JP2012147413A - Narrow directional sound reproduction processing method, device, and program - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a narrow directional sound reproduction processing technique that reproduces sound with a sufficient SN ratio, can reproduce sound toward any direction, and has sharper directivity for a desired direction than before.SOLUTION: A filter is calculated for a direction in which sound is to be reproduced using a transfer characteristic aof sound from each speaker in each direction φ included in one or plural directions assumed as a travelling direction of sound. Each transfer characteristic ais represented by the sum of a transfer characteristic of direct sound toward a direction φ and respective transfer characteristics of one or more pieces of dual sound, the dual sound being sound emitted from a speaker array and being sound with a travelling direction of reflection sound of the sound by a reflection object being parallel to the direction of the direct sound. The filter is applied to a frequency domain signal S obtained by converting a sound source signal into a frequency domain for each frequency to obtain M channels of frequency domain signals X.

Description

  The present invention relates to a signal processing technique (narrow-directed voice reproduction processing technique) for reproducing sound in a narrow range including a desired direction.

  As a situation of sound reproduction using a speaker, there is a situation where it is desired to reproduce sound at a sufficient volume in a specific direction. For example, when playing audio explaining an exhibit in a limited area in front of an exhibit in an exhibition hall, or in a limited area such as the edge of a station platform or near a staircase Is played. Such signal processing technology (narrow-directed speech reproduction processing technology) for reproducing sound in a narrow range including a desired direction (target direction) as viewed from the speaker has been researched and developed conventionally. The relationship (sound pressure distribution) between the surroundings of the speaker and the sound pressure of the sound emitted from the speaker is called directivity. The sharper the directivity in a certain direction, the more the sound is transmitted to a narrow range including that direction. It is possible to reproduce and suppress the sound pressure of the sound in a range other than the range. Here, three conventional techniques related to the narrow-directional sound reproduction processing technique are illustrated. In this specification, “speech” is not limited to a voice uttered by a person, but refers to a general “sound” such as a musical sound or an environmental noise as well as a voice of a person or an animal.

[1] Narrow-directional audio reproduction processing technology using physical characteristics Representative examples of this category include horn speakers and parabolic speakers. The horn speaker is, for example, a speaker in which a saddle-shaped horn whose cross-sectional area gradually increases toward the opening end in front of the speaker. The longer the horn length, the sharper the directivity of the horn speaker. A parabolic speaker has a configuration in which a speaker is arranged at the focal point of a parabolic plate (paraboloid), and radiates sound from the speaker toward the parabolic plate, thereby connecting the top of the parabolic plate and the focal point of the parabolic plate. Sound is transmitted in the direction of.

[2] Narrow-directional sound reproduction processing technology using ultrasonic waves A typical example of this category is a parametric speaker (see, for example, Patent Document 1). The parametric speaker radiates a modulated wave obtained by amplitude-modulating an ultrasonic wave with a sound source signal with a high sound pressure, using an ultrasonic wave having a strong straightness as a carrier wave. A distortion component is generated by the non-linear characteristic of air in the process in which the modulated wave propagates in the air, and sound in an audible band appears due to the distortion component and human auditory characteristics.

[3] Narrow-directional audio reproduction processing technology using signal processing A typical example of this category is a phased speaker array (see, for example, Non-Patent Document 1). A phased speaker array is a speaker array composed of a plurality of speakers, and signals obtained by performing signal processing to superimpose a sound source signal by applying a filter containing information on time difference or level difference to each speaker. And the sound is spatially radiated, and as a result, the sound is reproduced in the target direction.

JP 2010-258938 A

Yoichi Haneda, Akitoshi Kataoka, “Real-space performance of small speaker array based on multipoint control using free space transfer function”, Acoustical Society of Japan Spring Proceedings, pp.631-632, 2008.

  According to the narrow-directional sound reproduction processing technology described in category [1], as can be understood from the example of the horn speaker and the parabolic speaker, for example, the sound cannot be reproduced in the target direction unless the speaker itself is directed. . That is, when the target direction can be changed, if it does not depend on the human physical activity, a drive control means for changing the direction of the horn speaker or the parabolic speaker itself is required. In addition, it is difficult for both the horn speaker and the parabolic speaker to achieve a narrow directivity (a sharp directivity of about ± 5 ° to ± 10 ° with respect to the target direction) having an expected angle of about 5 ° to 10 °, for example. .

  The narrow-directional sound reproduction processing technology described in category [2] is excellent in terms of narrow directivity, but sound cannot be reproduced in the target direction unless the parametric speaker itself is directed in the target direction. That is, when the target direction can be changed, if it does not depend on the human physical activity, a drive control means for changing the direction of the parametric speaker itself is required. In addition, there is a problem that is still being investigated about ultrasonic exposure (whether there is no health problem when exposed to a high volume of ultrasonic waves).

  According to the narrow-directional sound reproduction processing technology described in category [3], in order to realize narrow directivity, it is necessary to increase the number of speakers and increase the array size (the total length of the array). It is not realistic to increase the array size indefinitely from the viewpoints of space restrictions for installing the phased speaker array, cost, the number of speakers that can execute real-time processing, and the like. For example, the maximum value of a signal that can be processed in real time with a commercially available speaker is about 100, and the directivity that can be realized with a phased speaker array using about 100 speakers is ± 30 with respect to the target direction. It is difficult to reproduce the sound in the target direction with a sharp directivity of, for example, about ± 5 ° to ± 10 °. Also, with the prior art of category [3], it is difficult to reproduce audio with a high S / N ratio in the target direction so as not to be buried in audio in directions other than the target direction.

  In view of such a current situation, the present invention reproduces sound with a sufficient signal-to-noise ratio and can reproduce sound in any direction without requiring physical movement of the speaker, but in a desired direction. On the other hand, an object of the present invention is to provide a narrow-directional sound reproduction processing technique having sharper directivity than conventional ones.

Using the audio transmission characteristics a _φ from the M speakers for each direction φ included in one or a plurality of directions assumed as the audio traveling direction, a filter is obtained for the direction that is the target of audio reproduction [Filter Design process]. M is an integer greater than or equal to 2, and M speakers constitute a speaker array. (1) The sound radiated from the speaker array, (2) the sound reflected by the reflector and the direction of travel of the reflected sound is the direction φ, the dual sound, each transfer characteristic a _φ is , Expressed as the sum of the direct sound transfer characteristics in the direction φ and the transfer characteristics of one or more dual sounds. The filter converts the frequency domain signal S obtained by converting the sound source signal into the frequency domain for each frequency into an M channel frequency domain signal X. The filter obtained by the filter design process is applied to the frequency domain signal S for each frequency to obtain an M channel frequency domain signal X [filter application process]. The M channel time domain signal x obtained by converting the M channel frequency domain signal X into the time domain is normally reproduced by a speaker array.

Each transfer characteristic a _φ is, as a specific example, a sum of a direct sound steering vector and one or more dual sound steering vectors in which a time difference with respect to the direct sound of the sound due to reflection attenuation and reflection is corrected, or It may be obtained by actual measurement in an actual environment.

  In the filter design process, a filter may be obtained for each frequency so that the power of the sound in a direction other than the direction for sound reproduction is minimized. Or you may obtain | require a filter for every frequency so that the S / N ratio in the direction used as the object of audio | voice reproduction may become the maximum. Alternatively, for each frequency, the sound power in one or more directions assumed as the sound traveling direction is minimized with the filter coefficient for one speaker among M speakers fixed at a constant value. You may ask for a filter.

Alternatively, in the filter design process, the sound reproduction is performed under the conditions of (1) passing the entire band of the sound in the direction to be reproduced and (2) suppressing the entire band of the sound to one or more blind spots. You may obtain | require a filter for every frequency so that the power of the audio | voice to directions other than a target direction and each blind spot may become the minimum. Alternatively, a filter may be obtained for each frequency by normalizing the transfer characteristic a _s in the direction φ = s to be reproduced. Or you may obtain | require a filter for every frequency using the spatial correlation matrix represented by the transfer characteristic a ( _phi) corresponding to each direction other than the direction used as the object of audio | voice reproduction | regeneration. Alternatively, for each frequency, the sound power in the direction other than the direction of the audio reproduction is minimized under the condition that the deterioration amount of the sound in the direction of the audio reproduction is a predetermined amount or less. A filter may be obtained. Or you may obtain | require a filter for every frequency using the spatial correlation matrix represented by the frequency domain signal obtained by converting the signal obtained by observing with a microphone array into a frequency domain.

According to the present invention, not only the direct sound in the sound reproduction target direction but also the reflected sound is used, so that the reproduction can be performed with a sufficiently large SN ratio in the direction and the sound reproduction in the direction by signal processing. Therefore, the sound can be reproduced in any direction without requiring physical movement of the speaker. Furthermore, as will be explained in detail in the item “Principle” described later, each transfer characteristic a _φ is expressed as the sum of the transfer characteristic of the direct sound in the direction φ and the transfer characteristic of one or more dual sounds. Thus, when designing a filter based on a general filter design standard, it is possible to design a filter that increases the degree of coherence suppression that determines the degree of directivity in the direction of sound reproduction. That is, it has a sharper directivity than the conventional one with respect to the direction of audio reproduction.

(A) A diagram schematically showing that narrow directivity cannot be sufficiently realized when only direct sound is considered, and (b) schematically showing that narrow directivity can be sufficiently realized when direct sound and reflected sound are considered. FIG. The figure which shows the direction dependence of the coherence in the case of based on the case of a prior art, and the principle of this invention. FIG. 3 is a diagram illustrating a functional configuration of the narrow-directional sound reproduction processing apparatus according to the first embodiment. FIG. 3 is a diagram illustrating a processing procedure of a narrow-directional sound reproduction processing method according to the first embodiment. The figure which shows the structure of a 1st Example. The figure which shows the experimental result of a 1st Example. The figure which shows the experimental result of a 1st Example. The figure which shows the directivity by filter W- ^> ((omega), (theta)) in a 1st Example. The figure which shows the structure of a 2nd Example. The figure which shows the experimental result of a 2nd Example. The figure which shows the experimental result of a 2nd Example. The figure which shows the implementation structural example of this invention. (A) Top view. (B) Front view. (C) Side view. (A) The side view which shows another implementation structural example of this invention. (B) The side view which shows another implementation structural example of this invention. The figure which shows the usage pattern in the implementation structural example shown in FIG.13 (b). The figure which shows the implementation structural example of this invention. (A) Top view. (B) Front view. (C) Side view. The side view which shows the implementation structural example of this invention. FIG. 5 is a diagram illustrating a functional configuration of a narrow-directional sound reproduction processing apparatus according to a second embodiment. FIG. 10 is a diagram illustrating a processing procedure of a narrow-directional sound reproduction processing method according to the second embodiment.

"principle"
The principle of the present invention will be described. The present invention is based on the essence of speaker array technology that can reproduce sound in an arbitrary direction based on signal processing, and on reproducing sound at a high S / N ratio by actively using reflected sound. On the other hand, it is characterized by combining signal processing technologies that enable sharp directivity.

Since signal processing in the frequency domain will be mainly described, symbols are defined prior to description. The index of the discrete frequency is ω (there is a relationship of ω = 2πf between the frequency f and the angular frequency ω, so the index ω of the discrete frequency may be identified with the angular frequency ω. The frequency index "is also simply referred to as" frequency "), and the frame number index is k. A filter designed with S (ω, k) as the frequency domain representation of the k-th frame of the sound source signal of one channel and the direction θ _s as the direction of the sound reproduction viewed from the center of the speaker array. The filter that converts the frequency domain signal S (ω, k) to the M-channel frequency domain signal W ^→ (ω, θ _s ), and the filter W ^→ (ω, k) to the frequency domain signal S (ω, k) of the sound source signal An M channel frequency domain signal (hereinafter referred to as a reproduction signal) obtained by applying θ _s ) is expressed as X ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k) ] M is an integer of 2 or more. At this time, the reproduction signal X ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k)] of the k-th frame is given by Expression (1). H represents Hermitian transpose. Note that the reproduction signal X of the k-th frame ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k)] is converted into an M channel time domain signal. Each time domain signal is reproduced by a speaker corresponding to each channel (details will be described later). The number of speakers is M.

The “center of the speaker array” can be arbitrarily determined. In general, the geometric center of the arrangement of the M speakers is set as the “center of the speaker array”. For example, a linear speaker array (M speakers are In the case of a speaker array arranged in a straight line), the midpoint of the speakers at both ends is set as the “center of the speaker array”. For example, a flat speaker array arranged in a square matrix of m × m (m ² = M) If there is, the position where the diagonal lines of the four corners of the speaker intersect is defined as the “center of the speaker array”.

There are various design methods for the filter W ^→ (ω, θ _s ), but here, a case based on a minimum variance distortion response method (MVDR method) will be described. In the minimum variance distortion-free response method, the filter W ^→ (ω, θ _s ) uses the spatial correlation matrix Q (ω) under the constraint condition of Equation (3) to generate a voice (in the direction other than the target direction θ _s ) Hereinafter, the power of “sound in a direction other than the target direction θ _s ” is also referred to as “leakage sound”) is designed to be a minimum at the frequency ω (see Expression (2)). a ^→ (ω, θ _s ) = [a ₁ (ω, θ _s ), ..., a _M (ω, θ _s )] ^T is assumed that there are listening positions in the direction θ _s , and M It is a transfer characteristic in the frequency (omega) between these speakers. T represents transposition. In other words, a ^→ (ω, θ _s ) = [a ₁ (ω, θ _s ), ..., a _M (ω, θ _s )] ^T is obtained from each speaker included in the speaker array in the direction θ _s . It is a transfer characteristic at the frequency ω of sound. The spatial correlation matrix Q (ω) is a collected sound signal obtained by observing with a microphone array composed of M microphones (preferably a microphone array in which the speakers included in the speaker array are replaced with microphones). Can be expressed using a frequency domain signal obtained by converting the signal to the frequency domain, but can also be expressed using transfer characteristics. Hereinafter, the case where the spatial correlation matrix Q (ω) is expressed using transfer characteristics for a while will be described.

It is known that the filter W ^→ (ω, θ _s ), which is the optimal solution of the equation (2), is given by the equation (4).
(Reference 1) by Simon Haykin, translated by Hiroshi Suzuki et al., "Adaptive Filter Theory", First Edition, Science and Technology Publishing Co., Ltd., 2001. pp.66-73,248-255

  As can be seen from the fact that the inverse matrix of the spatial correlation matrix Q (ω) is included in the equation (4), it can be seen that the structure of the spatial correlation matrix Q (ω) is important in realizing sharp directivity. It can also be seen from equation (2) that the power of the leaked speech depends on the structure of the spatial correlation matrix Q (ω).

A set to which the index p in the traveling direction (propagation direction) of the leaked speech belongs is assumed to be {1, 2,. It is assumed that the index s in the target direction θ _s does not belong to the set {1, 2,..., P-1}. At this time, the spatial correlation matrix Q (ω) is given by equation (5a). From the viewpoint of making a filter that realizes narrow directivity, P is preferably a somewhat large value, but it is assumed that P is an integer that satisfies P ≦ M. Here, from the viewpoint of explaining the principle of the invention in an easy-to-understand manner, the target direction θ _s is described as if it were a specific direction (therefore, directions other than the target direction θ _s are set as directions of “leakage sound”. ), As will be apparent from the embodiments described later, in practice, the target direction θ _s is an arbitrary direction that can be a target of audio reproduction, and generally, there are a plurality of directions as the directions that can be the target direction θ _s. is assumed. From this point of view, the distinction between the target direction θ _s and the direction of the leaked voice is almost subjective, and P is defined as a plurality of directions that are assumed as the voice traveling direction without distinguishing between the reproduced voice and the leaked voice. It is more accurate to predetermine different directions and to understand that one of the P directions selected is the target direction and the other direction is the direction of leaked speech. Therefore, if the union of the set {1, 2, ..., P-1} and the set {s} is Φ, the spatial correlation matrix Q (ω) is included in multiple directions that are assumed as the direction of speech progression. voice transmission characteristic a ^→ from the speakers for each direction theta _phi to _{(ω, θ φ) = [} a 1 (ω, θ φ), ..., a M (ω, θ φ)] T (φ∈Φ) Is a spatial correlation matrix expressed by Equation (5b). Note that | Φ | = P. | Φ | represents the number of elements of the set Φ.

Here, the sound transfer characteristic a ^→ (ω, θ _s ) in the target direction θ _s and the sound transfer characteristic a ^→ (ω, θ _{p in the} direction p∈ {1,2, ..., P-1} ) = [a ₁ (ω, θ _p ), ..., a _M (ω, θ _p )] Assume that ^T are orthogonal to each other. That is, it is assumed that there are P orthogonal basis sets that satisfy the condition expressed by Equation (6). The symbol ⊥ represents orthogonality. If A ^→ ⊥B ^→ , the inner product value of vector A ^→ and vector B ^→ is zero. Here, P ≦ M is satisfied. Note that when the condition expressed by Equation (6) is relaxed and it can be assumed that there are P basis sets that can be approximately regarded as orthogonal basis sets, P is about M, or somewhat larger than M. It is preferably a value.

At this time, the spatial correlation matrix Q (ω) can be expanded as shown in Equation (7). Equation (7) is a matrix V (ω) = [a ^→ (ω, θ _s ), a ^→ (ω, θ ₁ ),..., A ^→ (ω , θ _P-1 )] ^T and unit matrix Λ (ω) means that the spatial correlation matrix Q (ω) can be decomposed. ρ is an eigenvalue and a real number of the transfer characteristic a ^→ (ω, θ _φ ) satisfying the equation (6) based on the spatial correlation matrix Q (ω).

At this time, the inverse matrix of the spatial correlation matrix Q (ω) is given by equation (8).

Substituting equation (8) into equation (2) reveals that the power of leaked speech is minimized. If the power of the leaked voice is minimized, directivity with respect to the target direction θ _s is realized. Therefore, the fact that orthogonality is established between transfer characteristics in different directions is an important condition for realizing directivity with respect to the target direction θ _s .

Hereinafter, the reason why it is difficult to realize a sharp directivity with respect to the target direction θ _s in the prior art will be considered.

In the prior art, the filter is designed on the assumption that the transfer characteristic is composed only of direct sound. In reality, the sound emitted from the speaker is reflected by walls, ceilings, and the like, so there is a reflected sound. However, the reflected sound is considered to be a factor that deteriorates the directivity, and the presence of the reflected sound is ignored. If the steering vector of only direct sound in the direction θ is h ^→ _d (ω, θ) = [h _d1 (ω, θ), ..., h _dM (ω, θ)] ^T , conventionally, the transfer characteristic a ^→ _conv (ω, θ) = [a ₁ (ω, θ),..., a _M (ω, θ)] ^T is set as a ^→ _conv (ω, θ) = h ^→ _d (ω, θ). The steering vector is a complex vector in which the phase response characteristics at the frequency ω of each speaker with respect to the reference point are arranged for sound waves in the direction θ as viewed from the center of the speaker array.

Assuming that sound is radiated as a plane wave from a linear speaker array, the m-th element h _dm (ω, θ) constituting the direct sound steering vector h ^→ _d (ω, θ) is given by, for example, equation (9a) It is done. m is an integer satisfying 1 ≦ m ≦ M. c represents the speed of sound, and u represents the distance between adjacent speakers. j is an imaginary unit. The reference point is a position half the total length of the linear speaker array (the center of the linear speaker array). The direction θ is defined as an angle formed by the direct sound direction and the arrangement direction of the speakers included in the linear speaker array as seen from the center of the linear speaker array (see FIG. 5). There are various ways to express the steering vector. For example, if the reference point is the position of the speaker at one end of the linear speaker array, the mth element constituting the steering vector h ^→ _d (ω, θ) of the direct sound. h _dm (ω, θ) is given by, for example, equation (9b). In the following description, it is assumed that the m-th element h _dm (ω, θ) constituting the direct sound steering vector h ^→ _d (ω, θ) is given by equation (9a).

The inner product value γ _conv (ω, θ) of the transfer characteristic in the direction θ and the transfer characteristic in the target direction θ _s is expressed by Expression (10). Note that θ ≠ θ _s .

Hereinafter, γ _conv (ω, θ) is referred to as coherence. The direction θ in which the coherence γ _conv (ω, θ) becomes 0 is given by the equation (11). q is an arbitrary integer other than 0. In addition, since 0 <θ <π / 2, the range of q is limited for each frequency band.

In Formula (11), the only parameters that can be changed are the parameters (M and u) related to the size of the speaker array. Therefore, when the direction difference (angle difference) | θ−θ _s | is small, the speaker array It is difficult to reduce the coherence γ _conv (ω, θ) without changing the parameters related to the size of. In this case, the power of the leaked voice is not sufficiently reduced, and the directivity having a wide beam width with respect to the target direction θ _{s is} obtained as schematically shown in FIG.

On the other hand, according to the present invention, based on such consideration, the filter design for having a sharp directivity with respect to the target direction θ _s is coherence even when the direction difference (angle difference) | θ−θ _s | is small. Unlike the prior art, based on the knowledge that it is important to be able to reduce the noise sufficiently, it is characterized by positively considering reflected sound.

  Here, “dual sound” is defined. (1) Voice radiated from a speaker array, and (2) voice that satisfies the condition that the voice is reflected by a reflector and the reflected sound travel direction is the target direction. "

Assuming that the sound wave is a plane wave, the sound from each speaker of the speaker array that travels without being reflected in any direction (direct sound) and the reflection in which the dual sound is reflected by the reflector 300 are assumed in a certain direction θ. Two kinds of plane waves with sound will head. Let the number of reflected sounds (or dual sounds) be Ξ. Ξ is a predetermined integer of 1 or more. At this time, the transfer characteristic a ^→ (ω, θ) = [a ₁ (ω, θ),..., A _M (ω, θ)] ^T is the direct sound transfer characteristic from the speaker array to the direction θ and the direct sound The sum of the transfer characteristics of one or more dual sounds corresponding to the sound, specifically, the time difference between the direct sound and the ξ-th (1 ≦ ξ ≦ Ξ) reflected sound is τ _ξ (θ), and α _{When ξ} (1 ≦ ξ ≦ Ξ) is a coefficient for considering the sound attenuation due to reflection, the direct sound steering vector, the sound attenuation due to reflection, and the direct sound of the reflected sound are expressed as in Expression (12a). It can be expressed as the sum of steering vectors of two dual tones with corrected time differences. h ^→ _rξ (ω, θ) = [h _r1ξ (ω, θ),..., h _rMξ (ω, θ)] ^T represents the steering vector of the dual sound corresponding to the direct sound in the direction θ. α _ξ (1 ≦ ξ ≦ Ξ) is usually α _ξ ≦ 1 (1 ≦ ξ ≦ Ξ). For each reflected sound, if the number of times that the sound (dual sound) from the speaker array is reflected by the reflecting object is 1, α _ξ (1 ≦ ξ ≦ Ξ) is the ξth dual sound reflecting object. You can think of it as representing the reflectance of sound.

Since it is desired that one or more reflected sounds exist for a speaker array composed of M speakers, it is preferable that one or more reflectors exist. From this point of view, assuming that there is a listening position in the target direction, the positional relationship between the listening position, the speaker array, and one or more reflectors is that the sound (dual sound) from the speaker array reflects at least one reflection. It is preferable that each reflector is arranged so as to be reflected by the object and reach the listening position. Each reflector has a two-dimensional shape (for example, a flat plate) or a three-dimensional shape (for example, a parabolic shape). Moreover, it is preferable that the size of each reflector is equal to or larger than the speaker array (about 1 to 2 times). In order to effectively use the reflected sound, the reflectance α _ξ (1 ≦ ξ ≦ Ξ) of each reflector is at least greater than 0, and more specifically, the amplitude of the reflected sound reaching the listening position is the direct sound. It is desirable that the amplitude is, for example, 0.2 times or more. For example, each reflector is a rigid solid. The reflecting object may be a movable object (for example, a reflector) or an immovable object (a floor, a wall, or a ceiling). If an immovable object is set as a reflection object, it is necessary to change the steering vector of the dual sound as the installation position of the speaker array is changed (functions Ψ (θ) and Ψ _ξ (θ) described later are changed). (Refer to the above), and the filter calculation must be redone (reset). Therefore, in order to be robust against environmental changes, it is preferable that each reflector is a follower of a speaker array (in this case, it is assumed that the estimated number of reflected sounds is due to each reflector. become). Here, “speaker array subordinate” refers to “a tangible object that can follow changes in the position and orientation of the speaker array while maintaining the positional relationship (geometric relationship) with respect to the speaker array”. A simple example is a configuration in which each reflector is fixed to a speaker array.

Hereinafter, from the viewpoint of specifically explaining the advantages of the present invention, Ξ = 1, the number of reflections of dual sounds is one, and there is one reflector at a position L meters away from the center of the speaker array. Assume. The reflector is a thick rigid body. In this case, since Ξ = 1, the subscript representing this is omitted, and the expression (12a) can be expressed as the expression (12b).

Dual sound steering vector h ^→ _r (ω, θ) = [h _r1 (ω, θ),…, h _rM (ω, θ)] The m-th element of ^T Similarly (see equation (9a)), it is represented by equation (13a). The function Ψ (θ) outputs the traveling direction of the dual sound viewed from the center of the speaker array. When the direct sound steering vector is expressed by equation (9b), the dual sound steering vector h ^→ _r (ω, θ) = [h _r1 (ω, θ),..., H _rM (ω, θ) ] The m-th element of ^T is expressed by Expression (13b). In general, the _ξth (1 ≦ ξ ≦ Ξ) steering vector h ^→ _rξ (ω, θ) = [h _r1ξ (ω, θ),…, h _rMξ (ω, θ)] The mth element of ^T Is represented by Expression (13c) or Expression (13d). The function Ψ _ξ (θ) outputs the traveling direction of the ξ-th (1 ≦ ξ ≦ Ξ) dual sound viewed from the speaker array.

  Since the position of the reflector can be set as appropriate, the traveling direction of the dual sound can be treated as a variable parameter.

Assuming that a flat reflector is in the vicinity of the speaker array (distance L is not extremely large compared to the size of the speaker array), coherence γ (ω, θ) is expressed by equation (14). Note that θ ≠ θ _s .

From equation (14), it can be seen that the coherence γ (ω, θ) of equation (14) may be smaller than the conventional coherence γ _conv (ω, θ) of equation (11). There are parameters (Ψ (θ) and L) that can be changed depending on how the reflector is placed in the second to fourth items of equation (14), so the first item h ^→ _d ^H (ω, θ) h ^→ _d (ω, θ) may be removed.

For example, for a linear speaker array, when a flat reflector is arranged so that the speaker arrangement direction is the normal of the reflector, Ψ (θ) = π-θ holds for the function Ψ (θ), and Since the equation (15) is established for the time difference τ (θ) between the sound and the reflected sound, the conditions of the equations (16) and (17) are generated for the elements constituting the equation (14). The symbol * is an operator representing a complex conjugate.

_{^{h → d H (ω, θ}} ) the absolute value of h ^→ _r (ω, θ) is _{^{h → d H (ω, θ}} ) h → d (ω, θ) sufficiently smaller than, the formula (14) If the second and third terms are ignored, the coherence γ (ω, θ) can be approximated as shown in equation (18).

If _{^{h → d H (ω, θ}} ) h → d (ω, θ) even as ≠ 0, approximate coherence γ ~ (ω, θ) has a local minimum θ of equation (19). q is an arbitrary positive integer. Further, the range of q is limited for each frequency band.

  That is, the coherence can be suppressed not only in the direction given by Expression (11) but also in the direction given by Expression (19). If the coherence can be suppressed, the power of the leaked voice can be reduced, so that a sharp directivity can be realized as schematically shown in FIG.

FIG. 1 schematically shows the difference in directivity between the case of using the principle of the present invention and the case of using the prior art. In FIG. 2, θ given by Expression (11) and Expression (19) are given. The difference in θ obtained will be specifically shown. ω = 2π × 1000 [rad / s], L = 0.70 [m], θ _s = π / 4 [rad]. FIG. 2 shows the direction dependency of the normalized coherence for comparison between the two. The direction indicated by the symbol ○ is θ given by the equation (11), and the direction indicated by the symbol + is It is (theta) given by Formula (19). As is apparent from FIG. 2, according to the prior art, the coherence becomes zero with respect to θ _s = π / 4 [rad] only in the direction indicated by the symbol ○, but according to the principle of the present invention. And θ _s = π / 4 [rad] and coherence is zero in many directions indicated by the symbol +, and in particular, θ _s = π / 4 [ Since the direction indicated by the symbol + exists in a direction much closer to rad], it can be understood that sharp directivity is realized as compared with the prior art.

As is apparent from the above description, the main feature of the present invention is that the transfer characteristic a ^→ (ω, θ) = [a ₁ (ω, θ),..., A _M (ω, θ)] ^T , for example, As shown in Expression (12a), this is expressed by the sum of the steering vector of the direct sound and the steering vectors of the dual dual sounds. Accordingly, since the filter design concept itself is not affected, the filter W ^→ (ω, θ _s ) can be designed by a method other than the minimum variance distortionless response method.

Other than the above-mentioned minimum variance distortion-free response method, <1> filter design method based on S / N ratio maximization criteria, <2> filter design method based on Power Inversion, <3> one or more blind spots (4) Filter design method based on minimum variance no distortion response method with the constraint (the direction in which the gain of leaked speech is suppressed), <4> Filter design method based on Delay-and-Sum Beam Forming method, <5> A filter design method using the maximum likelihood method and a filter design method using the <6> AMNOR (Adaptive Microphone-array for Noise Reduction) method will be described. Refer to Reference 2 for the filter design method based on <1> SNR maximization criteria and <2> filter design method based on power inversion. <3> See Reference 3 for the filter design method based on the minimum variance-distortion response method with one or more blind spots (the direction in which the gain of leaked speech is suppressed) as a constraint. <6> See Reference 4 for the filter design method based on the AMNOR (Adaptive Microphone Array for Noise Reduction) method.
(Reference 2) Nobuyoshi Kikuma, “Adaptive Antenna Technology”, 1st Edition, Ohm Corporation, 2003, pp.35-90
(Reference 3) Taiko Asano, “The Acoustical Society of Japan, Acoustic Techno Series 16 Sound Array Signal Processing-Sound Source Localization, Tracking and Separation”, First Edition, Corona Inc., pp.88-89, 259-261
(Reference 4) Yutaka Kaneda, "Directivity characteristics of adaptive noise suppression microphone array (AMNOR)", Journal of the Acoustical Society of Japan, Vol. 44 No. 1 (1988), pp.23-30

<1> Filter design method based on S / N ratio maximization criteria
In the filter design method based on the SN ratio maximization criterion, the filter W ^→ (ω, θ _s ) is determined based on the criterion for maximizing the SN ratio (SNR) in the target direction θ _s . The spatial correlation matrix of the audio in the target direction θ _s R _ss (ω), the spatial correlation matrix of the audio in a direction other than the target direction theta _s and R _nn (ω). At this time, the SNR is expressed by Expression (20). Note that R _ss (ω) is expressed by Expression (21), and R _nn (ω) is expressed by Expression (22). Transfer characteristic a ^→ (ω, θ _s ) = [a ₁ (ω, θ _s ),..., A _M (ω, θ _s )] ^T is expressed by equation (12a) (exactly, equation (12a ) Is θ _s ).

The filter W ^→ (ω, θ _s ) that maximizes the SNR in Expression (20) can be obtained by setting the gradient related to the filter W ^→ (ω, θ _s ) to zero, that is, Expression (23).

Thus, the filter W ^→ (ω, θ _s ) that maximizes the SNR in Expression (20) is given by Expression (24).

Although the formula (24) includes an inverse matrix of the target direction theta sound space correlation matrix in a direction other than _s R _nn (omega), an inverse matrix of R _nn (omega), the target direction theta _s It is known that it may be replaced with the inverse matrix of the spatial correlation matrix R _xx (ω) of the entire input including the voice of the voice and the voice in the direction other than the target direction θ _s . Note that R _xx (ω) = R _ss (ω) + R _nn (ω) = Q (ω) (see Expression (5a), Expression (21), and Expression (22)). That is, the filter W ^→ (ω, θ _s ) that maximizes the SNR in Expression (20) may be obtained by Expression (25).

<2> In the filter design method based on the filter design method power inversion based on power inversion, filter criteria to minimize the average output power of the beamformer filter coefficients in a fixed state to a constant value for one speaker W ^→ Determine (ω, θ _s ). Here, as an example, it is assumed that the filter coefficient for the first speaker among the M speakers is fixed. In this design method, the filter W ^→ (ω, θ _s ) is assumed to be omnidirectional (the direction of travel of the sound from the speaker array) using the spatial correlation matrix R _xx (ω) under the constraint condition of Equation (27). Are designed so as to minimize the power of the sound in all directions) (see equation (26)). Transfer characteristic a ^→ (ω, θ _s ) = [a ₁ (ω, θ _s ),..., A _M (ω, θ _s )] ^T is expressed by equation (12a) (exactly, equation (12a ) Is θ _s ). Note that R _xx (ω) = Q (ω) (see Expression (5a), Expression (21), and Expression (22)).

It is known that the filter W ^→ (ω, θ _s ), which is the optimal solution of the equation (26), is given by the equation (28) (see Reference 2).

<3> Filter design method using minimum variance distortionless response method with one or more blind spots as constraint conditions In the above minimum variance distortionless response method, the speech of the target direction θ _s is expressed as expressed by equation (3). A filter in which the average output power of the beamformer represented by Equation (2) is minimized (that is, the power of leaked speech that is speech in a direction other than the target direction is minimized) with the all-band pass as a constraint. The filter W ^→ (ω, θ _s ) was designed based on the criterion under the single constraint condition of obtaining. According to this method, the power of leaked voice can be suppressed as a whole, but it is not necessarily a preferable method when it is desired to strongly suppress voice propagation in one or more specific directions. In such a case, a filter that strongly suppresses one or more known specific directions (that is, blind spots) is required. For this reason, in the filter design method described here, (1) the entire band of the voice in the target direction θ _s is passed, and (2) known B (B is a predetermined integer of 1 or more) blind angle θ _N1. , Θ _N2 ,..., Θ _NB all the band suppression of the voice, and the average output power of the beamformer represented by Equation (2) is minimized (that is, the direction excluding the target direction and each blind spot) Find the filter that minimizes the power of the voice. As described above, if the set to which the index φ in the voice propagation direction belongs is {1, 2, ..., P}, Nj∈ {1,2, ..., P} (where j∈ {1,2, ..., B}), B ≦ P-1.

At this time, a ^→ (ω, θ _i ) = [a ₁ (ω, θ _i ), ..., a _M (ω, θ _i )] ^T , the listening position is in the direction θ _s and the direction θ _Nj (where j∈ {1,2, ..., B}) is a blind spot, and the frequency ω between direction θ _i (where i∈ {s, N1, N2, ..., NB}) and M speakers , In other words, a ^→ (ω, θ _i ) = [a ₁ (ω, θ _i ),..., A _M (ω, θ _i )] ^T is obtained from each speaker included in the speaker array. When the transfer characteristic at the frequency ω of the sound in the direction θ _i is assumed, the constraint condition is expressed by Expression (29). However, for index i, i∈ {s, N1, N2,..., NB}, and transfer characteristics a ^→ (ω, θ _i ) = [a ₁ (ω, θ _i ),…, a _M (ω, θ _i )] ^T is expressed by equation (12a) (more precisely, θ in equation (12a) is θ _i ). f _i (ω) represents the pass characteristic at the frequency ω with respect to the direction θ _i .

If Expression (29) is expressed in matrix form, it can be expressed as Expression (30), for example. However, A ^→ (ω, θ _s ) = [a ^→ (ω, θ _s ), a ^→ (ω, θ _N1 ), ..., a ^→ (ω, θ _NB )].

Considering the constraint conditions of (1) all-band passage of speech in the target direction θ _s and (2) all-band suppression of speech of known B dead angles θ _N1 , θ _{N2 ,.} In this case, f _s (ω) = 1.0 and f _i (ω) = 0.0 (i∈ {N1, N2,..., NB}) should be satisfied. This represents the full-band complete passage of the sound in the target direction θ _s and the full-band complete blocking of the sound of the known B dead angles θ _N1 , θ _{N2 ,.} However, in reality, it may be difficult to control full band full passage or full band full blocking. In such a case, the absolute value of f _s (ω) is set to a value close to 1.0, and the absolute value of f _i (ω) (i∈ {N1, N2,..., NB}) is set to a value close to 0.0. That's fine. Of course, f _i (ω) and f _j (ω) (i ≠ j, i, j∈ {N1, N2,..., NB}) may be equal or different.

According to the filter design method described here, the filter W ^→ (ω, θ _s ), which is the optimal solution of the expression (2) under the expression (29) representing the constraint condition, is given by the expression (31) (reference Reference 3).

<4> Filter Design Method Using Delay Synthesis Method According to the delay synthesis method, assuming that direct sound or reflected sound propagates through a plane wave, the filter W ^→ (ω, θ _s ) is given by equation (32). That is, the filter W ^→ (ω, θ _s ) is obtained by normalizing the transfer characteristic a ^→ (ω, θ _s ). Transfer characteristic a ^→ (ω, θ _s ) = [a ₁ (ω, θ _s ),..., A _M (ω, θ _s )] ^T is expressed by equation (12a) (exactly, equation (12a ) Is θ _s ). According to this design method, the filter accuracy may not always be good, but the calculation amount is small.

<5> Filter design method based on maximum likelihood method In the above-mentioned minimum variance distortion-free response method, the degree of freedom to suppress leaked speech by not including spatial information of speech in the target direction in the spatial correlation matrix Q (ω) And the power of leaked voice can be further suppressed. For this reason, in the filter design method described here, the spatial correlation matrix Q (ω) is represented by the second term on the right side of Equation (5a), that is, Equation (5c). The filter W ^→ (ω, θ _s ) is given by Equation (4) or Equation (31). At this time, Q (ω) included in Equation (4) or Equation (31) or R _xx (ω) = Q (ω) included in Equation (25) or Equation (28) is expressed by Equation (5c). Is a spatial correlation matrix.

<6> Filter design method by AMNOR method
The AMNOR method allows a certain amount of speech degradation amount D in the target direction based on the trade-off relationship between the speech degradation amount D in the target direction and the power of noise remaining in the filter output signal (for example, the degradation amount D). Is kept below a certain threshold value D ^), [a] a signal obtained by applying a transfer characteristic between a sound source and a microphone to a virtual signal in a target direction (hereinafter referred to as a virtual target signal) [b] The filter output signal when the mixed signal with noise (for example, obtained by observation with M microphones in a noise environment where there is no voice in the target direction) is input is the virtual target signal most in terms of the least square error. This is a method for obtaining a filter that reproduces well (that is, the noise power included in the filter output signal is minimized).

The filter design method described here can be considered in the same way as the AMNOR method except that the input and output of the filter are reversed. That is, based on the trade-off relationship between the speech degradation amount D in the target direction and the power of leaked speech remaining in the filter output signal, the speech degradation amount D in the target direction is allowed to some extent (for example, the degradation amount D is kept below a certain threshold value D ^), and when the frequency domain signal S (ω, k) of the sound source signal is input, the filter output signal is the frequency domain signal S (ω , k) is obtained best (that is, the power of the leaked voice included in the filter output signal is minimized). The filter output signal is a signal (hereinafter referred to as an audible signal) obtained by applying a transfer characteristic at the frequency ω of sound from each speaker included in the speaker array in the target direction θ _s to the frequency domain signal S (ω, k). And [b] noise (obtained for example by observation with M microphones in a noisy environment).

According to the filter design method described here, as in the AMNOR method, the filter W ^→ (ω, θ _s ) is given by Equation (33) (see Reference 4). Note that R _ss (ω) is expressed by Expression (21), and R _nn (ω) is expressed by Expression (22). Transfer characteristic a ^→ (ω, θ _s ) = [a ₁ (ω, θ _s ),..., A _M (ω, θ _s )] ^T is expressed by equation (12a) (exactly, equation (12a ) Is θ _s ).

P _s is a coefficient for weighting the level of the listening signal, and is referred to as the listening signal level. The listening signal level P _s is a constant independent of frequency. The listening signal level P _s may be determined based on an empirical rule, or determined so that the difference between the audio degradation amount D in the target direction and the threshold D ^ is within an arbitrarily determined error range. May be. The latter example will be described. At the frequency ω, the frequency response F (ω) of the sound in the target direction θ _s of the filter W ^→ (ω, θ _s ) is expressed by Expression (34). When the deterioration amount D when using the filter W ^→ (ω, θ _s ) given by the equation (33) is expressed as D (P _s ), the deterioration amount D (P _s ) is defined by the equation (35). . ω ₀ represents the upper limit of the target frequency ω (usually a high-frequency side adjacent to the discrete frequency ω). The deterioration amount D (P _s ) is a monotone decreasing function of P _s . Therefore, the monotonicity of the D (P _s), by repeating the determination of the deterioration amount D (P _s) while changing the P _s, deterioration amount D (P _s) and the difference between any of the threshold D ^ A listening signal level P _s within a predetermined error range can be obtained.

<Modification>
In the above description, the spatial correlation matrices Q (ω), R _ss (ω), and R _nn (ω) are expressed using transfer characteristics. However, as described above, the spatial correlation matrix Q (ω), R _ss (ω), R _nn (using the frequency domain signal obtained by converting the analog signal obtained by observation with the microphone array into the frequency domain. ω) can also be expressed. The spatial correlation matrix Q (ω) will be described below, but the same applies to R _ss (ω) and R _nn (ω) (Q (ω) can be read as R _ss (ω) or R _nn (ω). Just fine). Note that the spatial correlation matrix R _ss (ω) is obtained by the frequency domain representation of the analog signal obtained by observation with a microphone array (including M microphones) in an environment where only sound in the target direction exists. The correlation matrix R _nn (ω) is obtained by frequency domain representation of an analog signal obtained by observation with a microphone array (including M microphones) in an environment where there is no sound in the target direction (that is, a noise environment).

The spatial correlation matrix Q (ω) using the frequency domain signal U ^→ (ω, k) = [U ₁ (ω, k),..., U _M (ω, k)] ^T is expressed by Expression (36). . The operator E [•] is an operator representing a statistical average operation. When a discrete time series of analog signals received by M microphones is regarded as a stochastic process, the operator E [•] is the arithmetic mean (expected value) if it is a so-called steady or secondary stationary. It becomes an operation. In this case, the spatial correlation matrix Q (ω) is, for example, a frequency domain signal U ^→ (ω, ki) (i = 0, 1,..., Ζ of a total of ζ frames currently and past stored in a memory or the like. -1) is used to express the equation (37). When i = 0, that is, the k-th frame is the current frame. The spatial correlation matrix Q (ω) according to the equations (36) to (37) may be recalculated for each frame, or may be recalculated at regular or irregular intervals, or It may be calculated before the implementation of the embodiment described later (particularly when R _ss (ω) or R _nn (ω) is used for the filter design, it was obtained before the implementation of the embodiment. It is preferable to calculate the spatial correlation matrix Q (ω) in advance using the frequency domain signal). When recalculating the spatial correlation matrix Q (ω) for each frame, since the spatial correlation matrix Q (ω) depends on the current and past frames, the spatial correlation matrix is explicitly expressed as in Expression (36a) or Expression (37a). Let the correlation matrix be Q (ω, k).

If the spatial correlation matrix Q (ω, k) represented by the equations (36a) and (37a) is used, the filter W ^→ (ω, θ _s ) also depends on the current and past frames. W ^→ (ω, θ _s , k). At this time, the filter represented by any one of Expression (4), Expression (24), Expression (25), Expression (28), Expression (31), and Expression (33) described in the various filter design methods described above. W ^→ (ω, θ _s ) is corrected to the expression (4m), the expression (24m), the expression (25m), the expression (28m), the expression (31m), and the expression (33m) for notation.

Embodiment 1
The functional configuration and processing flow of the first embodiment of the present invention are shown in FIGS. The narrow-directional sound reproduction processing apparatus 1 according to the first embodiment includes an AD conversion unit 210, a frame generation unit 220, a frequency domain conversion unit 230, a filter application unit 240, a time domain conversion unit 250, a filter design unit 260, and a storage unit 290. Including.

[Step S1]
In advance, the filter design unit 260 calculates a filter W ^→ (ω, θ _i ) for each frequency for each discrete direction that can be a target of audio reproduction. The total number of I discrete directions that may be subject to audio playback (I is one or more predetermined integer, satisfying the I ≦ P) _{^{When, W → (ω, θ 1}} ), ..., W → (ω, θ _i ),..., W ^→ (ω, θ _I ) (1 ≦ i ≦ I, ω∈Ω; i is an integer, Ω is a set of frequencies ω) is calculated in advance.

For this purpose, except for the case described in <Modification> above, transfer characteristics a ^→ (ω, θ _i ) = [a ₁ (ω, θ _i ),..., A _M (ω, θ _i )] ^T (1 ≤ i ≤ I, ω ∈ Ω) needs to be obtained. This is because the speaker arrangement in the speaker array, the positional relationship of the reflectors such as the reflector, floor, wall, and ceiling with respect to the speaker array, directly Based on environmental information such as the time difference between the sound and the ξ-th (1 ≦ ξ ≦ Ξ) reflected sound, the reflectance of the sound of the reflector, etc., it can be specifically calculated by the equation (12a) (precisely, the equation (12a ) Of θ) is θ _i ). In the case of using the filter design method based on the above-mentioned <3> one or more blind spots and the minimum variance no distortion response method, the transfer characteristic a ^→ (ω, θ _i ) (1 ≦ i ≦ I, ω∈ It is desirable that the index i in the direction for obtaining Ω) spans all of the indices N1, N2,. In other words, the indexes N1, N2,..., NB in the direction of the B blind spots are set as any different integer from 1 to I.

The number 反射 of the reflected sounds (or dual sounds) is set to an integer that satisfies 1 ≦ Ξ, but the value of Ξ is not particularly limited and may be set appropriately according to the calculation capability. When one reflector is installed in the vicinity of the speaker array, the transfer characteristic a ^→ (ω, θ _i ) can be specifically calculated by the equation (12b) (more precisely, θ in the equation (12b) is θ _i ).

  For example, the formula (9a), the formula (9b), the formula (13a), the formula (13b), the formula (13c), and the formula (13d) can be used for the calculation of the steering vector. In addition, as a transfer characteristic used for filter design, you may use the transfer characteristic obtained by actual measurement in a real environment, for example, without depending on Formula (12a) and Formula (12b).

Then, except for the case described in <Modification> above, using the transfer characteristic a ^→ (ω, θ _i ), for example, Expression (4), Expression (24), Expression (25), Expression (28), W ^→ (ω, θ _i ) (1 ≦ i ≦ I) is obtained by any one of the equations (31), (32), and (33). Except for the case described in the filter design method based on the <5> maximum likelihood method described above, when using equation (4), equation (25), equation (28), or equation (31), the spatial correlation matrix Q ( ω) (or R _xx (ω)) can be calculated by equation (5b). Based on the above-described <5> maximum likelihood filter design method, when using equation (4), equation (25), equation (28), or equation (31), spatial correlation matrix Q (ω) (or R _xx (ω)) can be calculated by equation (5c). When using equation (24), the spatial correlation matrix R _nn (ω) can be calculated by equation (22). I × | Ω | filters W ^→ (ω, θ _i ) (1 ≦ i ≦ I, ω∈Ω) are stored in the storage unit 290. | Ω | represents the number of elements of the set Ω.

[Step S2]
The sound source 200 outputs a sound source signal ss (t). In the first embodiment, it is assumed that the sound source signal ss (t) from the sound source 200 is an analog signal. However, a digital signal can also be used as the sound source signal.

[Step S3]
The AD converter 210 AD converts the sound source signal ss (t) into a digital signal s (t). Here, t represents an index of discrete time. When the digital signal is a sound source signal, it is not necessary to perform the process of step S3, and the sound source signal can be regarded as s (t) that is an output signal of the AD conversion unit 210.

[Step S4]
The frame generation unit 220 receives the digital signal s (t) output from the AD conversion unit 210, stores N samples in a buffer, and outputs a digital signal s (k) in units of frames. k is an index of the frame number. s (k) = [s ((k−1) N + 1),..., s (kN)]. N depends on the sampling frequency, but in the case of 16kHz sampling, around 512 points is reasonable.

[Step S5]
The frequency domain converter 230 converts the digital signal s (k) of each frame into a frequency domain signal S (ω, k) and outputs it. ω is an index of discrete frequency. One method for converting a time domain signal to a frequency domain signal is a fast discrete Fourier transform, but the present invention is not limited to this, and other methods for converting to a frequency domain signal may be used. The frequency domain signal S (ω, k) is output for each frequency ω and for each frame k.

[Step S6]
The filter application unit 240 applies the filter W ^→ (ω, θ _s ) corresponding to the desired direction θ _s to be reproduced to the frequency domain signal S (ω, k) for each frequency ω∈Ω for each frame k. Thus, the reproduction signal X ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k)] is output (see Expression (38)). Since the index s of the target direction θ _s is s∈ {1,..., I}, and the filter W ^→ (ω, θ _s ) is stored in the storage unit 290, for example, every time the process of step S6 is performed, The filter application unit 240 may acquire the filter W ^→ (ω, θ _s ) corresponding to the target direction θ _s to be reproduced from the storage unit 290. When the index s of the target direction θ _s does not belong to the set {1,..., I}, that is, when the filter W ^→ (ω, θ _s ) corresponding to the target direction θ _s is not calculated in the process of step S1. The filter design unit 260 may calculate the filter W ^→ (ω, θ _s ) corresponding to the target direction θ _s temporarily, or the filter W ^→ ( ^→ corresponding to the direction θ _{s ′} close to the target direction θ _s (ω, θ _{s ′} ) may be used.

[Step S7]
The time domain conversion unit 250 uses the reproduction signal X ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k)] of each frequency ω∈Ω of the kth frame in the time domain. The frame unit time domain signal x ^→ (k) = [x ₁ (k),..., X _M (k)] of the k-th frame is obtained by conversion, and the obtained frame unit time domain signal x ^→ ( k) = [x ₁ (k),..., x _M (k)] are connected in the order of the index of the frame number, and the time domain signal x ^→ in which the sound is emphasized toward the target direction θ _s that is the reproduction direction (t) = [x ₁ (t), ..., x _M (t)] is output. The method for converting the frequency domain signal into the time domain signal is an inverse transformation corresponding to the transformation method used in the process of step S5, for example, a fast discrete inverse Fourier transform.

[Step S8]
M channel time domain signals x ₁ (t),..., X _M (t) are reproduced by the speakers corresponding to the channel among the M speakers 280-1,. Is done. That is, the time domain signal x _m (t) of the m-th (1 ≦ m ≦ M) channel is reproduced by the m-th speaker 280-m.

There is no limit to the way the M speakers are arranged. An array configuration in which speakers are linearly arranged like a linear speaker array may be used, or an array configuration in which M speakers are two-dimensionally or three-dimensionally arranged. In addition, in order to make the direction that can be set as the reproduction direction wide, the directivity of each speaker is such that the sound can be reproduced with a certain sound pressure in the direction that can be the target direction θ _s that is the reproduction direction. You should have it. Therefore, a speaker having relatively gentle directivity such as a non-directional speaker or a unidirectional speaker is preferable.

Here, the first embodiment in which the filter W ^→ (ω, θ _i ) is calculated in advance in the process of step S1 has been described. However, according to the calculation processing capability of the narrow-directional sound reproduction processing device 1 or the like, in the reproduction direction. An embodiment in which the filter design unit 260 calculates the filter W ^→ (ω, θ _s ) for each frequency after a certain target direction θ _s is determined may be employed.

<< Embodiment 2 >>
The functional configuration and processing flow of Embodiment 2 of the present invention are shown in FIGS. The narrow-directional sound reproduction processing apparatus 2 of the second embodiment includes an AD conversion unit 210, a frame generation unit 220, a frequency domain conversion unit 230, a filter application unit 240, a time domain conversion unit 250, a filter calculation unit 261, a storage unit 290, An AD conversion unit 310, a frame generation unit 320, and a frequency domain conversion unit 330 are included.

[Step S11]
The sound source 200 outputs a sound source signal ss (t). In the second embodiment, it is assumed that the sound source signal ss (t) from the sound source 200 is an analog signal. However, a digital signal can also be used as the sound source signal.

[Step S12]
The AD converter 210 AD converts the sound source signal ss (t) into a digital signal s (t). Here, t represents an index of discrete time. When the digital signal is a sound source signal, it is not necessary to perform the process of step S12, and the sound source signal can be regarded as s (t) that is an output signal of the AD conversion unit 210.

[Step S13]
The frame generation unit 220 receives the digital signal s (t) output from the AD conversion unit 210, stores N samples in a buffer, and outputs a digital signal s (k) in units of frames. k is an index of the frame number. s (k) = [s ((k−1) N + 1),..., s (kN)]. N depends on the sampling frequency, but in the case of 16kHz sampling, around 512 points is reasonable.

[Step S14]
The frequency domain converter 230 converts the digital signal s (k) of each frame into a frequency domain signal S (ω, k) and outputs it. ω is an index of discrete frequency. One method for converting a time domain signal to a frequency domain signal is a fast discrete Fourier transform, but the present invention is not limited to this, and other methods for converting to a frequency domain signal may be used. The frequency domain signal S (ω, k) is output for each frequency ω and for each frame k.

[Step S15]
The filter calculation unit 261 calculates a filter W ^→ (ω, θ _s , k) (ω∈Ω; Ω is a set of frequencies ω) for each frequency corresponding to the target direction θ _s used in the current k-th frame. To do.

For this purpose, it is necessary to prepare the transfer characteristic a ^→ (ω, θ _s ) = [a ₁ (ω, θ _s ), ..., a _M (ω, θ _s )] ^T (ω∈Ω) This is the arrangement of the speakers in the speaker array, the positional relationship of the reflectors such as the reflector, floor, wall, and ceiling with respect to the speaker array, the time difference between the direct sound and the ξth (1 ≦ ξ ≦ Ξ) reflected sound, environmental information such as the reflectivity of sound reflector can specifically calculated by the equation (12a) based on (accurately, in which a theta of formula (12a) was theta _s). In the case of using the filter design method based on the above-described <3> one or more blind spots and the minimum variance no distortion response method, the transfer characteristic a ^→ (ω, θ _Nj ) (1 ≦ j ≦ B, ω∈ Ω) also needs to be obtained, but these are the arrangement of speakers in the speaker array, the positional relationship of reflectors such as reflectors, floors, walls, and ceilings to the speaker array, direct sound and ξth (1 ≦ ξ ≦ Ξ) (12a) can be specifically calculated based on environmental information such as the time difference from the reflected sound and the reflectance of the sound of the reflector (more precisely, θ in the expression (12a) is θ _Nj. ).

The number 反射 of the reflected sound is set to an integer satisfying 1 ≦ Ξ, but the value of Ξ is not particularly limited and may be appropriately set according to the calculation ability. When one reflector is installed in the vicinity of the speaker array, the transfer characteristic a ^→ (ω, θ _s ) can be specifically calculated by the equation (12b) (more precisely, θ in the equation (12b) is θ _s ). In this case, similarly, the transfer characteristic a ^→ (ω, θ _Nj ) (1 ≦ j ≦ B, ω∈Ω) can be specifically calculated by the equation (12b) (more precisely, θ in the equation (12b) is θ _Nj ).

  For example, the formula (9a), the formula (9b), the formula (13a), the formula (13b), the formula (13c), and the formula (13d) can be used for the calculation of the steering vector. As a transfer characteristic used for filter design, a transfer characteristic obtained by actual measurement in an actual environment may be used, for example, without depending on Expression (12a) or Expression (12b).

Then, the filter calculation unit 261 calculates the transfer characteristic a ^→ (ω, θ _s ) (ω∈Ω) and, if necessary, the transfer characteristic a ^→ (ω, θ _Nj ) (1 ≦ j ≦ B, ω∈Ω). Using the filter W ^→ (ω, θ _s , k) (ω∈Ω), Equation (4m), Equation (24m), Equation (25m), Equation (28m), Equation (31m), Equation (33m) Ask according to either. Note that the spatial correlation matrix Q (ω) (or R _xx (ω)) can be calculated by, for example, Expression (36a) or Expression (37a). For the calculation of the spatial correlation matrix Q (ω), the frequency domain signal X ^→ (ω, ki) (i = 0, 1,..., Ζ− of the current and past frames accumulated in the storage unit 290 in total. 1) is used.

The frequency domain signal X ^→ (ω, k) is accumulated in the storage unit 290 as follows.
Sound is collected using M microphones 300-1,..., 300-M constituting the microphone array. The arrangement of the M microphones is preferably the same as that of the speaker array.
The AD converter 310 converts the analog signal (sound pickup signal) collected by the M microphones 300-1,..., 300-M into a digital signal x ^→ (t) = [x ₁ (t),. _M (t)]. t represents a discrete time index.
The frame generation unit 320 receives the digital signal x ^→ (t) = [x ₁ (t),..., X _M (t)] output from the AD conversion unit 310 and stores N samples in a buffer for each channel. The digital signal x ^→ (k) = [x ^→ ₁ (k),..., X ^→ _M (k)] is output in units of frames. k is an index of the frame number. x ^→ _m (k) = [x _m ((k−1) N + 1),..., x _m (kN)] (1 ≦ m ≦ M). N depends on the sampling frequency, but in the case of 16kHz sampling, around 512 points is reasonable.
The frequency domain transform unit 330 converts the digital signal x ^→ (k) of each frame into a frequency domain signal X ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k)]. Convert and output. ω is an index of discrete frequency. One method for converting a time domain signal to a frequency domain signal is a fast discrete Fourier transform, but the present invention is not limited to this, and other methods for converting to a frequency domain signal may be used. The frequency domain signal X ^→ (ω, k) is output for each frequency ω and for each frame k and accumulated in the storage unit 290.

[Step S16]
The filter application unit 240 applies the filter W ^→ (ω, θ _s , k) corresponding to the desired direction θ _s to be reproduced to the frequency domain signal S (ω, k) for each frequency ω∈Ω for each frame k. By applying this, the reproduction signal X ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k)] is output (see Expression (39)).

[Step S17]
The time domain conversion unit 250 uses the reproduction signal X ^→ (ω, k) = [X ₁ (ω, k),..., X _M (ω, k)] of each frequency ω∈Ω of the kth frame in the time domain. The frame unit time domain signal x ^→ (k) = [x ₁ (k),..., X _M (k)] of the k-th frame is obtained by conversion, and the obtained frame unit time domain signal x ^→ ( k) = [x ₁ (k),..., x _M (k)] are connected in the order of the index of the frame number, and the time domain signal x ^→ in which the sound is emphasized toward the target direction θ _s that is the reproduction direction (t) = [x ₁ (t), ..., x _M (t)] is output. The method of converting the frequency domain signal into the time domain signal is an inverse transform corresponding to the transform method used in the process of step S14, and is, for example, a fast discrete inverse Fourier transform.

[Step S18]
M channel time domain signals x ₁ (t),..., X _M (t) are reproduced by the speakers corresponding to the channel among the M speakers 280-1,. Is done. That is, the time domain signal x _m (t) of the m-th (1 ≦ m ≦ M) channel is reproduced by the m-th speaker 280-m.

The experimental results according to Embodiment 1 of the present invention (minimum dispersion no distortion response method under a single constraint condition) will be described. As shown in FIG. 5, 24 omnidirectional speakers are arranged linearly, and the reflector 300 is arranged such that the arrangement direction of the linear speaker array is a normal line of the reflector 300. Although there is no restriction | limiting in the shape of the reflecting plate 300, the reflecting surface is a plane, The flat reflecting plate which has a size of 1.0m x 1.0m, moderate thickness, and rigidity was used. The distance between adjacent speakers was 4 cm, and the reflectance α of the reflector was 0.8. The target direction θ _s was set to 45 degrees. Assuming that sound is radiated as a plane wave from the linear speaker array, the transfer characteristic is calculated by the equation (12b) (see the equations (9a) and (13a)), and the directivity of the generated filter is verified. . As a comparison object, the conventional method described in Non-Patent Document 1 (minimum dispersion no distortion response method without a reflector) was used.

The experimental results are shown in FIGS. Compared with the conventional method, it can be seen that the first embodiment of the present invention can achieve sharper directivity with respect to the target direction in any frequency band. In particular, the usefulness of the present invention is understood as the frequency band is lower (a human voice contains more frequency components from about 100 Hz to about 2 kHz). FIG. 8 shows the directivity by the filter W ^→ (ω, θ) generated according to the first embodiment of the present invention. From FIG. 8, it can be seen that not only the direct sound is transmitted in the target direction θ _s = 45 degrees, but also the sound is transmitted in the direction in which the reflector 300 is placed.

Further, as shown in FIG. 9, the case where the reflector 300 is arranged so that the angle formed by the arrangement direction of the speakers included in the linear speaker array and the plane of the reflector 300 is 45 degrees is the same as the above-described experiment. The experiment was conducted. The target direction θ _s was set to 22.5 degrees, and the other experimental conditions were the same as when the reflector 300 was arranged so that the arrangement direction of the speakers included in the linear speaker array was normal to the reflector 300.

  The experimental results are shown in FIGS. Compared with the conventional method, it can be seen that the first embodiment of the present invention can achieve sharper directivity with respect to the target direction in any frequency band. In particular, the lower the frequency band, the better the utility of the present invention.

  Next, an exemplary embodiment of the present invention will be described with reference to FIGS. In these examples, the configuration of the speaker array is illustrated as a linear speaker array, but is not limited to the configuration of the linear speaker array.

  In the embodiment shown in FIG. 12, M speakers 280-1,..., 280-M constituting the linear speaker array are fixed to a support member 400 having a rectangular flat plate shape. Are arranged on one plane (hereinafter referred to as an opening surface) of the support member 400 (M = 13 in the illustrated example). In addition, the wiring connected to each speaker 280-1, ..., 280-M is not illustrated. And the reflector 300 is being fixed to the edge part of the supporting member 400 so that the arrangement direction of each speaker 280-1, ..., 280-M may become the normal line of the reflector 300 of a rectangular flat plate shape. The opening surface of the support member 400 is a surface that forms 90 degrees with the reflector 300. 12, the preferable property of the reflector 300 is the same as the property of the reflector described above, and the property of the support member 400 is not particularly limited, and each speaker 280-1,. It is sufficient that the 280-M has sufficient rigidity to fix the 280-M.

  In the exemplary embodiment shown in FIG. 13A, the shaft portion 410 is fixed to the end portion of the support member 400, and the reflection plate 300 is rotatably attached to the shaft portion 410. According to this embodiment, it is possible to change the geometric arrangement of the reflector 300 with respect to the speaker array.

  In the implementation configuration example shown in FIG. 13B, two additional reflectors 310 and 320 are added to the implementation configuration example shown in FIG. The properties of the two added reflectors 310 and 320 may be the same as or different from those of the reflector 300. The properties of the reflector 310 may be the same as or different from those of the reflector 320. Hereinafter, the reflection plate 300 is referred to as a fixed reflection plate 300. The shaft 510 is fixed to the end of the fixed reflector 300 (the end opposite to the end of the fixed reflector 300 fixed to the support member 400), and the reflector 310 is rotated around the shaft 510. It is attached movably. Further, the shaft portion 520 is fixed to the end portion of the support member 400 (the end portion opposite to the end portion of the support member 400 to which the fixed reflection plate 300 is fixed), and the reflection plate 320 is attached to the shaft portion 520. It is pivotally attached. Hereinafter, the reflectors 310 and 320 are referred to as the movable reflectors 310 and 320. According to the embodiment shown in FIG. 13B, for example, when the position of the movable reflector 310 is set so that the reflection surface of the fixed reflector 300 and the reflection surface of the movable reflector 310 coincide with each other, the fixed reflector 300 and the movable reflector 300 are movable. The combination of the reflecting plates 310 can be made to function as a reflecting plate having a reflecting surface larger than that of the fixed reflecting plate 300. 13B, by setting the movable reflectors 310 and 320 to appropriate positions, for example, as shown in FIG. 14, the support member 400, the fixed reflector 300, and the movable reflector Since the sound can be reflected many times in the space surrounded by 310 and 320, the number of reflected sounds can be controlled. In the case of the embodiment shown in FIG. 13B, since the support member 400 plays a role as a reflector, it preferably has the same properties as those of the reflector described above.

  The embodiment configuration example shown in FIG. 15 is different from the embodiment configuration example shown in FIG. 12 in that the reflector 300 is also provided with a speaker array (a linear speaker array in the illustrated example). In the exemplary embodiment shown in FIG. 15, the arrangement direction of the M speakers fixed to the support member 400 and the arrangement direction of the M ′ speakers fixed to the reflector 300 are on the same plane. The arrangement is not limited (M ′ = 13 in the illustrated example). For example, M ′ speakers may be fixed to the reflection plate 300 so as to have an arrangement direction orthogonal to the arrangement direction of M speakers fixed to the support member 400. 15, the speaker array provided on the support member 400 and the reflector 300 (the speaker array provided on the reflector 300 is not used and the reflector 300 is used as a reflector). The present invention is implemented in combination, or a combination of the support member 400 (the speaker member provided on the support member 400 is not used and the support member 400 is used as a reflector) and the speaker array provided on the reflector 300 The present invention can be implemented.

  Further, as an example of an extended implementation configuration of the implementation configuration example shown in FIG. 15, similarly to the implementation configuration example shown in FIG. 13B, a configuration in which two reflectors 310 and 320 are further added to the implementation configuration example shown in FIG. 15. It is good also as (refer FIG. 16). Although not shown, a speaker array may be provided on at least one of the movable reflectors 310 and 320. The loudspeaker holes of each speaker constituting the speaker array provided in the movable reflector 310 are arranged on the plane (opening surface) of the movable reflector 310 that can be opposed to the opening surface of the support member 400, for example. The loudspeaker holes of each speaker constituting the speaker array provided in the movable reflector 320 are arranged on the plane (opening surface) of the movable reflector 320 that can form the same plane as the opening surface of the support member 400, for example. Even such an example of the configuration can be used in the same manner as the configuration example shown in FIG. Further, according to this embodiment, for example, when the position of the movable reflecting plate 320 is set so that the opening surface of the supporting member 400 and the opening surface of the movable reflecting plate 320 coincide with each other, the combination of the supporting member 400 and the movable reflecting plate 320 is changed. The speaker array can be made to function larger than the speaker array provided on the support member 400. Also in the embodiment configuration example shown in FIG. 16, in the embodiment configuration example in which at least one of the movable reflectors 310 and 320 is provided with a speaker array, the same usage pattern as the embodiment configuration example shown in FIG. 14 is possible. Also in the example of the configuration shown in FIG. 16, in the example of the configuration in which at least one of the movable reflectors 310 and 320 is provided with a speaker array, for example, the movable reflectors 310 and 320 are used as ordinary reflectors. A use form in which the speaker array provided on the support member 400 and the speaker array provided on the fixed reflector 300 are used as an integral speaker array is also possible. In this case, this is equivalent to an example of an embodiment in which a speaker array composed of (M + M ′) speakers and two reflectors are used.

  When the movable reflector 310 is provided with a speaker array, the loudspeaker holes of the speakers constituting the speaker array provided in the movable reflector 310 are opposite to the plane of the movable reflector 310 that can face the opening surface of the support member 400. A speaker array may be provided on the movable reflector 310 so as to be arranged on a flat surface (opening surface). Further, when a speaker array is provided on the movable reflector 320, the loudspeaker holes of the speakers constituting the speaker array provided on the movable reflector 320 can be formed on the same plane as the opening surface of the support member 400. A speaker array may be provided on the movable reflector 320 so as to be disposed on a plane (opening surface) opposite to the plane. Of course, at least one of the movable reflectors 310 and 320 may be provided with a speaker array on the movable reflector so as to have openings on both sides thereof.

[A] When the speaker array is provided on at least one of the movable reflectors 310 and 320, and the opening surface of the movable reflector 310 is a plane that can face the opening surface of the support member 400, or the movable reflector. When the aperture surface of 320 is a plane that can form the same plane as the aperture surface of the support member 400, in the usage mode shown in FIG. 14, the aperture surface of the movable reflector 310 and / or the movable reflector 320 with respect to the viewing direction. Although the apparent array size in the line-of-sight direction is reduced by disposing the movable reflector 310 and / or the movable reflector 320 so as not to be visible, the movable reflector 310 and / or the movable reflector 320 is provided. By using the speaker array, the same effect as when the array size is increased can be obtained.

[B] When the speaker array is provided on at least one of the movable reflectors 310 and 320, the opening surface of the movable reflector 310 is a plane opposite to the plane that can face the opening surface of the support member 400. If the opening surface of the movable reflecting plate 320 is a plane opposite to the plane that can form the same plane as the opening surface of the support member 400, in the usage mode shown in FIG. The same effect as when the array size is increased can be obtained while maintaining the size.

  When at least one of the movable reflectors 310 and 320 is provided with a speaker array on the movable reflector so that both sides thereof have an opening surface, the effects of both [A] and [B] can be obtained. Is possible.

<Application example>
Examples of services in which the narrow-directional sound reproduction technology of the present invention is useful will be described below.

  As a first example, there is an audio reproduction by digital signage. According to the present invention, since voice can be provided only in a narrow range in a specific direction as compared with the prior art, an advertisement can be transmitted only to people in the range without causing trouble to the surroundings.

  A second example is application to a TV conference system (which may be an audio conference system). According to the present invention, audio can be provided only in a narrow range in a specific direction as compared with the conventional case when a conference is performed in a situation where a room for exclusive use of a video conference cannot be prepared. It can be carried out.

<Hardware configuration example of narrow-directional sound reproduction processing device>
The narrow-directional sound reproduction processing apparatus according to the above-described embodiment may include an input unit to which a keyboard or the like can be connected, an output unit to which a liquid crystal display or the like can be connected, a CPU (Central Processing Unit) [cache memory, or the like. ] RAM (Random Access Memory) or ROM (Read Only Memory) and external storage device as a hard disk, and data exchange between these input unit, output unit, CPU, RAM, ROM, and external storage device It has a bus that can be connected. If necessary, the narrow-directional sound reproduction processing device may be provided with a device (drive) that can read and write a storage medium such as a CD-ROM. A physical entity having such hardware resources includes a general-purpose computer.

  The external storage device of the narrow-directional sound reproduction processing device stores a program for reproducing sound in a narrow range including the target direction, data necessary for processing of this program, etc. [in the external storage device] For example, the program may be stored in a ROM that is a read-only storage device. ]. Data obtained by the processing of these programs is appropriately stored in a RAM or an external storage device. Hereinafter, a storage device that stores data, addresses of storage areas, and the like is simply referred to as a “storage unit”.

  A program for obtaining a filter for each frequency by using the spatial correlation matrix expressed by the equations (5a) to (5b) for the direction to be reproduced, in the storage unit of the narrow-directional sound reproduction processing device. , A program for performing AD conversion on an analog signal, a program for performing frame generation processing, a program for converting a digital signal for each frame into a frequency domain signal in a frequency domain, and in a direction to be a target of audio reproduction A program for obtaining a reproduction signal by applying a corresponding filter to a frequency domain signal for each frequency and a program for converting the reproduction signal into a time domain signal are stored.

  In the narrow-directional sound reproduction processing apparatus, each program stored in the storage unit and data necessary for processing each program are read into the RAM as necessary, and are interpreted and executed by the CPU. As a result, narrow-directional sound reproduction is realized by the CPU realizing predetermined functions (filter design unit, AD conversion unit, frame generation unit, frequency domain conversion unit, filter application unit, time domain conversion unit).

<Supplementary note>
The present invention is not limited to the above-described embodiment, and can be appropriately changed without departing from the spirit of the present invention. For example, in the above-described embodiment, it is assumed that the sound wave travels as a plane wave. However, the sound wave may travel as a spherical wave. In this case, the steering vector is changed to an expression corresponding to the spherical wave. In addition, the processing described in the above embodiment may be executed not only in time series according to the order of description but also in parallel or individually as required by the processing capability of the apparatus that executes the processing. .

  Further, when the processing functions in the hardware entity (narrow-directed sound reproduction processing device) described in the above embodiment are realized by a computer, the processing contents of the functions that the hardware entity should have are described by a program. Then, by executing this program on a computer, the processing functions in the hardware entity are realized on the computer.

  The program describing the processing contents can be recorded on a computer-readable recording medium. As the computer-readable recording medium, for example, any recording medium such as a magnetic recording device, an optical disk, a magneto-optical recording medium, and a semiconductor memory may be used. Specifically, for example, as a magnetic recording device, a hard disk device, a flexible disk, a magnetic tape or the like, and as an optical disk, a DVD (Digital Versatile Disc), a DVD-RAM (Random Access Memory), a CD-ROM (Compact Disc Read Only). Memory), CD-R (Recordable) / RW (ReWritable), etc., magneto-optical recording medium, MO (Magneto-Optical disc), etc., semiconductor memory, EEP-ROM (Electronically Erasable and Programmable-Read Only Memory), etc. Can be used.

  The program is distributed by selling, transferring, or lending a portable recording medium such as a DVD or CD-ROM in which the program is recorded. Furthermore, the program may be distributed by storing the program in a storage device of the server computer and transferring the program from the server computer to another computer via a network.

  A computer that executes such a program first stores, for example, a program recorded on a portable recording medium or a program transferred from a server computer in its own storage device. When executing the process, the computer reads a program stored in its own recording medium and executes a process according to the read program. As another execution form of the program, the computer may directly read the program from a portable recording medium and execute processing according to the program, and the program is transferred from the server computer to the computer. Each time, the processing according to the received program may be executed sequentially. Also, the program is not transferred from the server computer to the computer, and the above-described processing is executed by a so-called ASP (Application Service Provider) type service that realizes the processing function only by the execution instruction and result acquisition. It is good. Note that the program in this embodiment includes information that is used for processing by an electronic computer and that conforms to the program (data that is not a direct command to the computer but has a property that defines the processing of the computer).

  In this embodiment, a hardware entity is configured by executing a predetermined program on a computer. However, at least a part of these processing contents may be realized by hardware.

Claims (14)

M is an integer of 2 or more, and an M channel time domain signal x reproduced by a speaker array composed of M speakers is converted into a frequency domain signal S obtained by converting a sound source signal into a frequency domain. A narrow-directional sound reproduction processing method obtained by converting the frequency domain signal X of the M channel obtained by applying a filter that converts S into the frequency domain signal X of the M channel for each frequency into the time domain,
Using the audio transmission characteristic a _φ from each speaker for each direction φ included in one or a plurality of directions assumed as the audio traveling direction, the above-mentioned for each frequency with respect to the direction to be reproduced. A filter design step for obtaining a filter;
Applying the filter obtained in the filter design step to the frequency domain signal S for each frequency to obtain the M channel frequency domain signal X, and
(1) The sound radiated from the speaker array, (2) the sound reflected by the reflector, and the reflected sound traveling in the direction φ is a dual sound, and each of the transfer characteristics a A narrow-directional sound reproduction processing method characterized in that _φ is represented by the sum of direct sound transfer characteristics in the direction φ and one or more dual sound transfer characteristics. The narrow-directional sound reproduction processing method according to claim 1,
Each of the transfer characteristics a _φ is the sum of the steering vector of the direct sound and one or more steering vectors of the one or more dual sounds in which the sound attenuation due to reflection and the time difference of the reflected sound with respect to the direct sound are corrected. A narrow-directional sound reproduction processing method characterized in that: The narrow-directional sound reproduction processing method according to claim 1,
Each of the transfer characteristics _aφ is obtained by actual measurement in an actual environment. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
A narrow-directional sound reproduction processing method characterized in that, in the filter design step, the filter is obtained for each frequency so that the power of sound in a direction other than the direction to be reproduced is minimized. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
The narrow-directional sound reproduction processing method according to claim 1, wherein, in the filter design step, the filter is obtained for each of the frequencies so that an SN ratio in the direction to be reproduced is maximized. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
In the filter design step, the power of the voice in the one or more directions assumed as the voice traveling direction with the filter coefficient for one of the M speakers fixed to a constant value is minimized. As described above, the narrow-directional sound reproduction processing method, wherein the filter is obtained for each frequency. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
In the filter design step, the target of audio reproduction under the conditions of (1) the entire band of audio in the above direction to be reproduced, and (2) the suppression of the entire band of one or more blind spots. The narrow-directional sound reproduction processing method, wherein the filter is obtained for each frequency so that the power of the sound in a direction other than the direction and the blind spot is minimized. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
In the filter design step, by normalizing the transfer characteristics a _s the direction phi = s to be audio playback, narrow directional audio reproduction processing method characterized by the filter is required for each of the frequencies. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
In the filter design step, the filter is obtained for each frequency by using a spatial correlation matrix represented by the transfer characteristic a _φ corresponding to each direction other than the direction to be reproduced. Narrow-directional sound reproduction processing method. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
In the filter design step, the power of the sound in the direction other than the direction to be reproduced is minimized under the condition that the deterioration amount of the sound in the direction to be reproduced is a predetermined amount or less. Thus, the narrow-directional sound reproduction processing method characterized in that the filter is required for each frequency. The narrow-directional sound reproduction processing method according to any one of claims 1 to 3,
In the filter design step, the filter is obtained for each frequency by using a spatial correlation matrix represented by a frequency domain signal obtained by converting a signal obtained by observing with a microphone array into a frequency domain. A narrow-directional audio reproduction processing method as a feature. M is an integer of 2 or more, and an M channel time domain signal x reproduced by a speaker array composed of M speakers is converted into a frequency domain signal S obtained by converting a sound source signal into a frequency domain. A narrow-directional sound reproduction processing apparatus obtained by converting the frequency domain signal X of the M channel obtained by applying a filter for converting S into the frequency domain signal X of the M channel for each frequency into the time domain,
Using the audio transmission characteristic a _φ from each speaker for each direction φ included in one or a plurality of directions assumed as the audio traveling direction, the above-mentioned for each frequency with respect to the direction to be reproduced. A filter design section for obtaining a filter;
A filter application unit that obtains the M channel frequency domain signal X by applying the filter obtained by the filter design unit to the frequency domain signal S for each frequency;
(1) The sound radiated from the speaker array, (2) the sound reflected by the reflector and the reflected sound traveling in the direction φ is a dual sound, and each of the transfer characteristics a A narrow-directional sound reproduction processing apparatus characterized in that _φ is represented by the sum of the direct sound transfer characteristics in the direction φ and the transfer characteristics of one or more dual sounds. The narrow-directional sound reproduction processing device according to claim 12,
A narrow-directional sound reproduction processing apparatus, further comprising one or more reflectors that give each of the reflected sounds to the dual sound.       The program for making a computer perform the process of the narrow directivity audio | voice reproduction | regeneration processing method in any one of Claims 1-11.

Images