JP5734328B2 - Sound field recording / reproducing apparatus, method, and program - Google Patents

Description

  The present invention relates to a wave field synthesis technique (Ambisonics) that collects a sound signal with a microphone installed in a certain sound field and reproduces the sound field with a speaker using the sound signal. Related to technology.

  The wavefront synthesis method and ambisonics are technologies that virtually reproduce a sound field in a remote place using a plurality of microphones and speakers. As such a technique, for example, a technique described in Non-Patent Document 1 is known. In applications such as remote communication systems, real-time sound collection and reproduction is required, so the sound pressure collected by a general microphone array is unique to a sound field reproduction signal for output by a general speaker array. It must be convertible to

Oyama, Furuya, Hiwazaki, Haneda, Suzuki, “Wavefront reconstruction filter for cylindrical microphone / speaker array”, September 2012, Proceedings of Autumn Meeting of Acoustical Society of Japan, pp.605-608

  In the technique described in Non-Patent Document 1, a filter for conversion is derived on the assumption that a microphone array and a speaker array arranged in a cylindrical shape are used. Therefore, if this filter is applied to a microphone / speaker array arranged in a spherical shape, the sound field cannot be reproduced with high accuracy, and a sound field in which sound comes from all directions around the listener can be accurately reproduced. There was a possibility that it could not be reproduced.

  The object of the present invention is to reproduce a sound field with higher accuracy than before when using a microphone array / speaker array arranged in a spherical shape, and sound that sounds from all directions around the listener. To provide a sound field sound collecting / reproducing apparatus, method, and program capable of reproducing a field with high accuracy.

In order to solve the above problems, sound field sound collecting and reproducing apparatus according to an aspect of the invention, the spherical radius R _m by at least three microphones are fixed to the baffle of the sound absorbing material of the spherical radius R _b Suppose that at least three loudspeakers are arranged on the upper surface, and that at least three speakers are arranged on the inner surface of the spherical surface with radius R _s , R _m > R _b , R _s ≧ R _b, and j is Imaginary unit, ω as frequency, c as sound velocity, k = ω / c, n, m as the order of spherical harmonic spectrum, j _n (・) as n order spherical Bessel function, h _n ^{(1 )} Generated based on the signal picked up by the microphone, where (・) is an nth-order first-class spherical Hankel function, A (ω) is a predetermined complex number, and w _nm is a weight determined based on n and m. is spherical harmonic spectrum signal P ~ _nm (omega) to the filter F ~ _nm (omega) by applying a filtering process after signals D ~ _nm defined by the following equation ( ) And conversion filter unit for generating,

The discrete spherical harmonic inverse transform unit converts the filtered signal D ~ _nm (ω) into a frequency domain signal by discrete spherical harmonic inverse transform, and the frequency domain signal is transformed into a time domain signal by inverse Fourier transform. A frequency inverse transform unit that outputs a time domain signal to a speaker.

Sound field sound collecting and reproducing apparatus according to another aspect of the invention, at least three microphones outward on a sphere of radius R _m by being fixed to the baffle spherical sound absorbing material of radius R _b disposed Suppose that at least three loudspeakers are arranged inward on a spherical surface of radius R _s , R _m > R _b , R _s ≧ R _b , j is an imaginary unit, and ω is the frequency Where c is the speed of sound, k = ω / c, n and m are the orders of the spherical harmonic spectrum, j _n (·) is the nth order spherical Bessel function, and h _n ⁽¹⁾ (·) is the nth order The frequency at which the signal collected by the microphone is converted into a frequency domain signal by Fourier transform, where A (ω) is a predetermined complex number and w _nm is a weight determined based on n and m. A discrete spherical tone that transforms the frequency domain signal into a spherical harmonic spectrum signal P ~ _nm (ω) by a transformer and discrete spherical harmonic transformation. Conversion filter that generates filtered signal D ~ _nm (ω) by applying sum conversion unit and filter F ~ _nm (ω) defined by the following equation to spherical harmonic spectrum signal P ~ _nm (ω) A section.

  Using a microphone array / speaker array filter arranged in a spherical shape, the sound collection signal of the microphone array arranged in a spherical shape can be converted into a drive signal for the speaker array arranged in a spherical shape to reproduce the sound field. Therefore, it is possible to accurately reproduce a sound field in which sound comes from all directions around the listener.

The functional block diagram which shows the example of a sound field sound collection reproducing | regenerating apparatus. The figure for demonstrating the example of arrangement | positioning of a microphone and a speaker. The figure for demonstrating the example of arrangement | positioning of a microphone and a speaker. The flowchart which shows the example of the sound field sound collection reproduction | regeneration method.

  Embodiments of the present invention will be described below with reference to the drawings. In the following explanation, the symbols “~”, “^”, etc. used in the text should be described immediately above the character immediately before, but are described immediately after the character due to restrictions on the text notation. To do. In the formula, these symbols are written in their original positions. Further, the processing performed for each element of a vector or matrix is applied to all elements of the vector or matrix unless otherwise specified.

<Arrangement of microphone array and speaker array>
As shown in FIG. 1, a sound field sound collecting / reproducing apparatus and method includes a microphone composed of N _{ch, θ} × N _{ch, φ} microphones arranged on a spherical surface having a radius R _m of a first space. Using a speaker array composed of N _{ch, θ} × N _{ch, φ} speakers arranged on a spherical surface with a radius R _s of a second space different from the first space, The sound field of the first space formed by the sound generated by the sound source So in the space is reproduced in the second space. In FIG. 1, the sound source So reproduced in the second space is expressed as a sound source So ′. The first space and the second space are different from each other. N _{ch, θ} , N _{ch, φ} are integers of 2 or more.

  Each of the microphones and speakers may be at least three. The number of microphones arranged in the first space and the number of speakers arranged in the second space may be different. If the number of microphones is larger than the number of speakers arranged in the second space, the reproduction signal may be thinned out. On the other hand, when the number of microphones is smaller than the number of speakers arranged in the second space, the reproduction signal may be interpolated by taking an average between channels. As a method for performing the interpolation, for example, linear interpolation, sinc interpolation, or the like can be applied.

For example, as shown in FIG. 3, a microphone is disposed outwardly on the spherical surface SM of radius R _m by being fixed to the baffle SB spherical sound absorbing material of radius R _b. In this case, the microphone will be located away R _m from the center of the baffle SB spherical radius R _b. In other words, with R _m > R _b , the microphone is placed at a position (R _m −R _b ) away from the surface of the circumferential surface of the spherical baffle SB. For example, the microphone is arranged by being supported by a thin rod-like member protruding vertically from the peripheral surface of the baffle SB.

  The sound absorbing material is a material having a performance of absorbing sound waves, and ideally a material having a sound absorption coefficient of ∞. Mainly used for buildings, etc. for the purpose of adjusting the sound of the room and absorbing noise. Specific examples include porous sound absorbing materials such as glass wool and felt, and vibration plate sound absorbing materials that absorb vibration using a thin plate. In the present invention, the type of the sound absorbing material is not limited, and any type of sound absorbing material may be used.

The microphones may be arranged at any interval. That is, θ c as the zenith angle and φ as the azimuth angle, θ _c , which is the interval between the microphones adjacent in the zenith angle direction, and φ _c , which is the interval between the microphones adjacent in the azimuth angle direction, take arbitrary values. Can do. However, by arranging the microphones at equal intervals, that is, by setting each of θ _c and φ _c to the same value, the sound field can be reproduced with high accuracy. In this way, so-called equiangular sampling is possible by making θ _c and φ _c equal angles.

Further, in order to perform the so-called Gaussian sampling, a microphone, the azimuthal direction φ is equal angle, zenith angle direction θ may be an arrangement in that the ^{_{P n m (cosθ) = 0}} . Furthermore, in order to perform so-called uniform sampling, the microphones may be arranged so as to be as uniform as possible on the spherical surface, such as the apex of a regular polygon.

Here, P ^m _n (·) is a Legendre power function and is defined as follows. P _n (·) represents a Legendre polynomial.

  The speaker is also arranged in the same manner as the microphone, but the speaker may be arranged in the second space in an acoustically transparent state, or arranged in the second space in a state that is not acoustically transparent. Also good. The acoustically transparent state is a state in which the same transmission characteristic as that of the second space where no speaker is arranged is maintained.

For example, the speaker is arranged in an acoustically transparent state in the air of the second space by being hung with a thread or fixed with a thin rod. The speaker is, like the microphone, for example, as shown in FIG. 3, by being arranged on the spherical surface of radius R _s by being fixed to the baffle SB spherical sound absorbing material or a rigid radius R _b The second space is not acoustically transparent.

Further, the microphone for being located outwardly on a sphere of radius R _m, the speaker is disposed inwardly on a sphere of radius R _s.

As with the microphone, the speakers do not need to be arranged at exactly equal intervals as long as they are arranged at approximately equal intervals. That is, θ _c , which is the distance between speakers adjacent in the zenith angle direction, and φ _c adjacent in the azimuth angle direction do not have to be exactly the same value, and may be approximately the same value. When equiangular sampling is performed, the sound field can be reproduced with high accuracy by arranging the speakers at substantially equal intervals in the same manner as the microphone, that is, by setting each of θ _c and φ _c to the same value.

In addition, in order to perform so-called Gaussian sampling, the microphone is set with P _n ^m (·) as the Legendre function, the azimuth direction φ is equiangular, and the zenith angle direction θ is P _n ^m (cos θ) = 0. when placed in a point, also the speaker, the azimuthal direction φ is equal angle, the zenith angle direction θ placing the point where the ^{_{P n m (cosθ) = 0}} .

  In addition, in order to perform so-called uniform sampling, if the microphones are arranged so as to be as uniform as possible on the spherical surface, such as a regular polygon vertex, the speaker is similarly arranged on the spherical surface, such as a regular polygon vertex. Arrange them to be as dense as possible.

Although it is assumed that R _s ≧ R _m , this need not be satisfied. R _m and R _s can reproduce a wider area as the values thereof are larger, but more microphones and speakers are required. R _m and R _s are desirably set experimentally in consideration of the frequency of the signal to be collected. For example, R _s = 1.5 m and R _m = 0.2 _m .

Further, the radius R _s of the cylinder in which the speaker is arranged is, for example, about 1.5 m. R _m ≦ R _s or R _s ≦ R _m may be satisfied, but the accuracy of sound field reproduction is improved when R _m ≦ R _s .

The position of the microphone Mi-j in the first space is (R _m , θ _{m, i} , φ _{m, j} ) [i = 1, 2,…, N _{ch, θ} , j = 1, 2,…, N _{ch , φ} ]. The position of the speaker Si-j in the second space is expressed as (R _s , θ _{s, i} , φ _{s, j} ) [i = 1, 2,..., N _{ch, θ} , j = 1, 2 _{,. , φ} ].

<Sound field recording and playback device>
As shown in FIG. 1, the sound field sound collecting and reproducing apparatus includes, for example, a frequency conversion unit 1, a discrete spherical harmonic transformation unit 2, a transformation filter unit 3, a discrete spherical harmonic inverse transformation unit 4, and a frequency inverse transformation unit 5. The process of each step illustrated in (1) is performed.

The microphones M1-1, M2-1,..., MN _{ch, θ-} N _{ch, φ} arranged in the first space pick up the sound emitted from the sound source S in the first space, Generate a signal. The generated signal is sent to the frequency converter 1. A signal at time t in the time domain collected by the microphone Mi-j of (R _m , θ _{m, i} , φ _{m, j} ) is denoted as p _ij (t).

<Frequency converter 1>
The frequency converter 1 converts the signal p _ij (t) collected by the microphones M1-1, M2-1,..., MN _{ch, θ} −N _{ch, φ} into a frequency domain signal P _ij (ω) by Fourier transform. Conversion is performed (step S1). The generated frequency domain signal P _ij (ω) is sent to the discrete spherical harmonic transform unit 2. ω is a frequency. For example, the frequency domain signal P _ij (ω) is generated by short-time discrete Fourier transform. Of course, the frequency domain signal P _ij (ω) may be generated by other existing methods. Further, the frequency domain signal P _ij (ω) may be generated using a method such as overlap add. When the input signal is long or when the signal is continuously input as in real time processing, the processing is performed for each frame such as every 10 ms. For example, the frequency domain signal P _ij (ω) is defined as follows. J in the argument of the function exp is an imaginary unit.

<Discrete spherical harmonic transformation unit 2>
Discrete spherical harmonic conversion unit 2, the discrete spherical harmonic transform converts the frequency domain signal P _ij of (omega) in the spherical harmonic spectrum signal P ~ _nm (ω) (Step S2). The spherical harmonic spectrum signal P _nm (ω) is calculated for each ω. The converted spherical harmonic spectrum signal P _nm (ω) is sent to the conversion filter unit 3.

When equiangular sampling is performed, the discrete spherical harmonic transformation unit 2 calculates P _nm (ω) specifically defined by Equation (1).

β _i is a value determined according to the sampling method. Y _n ^m is a spherical harmonic function defined by the following equation, and n and m are the orders of the spherical harmonic spectrum. 0 ≦ n ≦ N, −N ≦ m ≦ N, and n and m are integers. When a is a complex number, a ^* means a complex conjugate of a.

Expression (1) is an example of discrete spherical harmonic transformation, and the transformation into the spherical harmonic spectral region may be performed by other methods. Further, the spherical harmonic spectrum signal P _nm (ω) may be obtained by numerical calculation.

<Conversion filter unit 3>
The conversion filter unit 3 generates the filtered signal D ~ _nm (ω) by applying the filter F ~ _nm (ω) defined by the equation (2) to the spherical harmonic spectrum signal P ~ _nm (ω). (Step S3). The filtered signal _D˜nm (ω) is transmitted to the discrete spherical harmonic inverse transform unit 4.

In Equation (2), c is the speed of sound, and k = ω / c is the wave number. A (ω) is a predetermined complex number for adjusting the frequency characteristic. For example, A (ω) = 1 + 0 × j = 1. W _nm is a predetermined weight for attenuating the evanescent wave, which is determined as follows based on n and m, for example. In the following equations, n _c and m _c are predetermined values and are cutoff values of n and m, respectively. n _c and m _c are set to values that suppress evanescent waves, for example. α and β are predetermined values for determining the smoothness of the cutoff, and are 0.05, for example. Of course, other weight functions may be used as w _nm .

h ⁽¹⁾ _n (•) is an n-th order first-class spherical Hankel function. j _n (•) is an n-th order spherical Bessel function. h ⁽¹⁾ _n (•) and j _n (•) are defined as follows.

H _n ⁽¹⁾ (•) is an n-th order first-class Hankel function, and J _n (•) is an n-order Bessel function. H _n ⁽¹⁾ (•) and J _n (•) are defined as follows. Γ (z) is a gamma function, and Y _n (z) is a Neumann function.

<Discrete spherical harmonic inverse transform unit 4>
Discrete spherical harmonic inverse transform unit 4, the discrete spherical harmonic inverse transform filtering after signal D ~ _nm (ω) into a frequency domain signal D _ij (ω) (Step S4). The converted frequency domain signal D _ij (ω) is sent to the frequency inverse transform unit 5. When equiangular sampling is performed, the discrete spherical harmonic inverse transform unit 4 calculates the frequency domain signal D _ij (ω) specifically defined by Equation (3).

Expression (3) is an example of the inverse discrete spherical harmonic, and the conversion to the frequency domain may be performed by other methods. Further, the frequency domain signal D _ij (ω) may be obtained by numerical calculation.

<Inverse frequency converter 5>
The frequency inverse transform unit 5 transforms the frequency domain signal D _ij (ω) into a time domain signal P ^d _ij (t) by inverse Fourier transform (step S5). The time domain signal P ^d _ij (t) obtained for each frame by the inverse Fourier transform is appropriately shifted to obtain a linear sum to be a continuous time domain signal. For the inverse Fourier transform, an existing method such as a short-time discrete inverse Fourier transform may be used. The time domain signal P ^d _ij (t) is sent to the speakers Si-j, S2-1,..., SN _{ch, θ-} N _{ch, φ} .

The speaker arrays S1-1, S2-1,..., SN _{ch, θ} −N _{ch, φ} reproduce sound based on the time domain signal P ^d _ij (t). Specifically, as i = 1,..., N _{ch, θ} , j = 1,..., N _{ch, φ} , the speaker Si-j reproduces sound based on the time domain signal P ^d _ij (t). Thereby, the sound field of the first space can be reproduced in the second space.

  In this way, using the microphone array / speaker array filter arranged in a spherical shape, the collected sound signal of the microphone array arranged in a spherical shape is converted into a drive signal for the speaker array arranged in a spherical shape, and the sound field is converted. Since it can be reproduced, it is possible to accurately reproduce a sound field in which sound comes from all directions around the listener.

<Theoretical background>
Hereinafter, the reason why the filter _F˜nm (ω) is expressed as in Expression (2) will be described.

The secondary sound source signal is D (r _s ', ω), and the synthesized sound field by the secondary sound source is P _syn (r', ω). If the transfer function from the secondary sound source at position r _s 'to position r' is G (r'-r _s ', ω), P _syn (r', ω) is the convolution of the spherical harmonic spectral region It is expressed as

  Here, n and m are the orders of the spherical harmonic spectrum.

The sound collection field, spherical sound absorbing baffles radius R _b is, the microphone array and the center of the spherical radius R _m is provided to the matched position. Here, R _b ≦ R _m . If the incident sound field is P _inc (r ′, ω) and the scattered sound field is P _sct (r ′, ω), the sound collection field P _rcv (r ′, ω) can be written as follows.

  Since the sound pressure gradient is 0 on the surface of the sound absorbing material, it can be written as follows.

Here, P _inc (•) and P _sct (•) are expressed in the spherical harmonic spectrum region as follows.

  Substituting Equation (8) and Equation (7) into Equation (6) yields the following.

  Therefore, the relationship between the scattered sound field and the incident sound field can be derived as follows.

  From the equations (5), (7), (8), and (9), the following is obtained.

Therefore, the incident sound field P ^ _{inc, n} ^m (ω) can be expressed as follows using the sound collection field P ^ _{rcv, n} ^m (ω).

Since the desired sound field P ~ _des (·) corresponds to the incident sound field P _inc (·), from expression (10), P ~ _des (·) is the spherical harmonic spectrum P ~ _rcv on the receiving sphere. _{, n} ^m (R _m , ω) can be expressed as follows.

  From Equation (4) and Equation (11), the conversion equation can be derived as follows.

G (r′-r _s ′, ω) is assumed to have monopole characteristics as follows.

Since the filter characteristics are not essentially changed, the equation (12) may be multiplied by A (ω), w _nm . Then, equation (12) matches equation (2).

[Modifications, etc.]
Each unit constituting the sound field sound collecting / reproducing device may be provided in either the sound collecting device arranged in the first space or the reproducing device arranged in the second space. In other words, each processing of the frequency conversion unit 1, the discrete spherical harmonic transformation unit 2, the transformation filter unit 3, the discrete spherical harmonic inverse transformation unit 4, the frequency inverse transformation unit 5, and the window function unit 6 is performed in the first space. It may be executed by the sound collecting device arranged, or may be executed by the reproducing device arranged in the second space. The signal generated by the sound collection device is transmitted to the reproduction device.

  The positions of the first space and the second space are not limited to those shown in FIG. The first space and the second space may be adjacent to each other or separated from each other. Also, the orientation of the first space and the second space may be any.

  As long as the sound field sound collecting / reproducing apparatus includes the conversion filter unit 3, the sound field collecting / reproducing device may not include other units. For example, the sound field sound collecting / reproducing apparatus may include a transform filter unit 3, a discrete spherical harmonic inverse transform unit 4, and a frequency inverse transform unit 5. Further, the sound field sound collecting / reproducing apparatus may include a frequency conversion unit 1, a discrete spherical harmonic conversion unit 2, and a conversion filter unit 3.

  The processing of the frequency conversion unit 1 and the processing of the discrete spherical harmonic conversion unit 2 may be performed simultaneously. Similarly, the process of the discrete spherical harmonic inverse transform unit 4 and the process of the frequency inverse transform unit 5 may be performed simultaneously. Further, the discrete spherical harmonic transformation unit 2 and the discrete spherical harmonic inverse transformation unit 4 may be interchanged.

  The sound field sound collecting / reproducing apparatus can be realized by a computer. In this case, the processing content of each part of this apparatus is described by a program. Then, by executing this program on a computer, each unit in this apparatus is realized on the computer.

  The program describing the processing contents can be recorded on a computer-readable recording medium. In this embodiment, these apparatuses are configured by executing a predetermined program on a computer. However, at least a part of these processing contents may be realized by hardware.

  The present invention is not limited to the above-described embodiment, and can be modified as appropriate without departing from the spirit of the present invention.

DESCRIPTION OF SYMBOLS 1 Frequency conversion part 2 Discrete spherical harmonic transformation part 3 Conversion filter part 4 Discrete spherical harmonic inverse transformation part 5 Frequency inverse transformation part

Claims (5)

And is disposed outwardly on a sphere of radius R _m by at least three microphones are fixed to the baffle of the sound absorbing material of the spherical radius R _b, the spherical surface of the at least three loudspeakers radius R _s R _m > R _b , R _s ≧ R _b , j is an imaginary unit, ω is frequency, c is sound velocity, k = ω / c, n, m is the order of the spherical harmonic spectrum, j _n (•) is the nth-order spherical Bessel function, h _n ⁽¹⁾ (•) is the nth-order first-class spherical Hankel function, and A (ω) is a predetermined value. As a complex number and w _nm as a weight determined based on n and m,
Applying the filter F ~ _nm (ω) defined by the following equation to the spherical harmonic spectrum signal P ~ _nm (ω) generated based on the signal collected by the microphone, the filtered signal D ~ a conversion filter unit for generating _nm (ω);

A discrete spherical harmonic inverse transform unit that transforms the filtered signal D to _nm (ω) into a frequency domain signal by discrete spherical harmonic inverse transform;
A frequency inverse transform unit that transforms the frequency domain signal into a time domain signal by inverse Fourier transform and outputs the transformed time domain signal to the speaker;
Sound field collection and playback device including And is disposed outwardly on a sphere of radius R _m by at least three microphones are fixed to the baffle of the sound absorbing material of the spherical radius R _b, the spherical surface of the at least three loudspeakers radius R _s R _m > R _b , R _s ≧ R _b , j is an imaginary unit, ω is frequency, c is sound velocity, k = ω / c, n, m is the order of the spherical harmonic spectrum, j _n (•) is the nth-order spherical Bessel function, h _n ⁽¹⁾ (•) is the nth-order first-class spherical Hankel function, and A (ω) is a predetermined value. As a complex number and w _nm as a weight determined based on n and m,
A frequency converter that converts the signal collected by the microphone into a frequency domain signal by Fourier transform;
A discrete spherical harmonic transformation unit that transforms the frequency domain signal into a spherical harmonic spectrum signal P to _nm (ω) by discrete spherical harmonic transformation;
A conversion filter unit that generates a filtered signal D to _nm (ω) by applying a filter F to _nm (ω) defined by the following equation to the spherical harmonic spectrum signal P to _nm (ω):

Sound field collection and playback device including And is disposed outwardly on a sphere of radius R _m by at least three microphones are fixed to the baffle of the sound absorbing material of the spherical radius R _b, the spherical surface of the at least three loudspeakers radius R _s R _m > R _b , R _s ≧ R _b , j is an imaginary unit, ω is frequency, c is sound velocity, k = ω / c, n, m is the order of the spherical harmonic spectrum, j _n (•) is the nth-order spherical Bessel function, h _n ⁽¹⁾ (•) is the nth-order first-class spherical Hankel function, and A (ω) is a predetermined value. As a complex number and w _nm as a weight determined based on n and m,
The conversion filter unit applies a filter F to _nm (ω) defined by the following equation to the spherical harmonic spectrum signal P to _nm (ω) generated based on the signal collected by the microphone. A transform filter step for generating a post-process signal D ~ _nm (ω);

A discrete spherical harmonic inverse transform unit converts the filtered signal D to _nm (ω) into a frequency domain signal by a discrete spherical harmonic inverse transform,
A frequency inverse transform unit that transforms the frequency domain signal into a time domain signal by inverse Fourier transform, and outputs the transformed time domain signal to the speaker;
Sound field collection and playback method including And is disposed outwardly on a sphere of radius R _m by at least three microphones are fixed to the baffle of the sound absorbing material of the spherical radius R _b, the spherical surface of the at least three loudspeakers radius R _s R _m > R _b , R _s ≧ R _b , j is an imaginary unit, ω is frequency, c is sound velocity, k = ω / c, n, m is the order of the spherical harmonic spectrum, j _n (•) is the nth-order spherical Bessel function, h _n ⁽¹⁾ (•) is the nth-order first-class spherical Hankel function, and A (ω) is a predetermined value. As a complex number and w _nm as a weight determined based on n and m,
A frequency conversion step in which the frequency conversion unit converts the signal collected by the microphone into a frequency domain signal by Fourier transform;
A discrete spherical harmonic transformation unit that transforms the frequency domain signal into a spherical harmonic spectral signal P to _nm (ω) by discrete spherical harmonic transformation;
Conversion filter unit generates the spherical harmonic spectrum signal P ~ _nm (ω) for the filter F ~ _nm (ω) by applying a filtering process after signals D ~ _nm defined by the following equation (omega) conversion A filter step;

Sound field collection and playback method including   A sound field recording / reproducing program for causing a computer to function as each part of the sound field sound collecting / reproducing apparatus according to claim 1.

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