

  In other words, in the case of the speaker system of the sixth embodiment, it is possible to avoid deterioration of the sound image due to phase delay or group delay regardless of what equalizer curve is set, and free sound quality adjustment Can do.

  In the modification of the sixth embodiment, the parameter input unit 14 realizes the function of the accepting unit, the IIR filter 15 realizes the function of the characteristic information forming unit, and the FIR filter coefficient calculation unit calculates the adjustment information. Implement the function of the means.

  The convolution calculator 96L realizes a function as coefficient forming means, and the FIR filter 97L realizes a function as characteristic adding means.

<Others>
The equalizer device according to the above-described embodiment can be configured as a single equalizer device. The equalizer of the present invention can be applied to other acoustic devices such as an amplifier device, various sound reproducing devices, and an acoustic recording / reproducing device. The device can be applied.

It is a block diagram for demonstrating the equalizer apparatus 1 of 1st Embodiment of this invention. It is a figure which shows the characteristic of the FIR filter 12 when 1 kHz of the equalizer apparatus 1 shown in FIG. 1 is made into a center frequency. It is a figure which shows the mode of the impulse response of the FIR filter. It is a block diagram for demonstrating the structure of the equalizer apparatus 2 of the modification 1 of 1st Embodiment. It is a block diagram for demonstrating the structure of the equalizer apparatus 3 of the modification 2 of 1st Embodiment. It is a block diagram for demonstrating the equalizer apparatus 4 of 2nd Embodiment of this invention. It is a figure for demonstrating the process of the data transposition part 43 of the equalizer apparatus 4 shown in FIG. It is a block diagram for demonstrating the structure of the equalizer apparatus 5 of the modification of 2nd Embodiment. It is a block diagram for demonstrating the equalizer apparatus 6 of 3rd Embodiment of this invention. It is a block diagram for demonstrating the structure of the equalizer apparatus 7 of the modification of 3rd Embodiment. It is a figure which shows the target amplitude frequency characteristic and an actual amplitude frequency characteristic. It is a figure for demonstrating the basic composition of the headphone system of 5th Embodiment of this invention. It is a conceptual diagram of the virtual sound image localization process applied to the headphone system of FIG. It is a figure which shows the example of the impulse response of a transfer characteristic. 3 is a diagram for explaining a configuration example of a signal processing device 102. FIG. It is a figure for demonstrating the structural example of the signal processing apparatus 102 to which the equalizer apparatus of this invention was applied. It is a block diagram for demonstrating the signal processing apparatus 102X of the modification of this 5th Embodiment. It is a block diagram for demonstrating the specific structure of the part using the transfer function HLL located in the uppermost part among four parts enclosed with the dotted line in FIG. It is a block diagram for demonstrating the structural example of a general speaker system. It is a block diagram for demonstrating the structural example of 202 L of signal processing parts which convolve an input audio | voice signal by making SP inverse filter coefficient and an equalizer coefficient into one coefficient. It is a figure which shows the structural example of an IIR filter. It is a figure which shows the amplitude frequency characteristic (upper side) and group delay characteristic (lower side) at the time of implement | achieving the peak filter which makes 1 kHz the center frequency with an IIR filter. is doing. It is a figure which shows the impulse response in the case of FIG. It is a figure which shows the mode at the time of inputting a sweep sign signal to the IIR filter which has a characteristic as shown in FIG. 22, FIG. It is a figure which shows the structural example of a FIR filter.

Explanation of symbols

  DESCRIPTION OF SYMBOLS 1 ... Equalizer apparatus, 11 ... Digital audio signal input end 11, 12 ... FIR filter, 13 ... Digital audio signal output end, 14 ... Parameter input unit, 15 ... IIR filter, 16 ... FIR filter coefficient calculation unit 16, 161 DESCRIPTION OF SYMBOLS Amplitude equivalent process part 162 ... Phase equivalent coefficient generator, 2 ... Equalizer apparatus, 21 ... Orthogonal transformation part, 22 ... Multiplier, 23 ... Inverse orthogonal transformation part, 24 ... Orthogonal transformation part, 25 ... Adjustment information calculation part, 251 ... Amplitude equivalent processing unit, 252 ... Phase equivalent coefficient generator, 3 ... Equalizer device, 31 ... Impulse generator, 32 ... Window function processing unit, 4 ... Equalizer device, 41 ... Adjustment information calculation unit, 411 ... Amplitude equivalent processing , 412 ... phase equivalent coefficient generator, 42 ... inverse orthogonal transform part, 43 ... data transposition part, 5 ... equalizer device, 51 ... orthogonal transform part, 52 ... multiplier, 53 ... inverse AC converter, 54 ... orthogonal transform unit, 6 ... equalizer device, 61 ... IIR filter unit, 62 ... spectral amplitude calculator, 1027, 1028 ... equalizer device, 101L ... left channel audio input terminal, 1021S ... FIR filter, 1025 ... Adder, 95L ... convolution calculator, 201L ... left channel audio input terminal, 96L ... convolution calculator, 97L ... FIR filter

Claims (23)

An accepting means for accepting an instruction input from the user regarding the frequency characteristics;
Characteristic information forming means for forming frequency characteristic information in response to the instruction input from the receiving means;
Based on the frequency characteristic information from the characteristic information forming means, adjustment information calculation means for calculating adjustment information having the same amplitude characteristic as the frequency characteristic information and controlling the phase characteristic to a linear phase;
An equalizer device comprising: frequency characteristic adding means for adding a frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming means to the input audio signal using the adjustment information from the adjustment information calculating means. . The equalizer device according to claim 1,
The equalizer device is a finite impulse response filter in which the frequency characteristic adding means uses the adjustment information to add a frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming means by a convolution operation. The equalizer device according to claim 1,
Characteristic orthogonal transforming means for transforming the frequency characteristic information from the characteristic information forming means into frequency characteristic information of a frequency axis region;
An input signal orthogonal transforming means for converting the input voice signal in the time axis domain into an input voice signal in the frequency axis domain;
An inverse orthogonal transforming means for transforming an output voice signal in the frequency axis region from the frequency characteristic adding means into a signal in the time axis region;
The adjustment information calculation unit calculates the adjustment information of the frequency axis region based on the frequency characteristic information of the frequency axis region from the characteristic orthogonal transform unit,
The frequency characteristic adding unit multiplies the input audio signal in the frequency axis region from the input signal orthogonal transform unit by the adjustment information in the frequency axis region from the adjustment information calculation unit, thereby forming the characteristic information An equalizer device as multiplication means for adding a frequency characteristic according to the frequency characteristic information formed by the means. The equalizer device according to claim 3,
Impulse generating means for generating an impulse signal at a constant level per unit time,
The characteristic information forming means adds an amplitude information and phase information corresponding to the frequency characteristic information to the impulse signal from the generating means and outputs the impulse. The equalizer device according to claim 1,
Orthogonal transform means for converting the frequency characteristic information in the time axis region from the characteristic information forming means to frequency characteristic information in the frequency axis region;
Adjustment information inverse orthogonal transform means for converting the adjustment information in the frequency axis region from the adjustment information calculation means to adjustment information in the time axis region;
For the adjustment information in the time axis region from the adjustment information inverse orthogonal transform means, transposition means for replacing the first half and the second half in the time axis direction, and
The adjustment information calculating means calculates the amplitude characteristic in the frequency axis area based on the frequency characteristic information in the frequency axis area from the orthogonal transform means, and sets the phase characteristic to 0 (zero) in the frequency axis area. ) To calculate the adjustment information by performing phase equivalence processing that linearizes each time,
The frequency characteristic adding means adds a frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming means to the input audio signal using the adjustment information subjected to the transposition processing from the transposition means. Equalizer device. The equalizer device according to claim 3 or 4, wherein:
Adjustment information inverse orthogonal transform means for converting the adjustment information in the frequency axis region from the adjustment information calculation means to adjustment information in the time axis region;
For the adjustment information of the time axis region from the adjustment information inverse orthogonal transform means, transposition means for replacing the first half and the second half in the time axis direction,
Adjustment information orthogonal transformation means for converting the adjustment information of the transposed time axis region from the transposition means into adjustment information of the frequency axis region, and
The adjustment information calculating means calculates the amplitude characteristic in the frequency axis area based on the frequency characteristic information in the frequency axis area from the characteristic orthogonal transform means, and sets the phase characteristic to 0 ( The adjustment information is calculated by performing a phase equivalence process that linearizes at zero) degrees,
The frequency characteristic adding unit uses the adjustment information in the frequency axis region from the adjustment information orthogonal transform unit to convert the frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming unit into an input audio signal. Equalizer device to be added. The equalizer device according to claim 1,
The accepting means is capable of accepting an instruction input from a user regarding frequency characteristics for each predetermined frequency band,
A plurality of infinite impulse response filters for forming frequency characteristic information for each predetermined frequency band in response to the instruction input for each predetermined frequency band from the reception unit; For each of the plurality of infinite impulse response filters, a plurality of amplitude calculation means for calculating an amplitude value on the frequency axis based on the frequency characteristic information from each of the plurality of infinite impulse response filters, and the amplitude value from the plurality of amplitude calculation means And adding means for adding
Control data inverse orthogonal transform means for converting the frequency axis domain adjustment information from the adjustment information calculation means into time axis domain adjustment information;
For the adjustment information in the time axis region from the adjustment information inverse orthogonal transform means, transposition means for replacing the first half and the second half in the time axis direction, and
The adjustment information calculation means includes an amplitude equivalent process for calculating the amplitude characteristic in the frequency axis region based on the amplitude value from the addition means, and a phase equivalent process for making the phase characteristic a linear phase in the frequency axis region To calculate the adjustment information,
The frequency characteristic adding means adds a frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming means to the input audio signal using the adjustment information subjected to the transposition processing from the transposition means. Equalizer device. The equalizer device according to claim 7, wherein
Input signal orthogonal transforming means for transforming an input speech signal, which is information on the time axis, into a signal in the frequency axis region;
An inverse orthogonal transform means for transforming an output audio signal on the frequency axis from the frequency characteristic adding means to a signal on the time axis;
Adjustment information orthogonal transformation means for converting the adjustment information of the time axis region subjected to the transposition processing from the transposition means into adjustment information on the frequency axis,
The frequency characteristic adding means uses the adjustment information from the adjustment information orthogonal transform means for the input signal in the frequency axis region from the input signal orthogonal transform means, and is formed by the characteristic information forming means. An equalizer device that adds frequency characteristics according to frequency characteristic information. An equalizer device according to claim 1, claim 2, claim 3, claim 4, claim 5, claim 6, claim 7 or claim 8,
The characteristic information forming means is an equalizer device configured by an infinite impulse response filter. The equalizer device according to claim 4,
The characteristic information forming unit is an equalizer device that sets the decibel value to one half of the decibel value to be originally realized and sets the Q value to be smaller than the Q value to be originally realized as the amplitude frequency characteristic. An accepting step for accepting an instruction input from the user regarding the frequency characteristics through accepting means;
A characteristic information forming step in which the characteristic information forming means forms frequency characteristic information in response to the instruction input received in the receiving step;
Based on the frequency characteristic information formed in the characteristic information forming step, adjustment information calculation means that adjustment information calculation means calculates adjustment information having the same amplitude characteristic as the frequency characteristic information and controlling the phase characteristic to a linear phase. Process,
A frequency characteristic adding step in which a frequency characteristic adding unit adds a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information forming step to the input audio signal using the adjustment information calculated in the adjustment information calculating step. A frequency characteristic adding method having and. The frequency characteristic adding method according to claim 11,
A characteristic orthogonal transformation step in which the characteristic orthogonal transformation means transforms the frequency characteristic information formed in the characteristic information formation step into frequency characteristic information in a frequency axis region;
An input signal orthogonal transform step in which the input signal orthogonal transform means transforms the input speech signal in the time axis region into an input speech signal in the frequency axis region;
An inverse orthogonal transform step in which the inverse orthogonal transform means transforms the output speech signal in the frequency axis region to which the frequency property is added in the frequency property adding step into a signal in the time axis region, and
In the adjustment information calculation step, the adjustment information of the frequency axis region is calculated based on the frequency characteristic information of the frequency axis region converted in the characteristic orthogonal transformation step,
In the frequency characteristic adding step, the input audio signal in the frequency axis region converted in the input signal orthogonal transformation step is multiplied by the adjustment information in the frequency axis region calculated in the adjustment information calculating step, thereby A frequency characteristic adding method for performing multiplication processing for adding a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information forming step. The frequency characteristic adding method according to claim 12,
The impulse generating means has an impulse generating step of generating an impulse signal at a constant level per unit time;
In the characteristic information forming step, a frequency characteristic adding method of adding and outputting amplitude information and phase information corresponding to the frequency characteristic information to the impulse signal generated in the impulse generating step. The frequency characteristic adding method according to claim 11,
An orthogonal transformation step in which the orthogonal transformation means transforms the frequency characteristic information in the time axis region formed in the characteristic information formation step into frequency characteristic information in the frequency axis region;
An adjustment information inverse orthogonal transform step in which the adjustment information inverse orthogonal transform means converts the adjustment information in the frequency axis region calculated in the adjustment information calculation step into adjustment information in the time axis region;
For the adjustment information in the time axis region converted in the adjustment information inverse orthogonal transform process, the transposition means replaces the first half and the second half in the time axis direction so as to be used in the frequency characteristic adding step. And
In the adjustment information calculation step, based on the frequency characteristic information of the frequency axis region converted in the orthogonal transformation step, an amplitude equivalent process for calculating the amplitude characteristic in the frequency domain, and a phase characteristic of 0 ( The adjustment information is calculated by performing a phase equivalence process that linearizes at zero) degrees,
In the frequency characteristic adding step, a frequency to add to the input audio signal a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information forming step using the adjustment information transposed in the transposing step. Characteristic addition method. The frequency characteristic adding method according to claim 12 or 13, wherein:
An adjustment information inverse orthogonal transform step in which the adjustment information inverse orthogonal transform means converts the adjustment information of the frequency axis region calculated in the adjustment information calculation step into the adjustment information of the time axis region and outputs the adjustment information;
For the adjustment information of the time axis region converted in the adjustment information inverse orthogonal transform process, the transposition step in which the translocation means replaces the first half and the second half in the time axis direction,
An adjustment information orthogonal transform step in which the adjustment information orthogonal transform means converts the adjustment information in the time axis region transposed in the transposition step into adjustment information in the frequency axis region and makes it usable in the frequency characteristic adding step. And
In the adjustment information calculation step, amplitude equalization processing for calculating the amplitude characteristic in the frequency axis region based on the frequency characteristic information in the frequency region converted in the characteristic orthogonal transformation step, and phase characteristics in the frequency axis region are set to 0. The adjustment information is calculated by performing phase equivalence processing to linearize at (zero) degrees,
In the frequency characteristic adding step, using the adjustment information of the frequency axis region converted in the adjustment information orthogonal transformation step, a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information forming step is converted into an input audio signal. Frequency characteristic addition method to be added to. A reception step for receiving an instruction input from the user regarding the frequency characteristics;
A characteristic information forming step for forming frequency characteristic information in response to the instruction input received in the receiving step;
Based on the frequency characteristic information formed in the characteristic information forming step, an adjustment information calculation step for calculating adjustment information having the same amplitude characteristic as the frequency characteristic information and controlling the phase characteristic to a linear phase;
A frequency characteristic adding step for adding to the input audio signal a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information forming step using the adjustment information calculated in the adjustment information calculating step. A computer-readable frequency characteristic addition program executed by a computer mounted on an audio signal processing apparatus for processing the sound. A frequency characteristic adding program according to claim 16,
A characteristic orthogonal transformation step of transforming the frequency characteristic information formed in the characteristic information forming step into frequency characteristic information of a frequency axis region;
An input signal orthogonal transform step for transforming an input voice signal, which is information in the time axis region, into a signal in the frequency axis region;
The computer executes an inverse orthogonal transform step of converting the output voice signal in the frequency axis domain to which the frequency characteristic has been added in the frequency characteristic adding step into a signal in the time axis domain and outputting the signal.
In the adjustment information calculation step, based on the frequency characteristic information of the frequency axis region converted in the characteristic orthogonal transformation step, the adjustment information is calculated in the frequency axis region,
In the frequency characteristic adding step, by multiplying the input audio signal in the frequency axis region converted in the input signal orthogonal transformation step by the adjustment information in the frequency axis region calculated in the adjustment information calculating step, A frequency characteristic addition program for performing multiplication processing for adding a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information formation step. A frequency characteristic adding program according to claim 17,
An impulse generation step for generating a constant level of impulse signal per unit time;
A frequency characteristic addition program for adding and outputting amplitude information and phase information corresponding to the frequency characteristic information to the impulse signal generated in the impulse generation step in the characteristic information forming step. A frequency characteristic adding program according to claim 16,
An orthogonal transformation step for transforming the frequency characteristic information in the time axis region formed in the characteristic information forming step into frequency characteristic information in the frequency axis region;
An adjustment information inverse orthogonal transform step for converting the adjustment information of the frequency axis region calculated in the adjustment information calculation step into adjustment information of the time axis region;
The transposition step of replacing the first half part and the second half part in the time axis direction with respect to the adjustment information in the time axis region transformed in the adjustment information inverse orthogonal transform step so that it can be used in the frequency characteristic adding step. Runs and
In the adjustment information calculation step, amplitude equalization processing for calculating the amplitude characteristic in the frequency axis region based on the frequency characteristic information in the frequency axis region converted in the orthogonal transform step, and phase characteristics in the frequency axis region are set to 0. The adjustment information is calculated by performing phase equivalence processing to linearize at (zero) degrees,
In the frequency characteristic adding step, a frequency for adding to the input audio signal a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information forming step using the adjustment information transposed in the transposing step. Characteristic addition program. A frequency characteristic adding program according to claim 16 or claim 17,
An adjustment information inverse orthogonal transform step for converting the adjustment information of the frequency axis region calculated in the adjustment information calculation step into adjustment information of the time axis region and outputting the adjustment information;
For the adjustment information of the time axis region converted in the adjustment information inverse orthogonal transform step, a transposition step of replacing the first half and the second half in the time axis direction;
The computer executes the adjustment information orthogonal transformation step that converts the adjustment information of the time axis region transposed in the transposition step into adjustment information of the frequency axis region and makes it usable in the frequency characteristic addition step.
In the adjustment information calculation step, based on the frequency characteristic information in the frequency domain converted in the characteristic orthogonal transform step, amplitude equalization processing for calculating the amplitude characteristic in the frequency axis domain and phase characteristics in the frequency axis domain are set to 0. The adjustment information is calculated by performing phase equivalence processing to linearize at (zero) degrees,
In the frequency characteristic adding step, using the adjustment information of the frequency axis region converted in the adjustment information orthogonal transform step, a frequency characteristic corresponding to the frequency characteristic information formed in the characteristic information forming step is converted into an input audio signal. Frequency characteristic addition program to be added to. An accepting means for accepting an instruction input from the user regarding the frequency characteristics;
Characteristic information forming means for forming frequency characteristic information in response to the instruction input from the receiving means;
Based on the frequency characteristic information from the characteristic information forming means, adjustment information calculation means for calculating adjustment information having the same amplitude characteristic as the frequency characteristic information and controlling the phase characteristic to a linear phase;
An equalizer unit comprising: frequency characteristic adding means for adding a frequency characteristic corresponding to the frequency characteristic information formed by the characteristic information forming means to the input audio signal using the adjustment information from the adjustment information calculating means. When,
A sound reproduction apparatus comprising: a signal processing unit that performs sound image localization processing on an input audio signal. An accepting means for accepting an instruction input from the user regarding the frequency characteristics;
Characteristic information forming means for forming frequency characteristic information in response to the instruction input from the receiving means;
Based on the frequency characteristic information from the characteristic information forming means, adjustment information calculation means for calculating adjustment information having the same amplitude characteristic as the frequency characteristic information and controlling the phase characteristic to a linear phase;
In response to provision of the adjustment information from the adjustment information calculation means and a transfer function for sound image localization processing, a frequency characteristic corresponding to a user's instruction input is given and a target sound image localization is realized. Coefficient forming means for forming integrated coefficient information of
A sound reproduction apparatus comprising: characteristic addition means for performing frequency characteristic addition and sound image localization processing on an input audio signal using the integrated coefficient information. An accepting means for accepting an instruction input from the user regarding the frequency characteristics;
Characteristic information forming means for forming frequency characteristic information in response to the instruction input from the receiving means;
Based on the frequency characteristic information from the characteristic information forming means, adjustment information calculation means for calculating adjustment information having the same amplitude characteristic as the frequency characteristic information and controlling the phase characteristic to a linear phase;
In response to the provision of the adjustment information from the adjustment information calculation means and the inverse characteristic coefficient for realizing the inverse characteristic of the speaker, the frequency characteristic according to the user's instruction input is given, and the inverse characteristic of the speaker is Coefficient forming means for forming integrated coefficient information for realizing;
A sound reproduction apparatus comprising: characteristic addition means for performing frequency characteristic addition and speaker characteristic cancellation processing on an input audio signal using the integrated coefficient information.

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