JP2010109579A - Sound output element array and sound output method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To achieve desired directional characteristics in a sound output element array by simple work. <P>SOLUTION: The sound output element array includes: the sound output element array arranged on the same axis; a first digital filter group having the same number of digital filters as the respective sound output elements to which reproducing sound signals are input; and a second digital filter group having the same number of digital filters to which interference sound signals are input. When respective sound signals emitted from the respective sound output elements are convoluted and mixed by respective approximate transmission functions approximated such that a part from the respective sound output elements to respective control points is a free space, to the first digital filter group, a filter coefficient for turning the convoluted and mixed signals to non-zero at the control point of a sound reproducing area set in an axial direction in which the respective sound output elements are arranged and to zero at the control point of a sound interruption area excluding the sound reproducing area is set. To the second digital filter group, a filter coefficient for turning them to zero at the control point of the sound reproducing area and to non-zero at the control point of the sound interruption area excluding the sound reproducing area is set. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to an acoustic output element array that transmits signals from a plurality of acoustic output elements and an acoustic output method thereof, and more particularly, to a technique for providing a signal with reduced directivity in a specific direction.

  In recent years, communication conferences connecting remote locations using telephone lines, ADSL lines, optical networks, and the like have been frequently performed. Such communication conferences are generally held in conference rooms that are acoustically closed. However, communication conferences can be easily held in one corner of an open office (a large room with disks). It is desired. In communication conferences in such places, people who are not participating in the conference do not want to hear the loud sound, so a technology that can reproduce sound only in a specific area or in a specific direction is desired. .

  A directional speaker is known as an acoustic output element that radiates a sound for realizing this only in a specific direction. As the directional speaker, a horn speaker, a parametric speaker, an analog speaker array, a digital filter type speaker array, and the like are conventionally known.

The horn loudspeaker (Mitaro Goro, “Acoustic Engineering”, p. 101, Shosodo, 1993) achieves directivity by the geometric shape of the horn, and is usually limited to high frequencies. In order to obtain sharp directivity in a wide frequency range, a horn having a large size exceeding 1 m is required, and it cannot be used for general purposes. The parametric speaker (Patent Document 1) is a method for obtaining a sharp directivity for an audible sound by using an ultrasonic wave having a sharp directivity as a carrier of an audible sound (carrier wave). This is a method in which the original audio signal is demodulated by the nonlinearity of the modulated ultrasonic wave in the air. This method is known to obtain a sharp directivity, but there are problems such as requiring a special amplifier for emitting ultrasonic waves and suppressing the influence of ultrasonic waves on the human body.

  On the other hand, a speaker array using digital filter signal processing has recently been proposed (Patent Document 2). In this method, an independent digital filter is connected to each of a plurality of speakers arranged in a space, and the characteristics of these digital filters are determined based on sound signals obtained by setting observation signals at a plurality of control points arranged in the same space. It is characterized in that it is determined so that a desired plane wave is obtained in the reproduction direction and the observation signal is zero in a preset sound cutoff region. As a result, a directional speaker that reproduces sound only in a specific direction can be realized.

  FIG. 1 is a configuration diagram showing a configuration of a speaker array in which a plurality of speakers 12 shown in Patent Document 2 are arranged in a straight line. In this figure, a signal input terminal 10 is an input terminal for inputting a signal corresponding to an acoustic signal (sound) to be reproduced to a speaker array, and a digital filter 11 determines the directivity of the reproduced acoustic signal. The speaker 12 is an acoustic output element that emits an acoustic signal into space. Further, the emphasis control points 13a and 13b indicated by ● and the suppression control point 14 indicated by × surrounded by ○ are control points for observing the acoustic signal emitted to the space in order to determine the filter coefficient of the digital filter 11. It is. The sound reproduction area 15 is an area that emits an acoustic signal, and the listener 16 listens to the acoustic signal emitted from the speaker 12. When the signal input to the signal input terminal 10 is an analog signal, an A / D converter is required between the signal input terminal 10 and the digital filter 11, and D is connected between the digital filter 11 and the speaker 12. A / A converter and an amplifier are required, but these are omitted in FIG. Further, in this example, it is assumed that a linear speaker array is installed under a TV monitor or the like in a communication conference or the like, and an axis on which the speakers 12 constituting the speaker array are arranged (referred to as “speaker array axis”). It is assumed that there is a listener 16 in a direction perpendicular to).

In addition, the digital filter 11 is usually configured by the following FIR (Finite Impulse Response) filter.

ただし、ｚは周波数依存の複素変数である。このようにインパルス応答（フィルタ係数）の系列をｚ変換したものを伝達関数と呼ぶ。以下では、ωを周波数として式（２）の伝達関数をＨ_ｍ(ω)と表現する。 Here, L is the number of filter taps, m (m = 1,..., M, M is the total number of speakers 12) is an identifier for identifying each digital filter, and x _m (τ−k) is , An input signal at a discrete time τ-k to the digital filter with the identifier m, and h _m (k) is an impulse response (filter coefficient) of the digital filter with the identifier m. Further, when the impulse response (filter coefficient) series of the equation (1) is z-transformed, the following is obtained.

However, z is a frequency dependent complex variable. A series of impulse responses (filter coefficients) thus z-transformed is called a transfer function. In the following, the transfer function of Equation (2) is expressed as H _m (ω) with ω as the frequency.

In this example, in order to confirm that the wave is a plane wave, the emphasis control point 13a is arranged perpendicularly to the traveling direction of the acoustic signal, and further, in order to ensure that the advanced sound wave maintains the plane wave, Two emphasis control points 13b are arranged. Here, in each digital filter 11, the acoustic signal emitted from the acoustic output element array becomes 1 at all the emphasis control points 13a in the first column, and at all the emphasis control points 13b in the second column. The sound pressure of the acoustic signal whose phase is delayed from the acoustic signal of the first row by the time difference obtained by dividing the distance difference between the first row and the second row by the speed of sound becomes the sound pressure of the acoustic signal at the suppression control point 14. Set to zero. A filter characteristic satisfying such a condition can be described as follows.

Here, G _mn (ω) is a frequency obtained by z-transforming a series of impulse responses from the m-th speaker 12 to the n-th control point (n = 1,..., N, N is the total number of control points). It is a dependent (indoor) transfer function, and is set based on acoustic signals actually observed at each control point 13a, 13b, 14 using a microphone. Each element of the N-dimensional vector, which is the calculation result on the left side of Equation (3), is a response output from each FIR filter when an impulse signal is input to each FIR filter of the filter coefficient shown in Equation (2). A signal is emitted from each speaker 12 and is convolved with an impulse response from each speaker 12 to each control point, and shows a convolution mixed signal mixed at each control point. In this equation, control points from n = 1 to P-1 indicate suppression control points 14 for controlling the sound pressure to zero, and control points from n = P to Q-1 are emphasized in the first row. The control point 13a is shown, and the control points from n = Q to N indicate the emphasis control point 13b in the second row. Further, Δ is an emphasis control point 13a in the first row (control point from n = 1 to P-1) and an emphasis control point 13b in the second row (control point from n = P to Q-1). Is the delay time divided by the speed of sound. Further, exp indicates the number of Napiers, and j indicates an imaginary unit.

Now, assuming that the above equation (3) is G · H = R, the characteristics of the digital filter connected to the acoustic output element array can be obtained by the following equation. Α ^T represents a transposed matrix of α.
H = G ^T · (G ^T · G) ⁻¹ · R (4)
Also, as the inverse matrix does not become unstable when a small positive constant compared to the maximum eigenvalue of G T ^· G [delta], may be obtained H by substituting the following equation. Here, I is an M × M unit matrix.
^{H = G T · (G T} · G + δ · I) -1 · R ... (5)
With the configuration as described above, a plane wave with a spread larger than the size of the speaker array is realized.
JP 2003-153369 A JP 2005-236636 A

  When the listener 16 is not in the direction perpendicular to the axis of the speaker array but in the axial direction of the speaker array as shown in FIG. 2, that is, the sound reproduction region is in the axial direction of the speaker array. Consider the case of setting 15. In addition, the axis | shaft 21 shows the axis | shaft of a speaker array and has shown that the listener 16 exists in the axis | shaft 21 direction of a speaker array, and the sound reproduction area | region 15 is set to the direction. The sound blocking area 22 is an area where sound should be suppressed, and the directivity characteristics of the speaker array are determined by the sound pressure difference between the sound blocking area 22 and the sound reproduction area 15. The listener 16 is located in the direction of the axis 21 of the speaker array, for example, when the speaker array is configured on a desk with a narrow width. This is a case where a width can be taken in the depth direction.

At this time, when the front surface of each speaker 12 is oriented perpendicularly to the axis of the speaker array as shown in FIG. 2, a desired directivity characteristic is realized unless a control point is provided so as to surround each speaker 12 as a whole. The filter coefficient of the digital filter to be determined cannot be determined. This is because each speaker 12 itself has asymmetric directivity on the front and back surfaces, and it is necessary to control the directivity separately on the front and back sides of each speaker 12. In this case, a very large number of suppression control points must be arranged. For example, as shown in FIG. 2, if the suppression control points are arranged at intervals of 22.5 degrees from the direction of 45 degrees to the direction of 315 degrees from the axis of the speaker array, a total of 13 suppression control points must be arranged. I must. Further, considering the control up to the vertical direction, about 60 suppression control points must be arranged. Then, in order to determine the filter coefficient, it is necessary to perform complicated operations such as actually observing the acoustic signal at each of the set many control points and calculating the transfer function G _mn (ω) from the observation results.

  Such a problem is not a problem limited to a speaker array composed of a plurality of speakers, but is a problem common to an acoustic output element array composed of a plurality of acoustic output elements that output some acoustic signal.

  The present invention has been made in view of such a point, and an acoustic reproduction region in an axis direction (hereinafter referred to as an “axis of an acoustic output element array”) in which acoustic output elements constituting the acoustic output element array are arranged. An object of the present invention is to provide a technique capable of realizing a desired directivity with a simple operation. It is another object of the present invention to realize an acoustic output element array in which guidance is given only to a specific individual in a scene where a plurality of people are present, and an operation can be explained only to that individual.

  In the present invention, in order to solve the above-described problem, a plurality of acoustic output elements arranged on the same axis and arranged in directions in which each directivity characteristic is axially symmetric with respect to the axis, and a reproduced sound signal are input. A first digital filter group having the same number of digital filters as the acoustic output elements, a second digital filter group having the same number of digital filters as the acoustic output elements to which the interference signal is input, a first digital filter group and a second digital filter; An adder group for adding the output signals of the respective filters of the filter group, and each supply generated by adding the respective output signals corresponding to the plurality of acoustic output elements of the first digital filter group and the second digital filter group A signal supply unit that supplies a signal to a corresponding acoustic output element, and each acoustic output element spatially transmits an acoustic signal corresponding to each supplied supply signal. The first digital filter group emits each acoustic signal emitted from each acoustic output element to each control point from each acoustic output element when approximating that each acoustic output element to each control point is a free space. When convolved with each approximate transfer function up to and mixed at each control point, the convolution mixed signal becomes non-zero at the control point in the sound reproduction region set in the axial direction where the sound output elements are arranged. , A first digital filter group in which a filter coefficient that is zero is set at a control point in the sound blocking area excluding the sound reproduction area, and the second digital filter group receives each acoustic signal emitted from each acoustic output element. When approximating that each acoustic output element to each control point is free space, convolution with each approximate transfer function from each acoustic output element to each control point and mixing at each control point. In this case, the convolution mixed signal becomes zero at the control point of the sound reproduction region set in the axial direction in which the respective sound output elements are arranged, and becomes zero at the control point of the sound blocking region excluding the sound reproduction region. A second digital filter group in which filter coefficients are set, and each approximate transfer function is a function calculated using each distance from each acoustic output element to each control point. An output element array is provided. The free space means an ideal anechoic space (nonreflective space).

  In the present invention, when the sound reproduction area is set in the axial direction of the sound output element array, desired directivity can be realized with a simple operation. In particular, since the disturbing sound is generated for the sound blocking area, it is possible to give guidance or the like to a specific individual located in the sound reproduction area.

Hereinafter, embodiments of the present invention will be described with reference to the drawings.
〔principle〕
First, the principle of this embodiment will be described.
In this embodiment, instead of outputting an acoustic signal in a direction perpendicular to the axis of the acoustic output element array, there is a listener in the axial direction of the acoustic output element array, and the acoustic signal is transmitted in the axial direction of the acoustic output element array. It outputs and suppresses the acoustic signal in the other direction. That is, the acoustic output element array of the present embodiment emits an acoustic signal to the acoustic reproduction area set on the listener side in the axial direction of the acoustic output element array, and suppresses the acoustic signal in the acoustic blocking area excluding the acoustic reproduction area. To do.

  Therefore, in this embodiment, a plurality of sound output elements arranged on the same axis and arranged in directions in which the respective directivity characteristics are axially symmetric with respect to the axis, and independent from each other corresponding to each sound output element, respectively. A signal supply unit that includes a digital filter and supplies each supply signal generated using each digital filter to a corresponding acoustic output element, and each acoustic output element is supplied with each supplied signal Each sound output when the digital filter approximates each sound signal emitted from each sound output element to a free space from each sound output element to each control point. When convolution is performed with each approximate transfer function from the element to each control point and mixing is performed at each control point, the convolution mixed signal is set in the axial direction in which each acoustic output element is arranged. A digital filter in which a filter coefficient is set to be zero at a control point in the sound reproduction region and zero at a control point in the sound blocking region excluding the sound reproduction region, and each approximate transfer function is An acoustic output element array that is a function calculated using each distance to the control point is used.

  Here, a plurality of sound output elements (for example, speakers) constituting the sound output element array of the present embodiment are arranged on the same axis, and are arranged in directions in which each directional characteristic is axially symmetric with respect to the axis. . As a result, the directivity of each acoustic output element itself is axisymmetric with respect to the axis of the acoustic output element array. Therefore, if the transfer function of the space is also axially symmetric with respect to the axis, it is expected that directivity characteristics in all directions can be controlled without providing control points that are axially symmetric with respect to each other. However, in an actual environment, even if the acoustic output element is arranged as described above, the transfer function is often not axisymmetric with respect to the axis due to the influence of reflection. Therefore, a microphone is installed only at one of the control points that are axisymmetric to each other, a transfer function is obtained based on the observation result of the control point at which the microphone is installed, and the transfer function is determined by the control at which the microphone is installed. Even as a transfer function corresponding to a control point that is axisymmetric with respect to a point, an appropriate filter coefficient suitable for the actual environment cannot be set. Therefore, when setting the transfer function using the result measured with the microphone arranged at the control point, even if each acoustic output element is arranged as described above, many suppression control points are provided, and microphones are attached to them. Must be installed.

  Therefore, in this embodiment, the acoustic signal is not actually measured by the microphone and the transfer function is set, but the space from each acoustic output element to each control point is free space (non-reflection state, ideal anechoic space). The approximate function (approximate transfer function) of each transfer function is calculated from only the distance from each acoustic output element to each control point. An approximate transfer function based on such approximation can be easily calculated using each distance from each acoustic output element to each control point without actually observing the acoustic signal at the observation point. Furthermore, a digital filter with filter coefficients set based on an approximate transfer function in this way uses a transfer function calculated from an actual observation result at a certain control point as a transfer function of an observation point that is axisymmetric with that observation point. It is more suitable for the actual environment than a digital filter with a set filter coefficient. This is because the influence of reflection is not axisymmetric, but rather an appropriate solution can be obtained if the influence is ignored.

For example, the following equation can be used as such an approximate transfer function.
G _mn (ω) = (α / r _mn ) ・ exp (‐j ・ ω ・ r _mn / c) (6)
Here, ω represents the frequency [Hz], and G _mn (ω) is the nth (n = 1) from the mth (m = 1,..., M is an integer of 2 or more) acoustic output elements. ,..., N, N are approximate transfer functions up to a control point of an integer of 1 or more. R _mn represents the distance [m] from the mth acoustic output element to the nth control point, c represents the speed of sound [m / s], j represents the imaginary unit, and exp represents the Napier number. , Α ≠ 0 indicates a constant. In this embodiment, since it is not necessary to observe a signal at each control point, it is not necessary to actually arrange each control point in space. Each control point may be virtually arranged in space so that each distance r _mn is determined.

  When each acoustic output element is arranged as described above and each approximate transfer function from each acoustic output element to each control point is set according to the equation (6), the same acoustic output element to the axis of the acoustic output element array is set. Thus, the approximate transfer functions up to two control points arranged in an axially symmetrical manner are equal to each other. For this reason, each control point is assumed to be virtual only on one axis of the set of points that are axially symmetric with respect to the axis of the acoustic output element array (on the axis where the acoustic output elements are arranged) and / or the axis. The filter coefficient for constructing the acoustic output element array having the desired directivity can be calculated simply by setting it.

When the acoustic output element is a dual sound source element that outputs the first acoustic signal from the first part and outputs the second acoustic signal whose phase is inverted from the first acoustic signal from the second part, For example, the following equation can be used as an approximate transfer function.

Here, r _Fmn represents the distance [m] from the first part of the mth acoustic output element to the nth control point, and r _Rmn represents the nth control point from the second part of the mth acoustic output element. Distance [m] is shown. Α _F and α _R are positive constants. Note that the sign of the two items in Expression (7) is negative because the phase of the first acoustic signal and the phase of the second acoustic signal are inverted. As a specific example of such a dual sound source element, for example, there is a speaker that is not contained in an enclosure (speaker box). In this case, for example, the central part of the acoustic signal emission surface (front surface) of the speaker is the first part, and the central part of the back surface is the second part. Normally, the state in which the speaker diaphragm is pushed to the front side is the state in which the diaphragm is pushed to the front side when viewed from the back side, so that an acoustic signal emitted from the front side of such a speaker And the phase of the acoustic signal emitted from the back surface are mutually inverted.

  Then, each approximate transfer function is calculated using each distance from each acoustic output element set as described above to each control point, and each acoustic signal output from each acoustic output element is calculated from each acoustic output element. When convolved with each approximate transfer function up to each control point and mixed at each control point, the convolution mixed signal is the control point of the sound reproduction region set in the axial direction where each sound output element is arranged. By setting the filter coefficient so that it becomes zero and becomes zero at the control point of the sound blocking area excluding the sound reproduction area, an acoustic output element array having desired directivity characteristics can be realized.

  As described above, in this embodiment, the axis of the acoustic output element array is arranged in the direction in which the acoustic signal is desired to be radiated, and the directional characteristics of each acoustic output element are axisymmetric with respect to the axis. By placing control points on the fly, there is no need to actually place a microphone at the control point and use the observation results with the microphone to set the transfer function between the acoustic output element and the control point. It is possible to easily obtain a filter coefficient that realizes a desired directivity characteristic. That is, the filter coefficient can be determined by the placement of virtual control points without actually placing the control points in the space, thereby freeing from troublesome work such as microphone installation.

[First Embodiment]
Next, a first embodiment of the present invention will be described. This embodiment is an embodiment in which the present invention is applied to a directional speaker.

<Arrangement configuration>
FIG. 3 is a diagram for explaining the arrangement of the sound output elements and control points constituting the sound output element array 100 of the first embodiment, and FIG. 4 is the structure of the sound output element array 100 of the first embodiment. It is a block diagram for demonstrating.

  As shown in FIG. 3, in this embodiment, a listener 16 is present in the direction of the axis 21 of the acoustic output element array 100 instead of outputting an acoustic signal in a direction perpendicular to the axis 21 of the acoustic output element array 100. An acoustic signal is output in the direction of the axis 21 of the acoustic output element array 100, and the acoustic signal is suppressed in other directions. That is, the acoustic output element array 100 of the present embodiment emits an acoustic signal to the acoustic reproduction region 15 set on the listener 16 side in the direction of the axis 21, and transmits the acoustic signal in the acoustic blocking region 22 excluding the acoustic reproduction region 15. Repress.

Therefore, in this embodiment, a plurality of (four in the examples of FIGS. 3 and 4) acoustic output elements 1 on the shaft 21.
The sound output elements 12 are arranged in such a direction that their directivity characteristics are axially symmetric with respect to the axis 21. In this embodiment, for example, a plurality of acoustic output elements 12 (for example, speakers) having directivity characteristics that are axisymmetric with respect to the acoustic signal emission axis are used, and the emission axes of these acoustic signals are respectively arranged on the shaft 21. Furthermore, the acoustic signal emitting surface of each acoustic output element 12 is arranged so as to face the listener 16 side (see FIG. 3). Further, the performances of the respective sound output elements 12 are the same, and these are arranged on the shaft 21 at the same interval or unequal intervals.

  In this embodiment, the sound reproduction area 15 is set on the listener 16 side in the direction of the axis 21 where the sound output elements are arranged, and the control point 13 (emphasis control point) is set on the axis 21 of the sound reproduction area 15. To do. Thereby, each acoustic output element 12 arranged as described above outputs at least a part of the acoustic signal corresponding to each supplied supply signal toward the sound reproduction region 15.

  Furthermore, in this embodiment, the sound blocking area 22 is set in an area excluding the sound reproduction area 15, and a plurality of control points 14 (suppression control points) are set in the sound blocking area 22. However, each control point 14 in the sound blocking region 22 is set only to one point of a set of points that are axially symmetric with respect to the axis 21 on which each acoustic output element 12 is arranged. In the example of FIG. 3, the intersection of the circumference of the circle passing through the control point 13 and the shaft 21 in the sound blocking region 22 with the center point of the acoustic output element 12 row arranged on the shaft 21 as the center, and the shaft Seven virtual control points 14 are arranged only on the circumference of the sound blocking region 22 on one side (left side in FIG. 3) with 21 as a boundary.

  Further, as shown in FIG. 4, a signal supply unit 110 that supplies a supply signal to the plurality of acoustic output elements 12 arranged as described above is electrically connected. The signal supply unit 110 includes an independent digital filter 111, an amplifier 112, a D / A converter 113, an A / D converter 114, and a signal input terminal 115 to which an analog signal is input, corresponding to each acoustic output element 12. And each supply signal generated using each digital filter 111 is supplied to a corresponding acoustic output element.

となる。このように求めた距離ｒ_ｍｎを式（６）に代入することで、各音響出力素子１２から各制御点１４までの近似伝達関数Ｇ_ｍｎ（ω）を決定することができる。 <Filter coefficient setting method>
Next, a method for setting the filter coefficient of each digital filter 111 will be described. Here, as illustrated in FIG. 4, each acoustic output element 12 is arranged on the axis 21 of the acoustic output element array 100 toward the listener 13 at intervals of s [m], and is arranged on the axis 21. The control points 13 (enhancement control points) of the sound reproduction region 15 and the control points 14 (suppression control points) of the sound blocking region 22 are arranged at a distance of L [m] from the center point of the acoustic output element 12 row. It shall be. Also, the acoustic output elements 12 in FIG. 4 correspond to m = 1,..., M (M is the total number of acoustic output elements 12, M = 4 in the example of FIG. 4) in order from the top. And the control point 13 is a control point corresponding to n = 1, and each control point 14 is sequentially n = 2,..., N (N is the total number of control points, N = 8 in the example of FIG. 4) ) Is a control point corresponding to each. In FIG. 4, an angle θ _n formed by a straight line connecting the center point of the acoustic output element 12 row and the nth control point and the axis 21 is defined with the front direction of each acoustic output element as 0 degree ( θ ₁ = 0). In this case, the distance from the mth acoustic output element 12 to the nth control point is

It becomes. The approximate transfer function G _mn (ω) from each acoustic output element 12 to each control point 14 can be determined by substituting the distance r _mn obtained in this way into the equation (6).

Next, filter coefficients H _m (ω) (m = 1,..., M) satisfying the following equation are obtained.

Here, the element in the top row of the equation (9) is an element corresponding to the control point 13 (emphasis control point), and the 2nd to Nth elements from the top are each control point 14 (suppression control point). Corresponding element. In other words, the equation (9) is obtained by converting each acoustic signal H _m (ω) output from each acoustic output element (m = 1,..., M) by each approximate transfer function G _mn (ω). When convolution is performed and mixing is performed at each control point, the convolution mixed signal becomes non-zero at the control point (n = 1) of the sound reproduction region 15 set in the axial direction where the sound output elements 12 are arranged. , The control points (n = 2,..., M) of the sound blocking area 22 excluding the sound reproduction area 15 are zero. Each acoustic signal H _m (ω) means z conversion of an output signal sequence when the same impulse is input to each digital filter 111. Therefore, if H _m (ω) satisfying such a relationship is used as the filter coefficient of the digital filter 11 corresponding to the mth acoustic output element 12, the acoustic signal is reproduced at the control point 13 (emphasis control point). A directional characteristic in which the acoustic signal is suppressed at the control point 14 (suppression control point) can be realized.

In addition, as described above, each sound output element to each control point is approximated as a free space, and G _mn (ω) can be simplified. As a result, even if the number of N is larger or smaller than M, to some extent Appropriate directivity can be achieved. However, in order to optimize directivity, it is desirable that N + 1 ≦ M. L is preferably larger than the wavelength of the lower limit frequency of the frequency band to be reproduced, but may be half or more. Further, although the interval between the angles θ _n (difference between θ _n and θ _n−1 ) depends on L, it may be between 10 and 45 degrees.

Moreover, the structure which the filter coefficient from which the value of a convolution mixing signal changes according to a frequency may be set in the control point 13 of at least one part of the sound reproduction area | region 15 may be sufficient. In this case, a filter coefficient H _m (ω) (m = 1,..., M) that satisfies the following equation is obtained instead of the equation (9).
Note that D (ω) is a function that takes different values depending on the frequency ω.

Now, if the above equation (9) or (10) is set to G · H = R, each filter coefficient H _m (ω) of the digital filter 11 connected to the acoustic output element array 100 can be obtained by the following equation. . Α ^T represents a transposed matrix of α.
H = G ^T · (G ^T · G) ⁻¹ · R (11)
Also, as the inverse matrix does not become unstable when a small positive constant compared to the maximum eigenvalue of G T ^· G [delta], seeking a substitute to the filter coefficient H _{m (ω)} by the following formula Also good. Here, I is an M × M unit matrix.
^{H = G T · (G T} · G + δ · I) -1 · R ... (12)
Expressions (9) and (10) are merely examples, and the number of emphasis control points may be two or more, and elements other than expressions (9) and (10) may be used as elements on the right side corresponding to the emphasis control points of these expressions. Values and functions may be used.

<Sound signal output processing>
Next, acoustic signal output processing of the acoustic output element array 100 will be described.
First, an analog signal corresponding to reproduced sound is input from the signal input terminal 115. The input analog signal is converted into a digital signal by each A / D converter 114 and input to each digital filter 111 in which the filter coefficient is set as described above. Each digital filter 111 converts each input digital signal into each supply signal for realizing a desired directivity characteristic, and outputs them to each D / A converter 113. Each D / A converter 113 converts them into analog signals and inputs them to each amplifier 121, and each amplifier 121 amplifies them and supplies them to each acoustic output element 12. Each acoustic output element 12 outputs an acoustic signal corresponding to the input signal.

<Measurement results>
Next, the measurement result of the directional characteristics of the acoustic output element array 100 of the present embodiment configured as described above will be exemplified.
FIG. 5 is a graph illustrating the measurement result of the directivity of the acoustic output element array 100 of this embodiment. Here, the upper direction is the direction of the listener 13, the angle is 0 degree, the counterclockwise direction is 90 degrees and 180 degrees, and the point is 50 cm away from the center of the acoustic output element array 100 every 22.5 degrees. Sound pressure was measured. Also, here, the interval s between the acoustic output elements 12 is 0.048 m (4.8 cm), L is 0.5 m (50 cm), and the virtual control point 13 (emphasis control point) is at an angle θ ₁ = 0. The virtual control points 14 (suppression control points) are at angles θ ₂ = 45 °, θ ₃ = 67.5 °, θ ₄ = 90 °, θ ₅ = 112.5 °, Assume that θ ₆ = 135 °, θ ₇ = 157.5 °, and θ ₈ = 180 °. Also, the sound velocity c is assumed that the room temperature T is 24 degrees Celsius,
c = 331.5 + 0.61T (13)
Calculated by Then, by substituting these values into equation (8), each distance r _mn is calculated, approximate transfer function G _mn (ω) is calculated by equation (6), and H satisfying the equation of equation (9) _m (ω) was calculated as the filter coefficient of the digital filter 11 corresponding to the mth acoustic output element 12.

  Further, in FIG. 5, the center of the broken line extending concentrically indicates the lowest sound pressure value, and the sound pressure value increases as it spreads outward. The interval between the broken lines indicates a sound pressure difference of 10 dB. 5 indicates the sound pressure value measured at a point 50 cm away from the center of the acoustic output element array 100 as described above. As shown in FIG. 5, the suppression effect is higher by 15 dB or more on the side surface (90 degrees, 270 degrees) and rearward (180 degrees) compared to the front (listener 13 direction, 0 degrees), and as a result, the sound is heard only by the listener 13. It can be seen that the acoustic output element array 100 having the directivity characteristic that the sound reaches the other direction and the sound is difficult to hear in other directions is formed.

  However, when the suppression effect of diagonally backward (135 degrees, 225 degrees) and backward (180 degrees) is compared, the amount of backward suppression is small. The sound pressure in the rear is about 7 dB larger than the oblique rear, and there is a sound leak in the rear. Therefore, a second embodiment in which the rear suppression amount is increased will be described next.

[Second Embodiment]
In the second embodiment, as shown in the equation (14), weight coefficients (a ₁ , a ₂ ,..., A _N ) corresponding to the number N of control points are multiplied when obtaining the filter coefficients. It is.

In Equation (14), assuming that the acoustic output elements are arranged in order from the top of the matrix row and correspond to the direction from the front (0 degrees) to the rear (180 degrees), the sound leakage to the rear is reduced. to desirably weighting factor is _{a 1 <a 2 <... <} a n-1. The filter coefficient H that minimizes the error by the least square method is obtained by the expression (15) by setting the expression (14) as AGH = AR.
^{H = (G T · A T} AG) -1 G T · A T A · R ... (15)

In Equation (14), the last row is assumed to be an equation for controlling the 180-degree control point of the acoustic output element array 100. Then, for example, a _n = 1 (from n = 1 to N−1), a _n = 3 (n = N), and the number of acoustic output elements 12 is M = 4 to obtain filter coefficients. That is, the weighting coefficient corresponding to the control point on the relevant axis in the sound blocking region where the acoustic output elements 12 are arranged on the same axis is set to the maximum.

FIG. 6 shows directivity characteristics obtained by measuring the filter coefficients H _{1 to} H ₄ obtained by multiplying the weight coefficients as the filter coefficients of the digital filter 111 of the first embodiment. FIG. 6 is basically the same diagram as FIG. 5 illustrating the experimental results of the directivity characteristics of the acoustic output element array 100 described above, but the measurement points of the sound pressure are different. In FIG. 6, the sound pressure at a distance of 1.0 m from the center of the acoustic output element array 100 is measured every 15 degrees in the counterclockwise direction.

The characteristic indicated by the alternate long and short dash line in FIG. 6 is the characteristic of one acoustic output element (single speaker). A characteristic indicated by a broken line is a directivity characteristic of the acoustic output element array 100 that does not use a weighting factor. With respect to these characteristics, the directivity characteristics when the weight coefficient a _N = 3 is shown by a solid line. The directivity characteristic in the sound reproduction region centered at 0 degrees (for example, a range of 0 ± 30 degrees) does not change, but the sound pressure in the rear (180 degrees) is suppressed by about 4 dB.

FIG. 7 shows the directivity characteristics when the weight coefficient a _N = 8. Weighting coefficient _{a n} from n = 1 at this time to N-1 is _a n = 1. The sound pressure at the rear (180 degrees) is further suppressed by about 2 dB, but the sound pressure at the oblique rear (120 degrees, 240 degrees) is increased. Therefore, for the purpose of not increasing the sound pressure behind the diagonal, the weighting coefficients are a ₁ = 1, a ₂ = 1, a ₃ = 2, a ₄ = 4, a ₅ = 6, a ₆ = 8, a ₇ = FIG. 8 shows the directivity characteristics when 10, a ₈ = 12. The sound pressure behind the diagonal can be made smaller than the directivity (dashed line) when the weighting factor is not used. However, the sound pressure in the diagonally forward (60 degrees, 330 degrees) direction increases by about 6 dB, and the entire width of the directional characteristics is widened.

As described above, the directivity characteristics of the acoustic output element array 100 can be controlled by the weighting factors (a ₁ , a ₂ ,..., A _N ) corresponding to the number N of control points in the acoustic cutoff region.
Further, when the filter coefficient H is obtained by the least square method instead of the exact solution as shown in the equation (15), it is not guaranteed that the characteristic at the emphasis control point becomes 1. Therefore, the timbre of the acoustic output element array 100 may be different from the timbre of a single speaker.

Therefore, a method of introducing a correction filter F (ω) can be considered for the purpose of compensating for the same timbre as that of a single speaker. The idea is that the final filter coefficient H _m ^F has a characteristic corrected by the correction filter F (ω) as shown in the equation (16). In this case, the correction filter F (ω) that satisfies Expression (17) is used as the correction filter F (ω). The subscript F means the final (Fianl).
H _m ^F (ω) = F (ω) H _m (ω) (16)

  By this correction filter F (ω), for example, the characteristic of the convolution mixed signal at the emphasis control point n = 1 is corrected to be 1. FIG. 9 shows the frequency characteristics of the sound pressure output from the acoustic output element array 100 representing the effect of the correction filter F (ω). FIG. 9 shows the frequency characteristics of the sound in the front of the acoustic output element array 100, that is, in the direction in which the sound is desired to be reproduced. The horizontal axis is frequency (kHz), and the vertical axis is sound pressure (dB).

  The characteristic indicated by the thin line in FIG. 9 is the frequency characteristic of a single speaker. The dotted line is the frequency characteristic of the acoustic output element array 100 using the filter coefficient obtained by the method of least squares. The sound pressure at 1.5 kHz to 3 KHz is about 0.25 dB higher than the characteristics of a single speaker. That is, the tone color differs between the acoustic output element array 100 and a single speaker.

  A characteristic using the correction filter F (ω) with respect to the characteristic of the thin line is indicated by a thick line. It can be seen that the characteristic of the thick line shows a frequency characteristic substantially equal to the characteristic of the dotted line. As described above, the correction filter F (ω) corrects the convolution mixed signal characteristic at the emphasis control point n = 1 to be 1, so that the sound of the acoustic output element array 100 according to the present invention can be obtained from a single speaker. The sound can be almost the same as the sound. Note that the control point to be corrected so that the characteristic of the convolution mixed signal is 1 may be an arbitrary control point.

[Third Embodiment]
The effect of suppressing the sound pressure in the sound blocking area of the first embodiment was about 15 dB at the maximum. For example, in a public place where multiple people gather, if you want to give guidance only to a specific individual using a speaker, or if you want to explain the operation to only that person, the suppression effect is about 15 dB. It may be insufficient.

  Therefore, as a third embodiment, an example will be described in which, in addition to the reproduced sound to be heard by the listener, a disturbing sound is generated so that the reproduced sound cannot be heard by anyone other than the listener. An example of the functional configuration of the acoustic output element array 900 of the third embodiment is shown in FIG.

  The acoustic output element array 900 is different from the first embodiment in that an adder group 911, a second digital filter group 912, an A / D converter 913, and an input terminal 914 are added to the signal supply unit 910. The configuration of the acoustic output element array 100 is different. The digital filter 111 to which the reproduction sound signal of the first embodiment is input is referred to as a first digital filter group 111 for the purpose of distinguishing from the second digital filter 912 here. The first digital filter group 111 is set with a filter coefficient that becomes zero at the control point in the sound reproduction area and zero at the control point in the sound cutoff area. In the following description, the term “group” meaning plural is omitted.

  Similar to the first digital filter 111, the same number of second digital filters 912 as the acoustic output elements 12 are provided. In this example, four second digital filters 912 are provided. Each second digital filter 912 receives a signal obtained by converting an analog interference sound signal input to the input terminal 914 into a digital signal by the A / D converter 913.

A method for setting the filter coefficient P _m (ω) of the second digital filter 912 will be described. It is desirable that the interfering sound is not heard by the listener in the sound reproduction area but is heard in the sound blocking area. Therefore, a control point in the sound reproduction area is set as a suppression control point, and a control point in the sound cutoff area is set as an emphasis control point. An example is shown in FIG. In FIG. 11, three control points positioned at an angle of less than 90 degrees from the axis of the speaker array are set as suppression control points, and three control points positioned at an angle of 90 degrees to 180 degrees are set as emphasis control points. ing.

By setting the control points in this way, the filter coefficient P _m (ω) can be described by Expression (18).

In Expression (18), the top three lines correspond to suppression control points, and the bottom three lines correspond to emphasis control points. Each control point of suppression and emphasis can be arbitrarily set according to the directivity to be realized. The filter coefficient P that minimizes the error by the method of least squares with Equation (18) as G · P = R is obtained by Equation (19) or Equation (20).
P = G ^T · (G ^T · G) ⁻¹ · R (19)
P = G ^T · (G ^T · G + δI) ⁻¹ · R (20)
Here, α ^T is a transposed matrix of α, δ is a small positive constant, and I is an M × M unit matrix.

As shown in the equation (18), the control points are arranged, and the filter coefficients P ₁ (ω), P ₂ (ω), P ₃ (ω), obtained under the condition that the number of the acoustic output elements 12 is four, FIG. 12 shows the directivity characteristics of the acoustic output element array 900 in which P ₄ (ω) is set to each of the four second digital filters 912. FIG. 12 shows the reproduced sound with a solid line and the disturbing sound with a broken line in the same graph as FIG. 5 described above. With respect to the direction of 0 degrees where the listener is present, the reproduced sound is reproduced at 0 dB and the disturbing sound is reproduced at -17 dB. That is, the reproduced sound signal can be heard with an S / N ratio of 17 dB.

  On the other hand, in the direction of 60 degrees, the reproduced sound signal is −15 dB, the disturbing sound signal is −7 dB, and the S / N ratio of the reproduced sound signal is −8 dB. In this direction, the reproduced sound signal is heard in a large disturbance sound, and the reproduced sound signal is almost inaudible. In the 90 ° (side) and 180 ° (back) directions, the reproduced sound signal cannot be heard because the sound pressure of the interfering sound signal is high at an S / N ratio of −15 dB or more. As described above, according to the third embodiment, it is possible to further enhance the effect of suppressing the reproduced sound signal in the sound blocking area by using the disturbing sound signal.

  When the reproduced sound signal is, for example, a voice in which a person talks, the amplitude has a characteristic inversely proportional to the frequency. In this case, if the disturbing sound signal is white noise, the reproduced sound signal may be heard even when the S / N ratio of the reproduced sound signal is low.

  FIG. 13 schematically shows an example of frequency characteristics of the reproduction sound signal and the interference sound signal. In FIG. 13, the horizontal axis represents frequency and the vertical axis represents amplitude. In FIG. 13A, the frequency characteristic when a human voice is a reproduced sound signal is represented by a solid line, and the white noise interference sound signal is represented by a broken line. In this case, since the amplitude of the interference sound signal is uniform even if the frequency changes, the situation where the amplitude of the reproduced sound signal exceeds the amplitude of the interference sound signal is likely to occur although it varies depending on the position of the listener. As a result, it becomes easy to hear the reproduced sound signal even in the sound blocking area.

  Therefore, a method is conceivable in which the frequency characteristic of the interfering sound signal is made to have the same tendency as that of the reproduced sound signal. By making the interference sound signal so-called pink noise having a frequency amplitude characteristic in which the amplitude of a high frequency gradually decreases, the suppression performance of the reproduced sound signal in the sound cutoff region can be further improved.

  This is shown in FIG. By making the interfering sound signal pink noise, the amplitude of the reproduced sound signal does not exceed the amplitude of the interfering sound signal, and the reproduced sound signal can be made more difficult to hear in the sound cutoff region. If the frequency amplitude characteristic of the reproduced sound signal is a characteristic that the amplitude increases at a high frequency, the same effect can be obtained by making the frequency amplitude characteristic of the interfering sound signal have the same tendency. It is.

  As a suitable interfering sound signal, a signal obtained by adding sounds obtained by recording a plurality of sounds having different utterance contents can be considered. In the sound blocking area, singing voice or music that can be used as BGM (Back Ground Music) may be used.

  As a modification of the third embodiment, a method of matching the time envelope characteristic of the disturbing sound signal with the time envelope characteristic of the reproduced sound signal is also conceivable. FIG. 14 shows a functional configuration example of the modification. The signal supply unit 910 ′ according to the modification is different from the third embodiment in that it includes an interference sound generation unit 141. The interference sound generation unit 141 generates an interference sound having the same time envelope characteristic as that of the reproduced sound signal and supplies it to the second digital filter 912.

  The interference sound signal generation unit 141 squares the input digital reproduction sound signal to calculate a time average, and obtains a square root thereof to extract a time envelope characteristic of the reproduction sound signal. Then, the time envelope characteristic is multiplied by, for example, white noise to generate an interference sound signal. The time envelope characteristic of the disturbing sound signal generated in this way is the same as that of the reproduced sound signal. Therefore, the disturbing sound signal can make it difficult to hear the reproduced sound signal in the sound blocking area. Pink noise may be used instead of white noise.

[Fourth Embodiment]
Next, a fourth embodiment of the present invention will be described.
This embodiment is an embodiment in which the present invention is applied to a hands-free device used when performing mutual voice communication between communication terminal devices as in a communication conference system. In the following description, differences from the contents described so far will be mainly described.

FIG. 15 is a configuration diagram showing the configuration of the acoustic output element array 200 of the third embodiment.
As shown in FIG. 15, the acoustic output element array 200 according to the present embodiment includes a plurality of acoustic output elements 12, a signal supply unit 110 including digital filters 111 that are independent from each other, and a voice communication unit disposed in the acoustic blocking area. A sound receiving element 263 (for example, a microphone), an echo canceller 266, an input terminal 264, and an output terminal 265 are provided. In FIG. 15, the description of the amplifier 112, the D / A converter 113, and the A / D converter 114 included in the signal supply unit 110 is omitted.

  Here, similarly to the first embodiment, the plurality of sound output elements 12 are arranged on the same axis, and are arranged in directions in which the respective directivity characteristics are axially symmetric with respect to the axis. However, the acoustic output elements 12 are not disposed so that the acoustic signal emission surfaces of the acoustic output elements 12 face the listener 261, but are disposed so that they face upward perpendicular to the horizontal plane. That is, the axis on which each acoustic output element 12 is arranged is an axis perpendicular to the ground, and each acoustic output element 12 is arranged with its respective acoustic signal emission surface facing away from the ground (see FIG. 15). Note that “vertical” is a concept that includes not only mathematically strict vertical but also substantially vertical that can achieve practically similar effects.

  The configuration of the signal supply unit 110 may be the same as that of the first embodiment. According to the arrangement of the sound output elements 12, the sound reproduction area is set above the sound output elements 12, and the sound is removed in the other areas. It is only necessary to set a suppression region, set each virtual control point, and set the filter coefficient of the digital filter 111 as in the first embodiment. Further, the sound receiving element 263 for voice communication is arranged in the sound blocking area, but it is more desirable to set the filter coefficient with this installation position as a virtual control point of the sound blocking area.

  In the case of the acoustic output element array 200 according to the present embodiment, the sound emitted from the listener 261 is received by the sound receiving element 263, converted into an electric signal, and output from the output terminal 265 through the echo canceller 266. Sent to the device. On the other hand, the transmission signal transmitted from the communication partner apparatus is used for the processing of the echo canceller 266 and is input to the signal input terminal 115. The signal input terminal 115 generates each supply signal using each digital filter 111 as in the first embodiment, supplies each generated supply signal to the corresponding acoustic output element 12, and each acoustic output element 12 An acoustic signal corresponding to this is emitted.

  In the case of such an acoustic output element array 200, the directivity is upward, and thus the listener 261 listens to a sound that deviates from the lateral direction from the region where the acoustic signal is emitted. In this case, the sound can be heard sufficiently near the acoustic output element array 200, but cannot be heard when it is separated from the sound output element array 200. As a result, the sound is reproduced only within a specific area, and a conference apparatus that does not leak sound around. It becomes possible to form.

  In addition, by arranging the sound receiving element 263 for picking up the voice of the listener 261 in the sound blocking area of the sound output element array 200, the acoustic signal emitted from each sound output element 12 is echoed into the sound receiving element 263. The signal can be suppressed, and as a result, it can be expected to reduce the load of the echo canceller 266 at the subsequent stage.

  FIG. 16 is a diagram in which the directivity characteristics of the acoustic output element array 200 of the third embodiment and the arrangements of the acoustic output elements 12, the sound receiving elements 263 and 267, and the listeners 261 and 262 are overlapped. Note that the sound receiving element 267 is the same as the sound receiving element 263. The notation in this figure is the same as in FIG. 5, and the orientation specification is performed by specifying the filter coefficient by setting the virtual suppression control point to L = 50 cm and angles of 90 degrees, 120 degrees, 150 degrees, and 180 degrees. This is the case.

  As shown in FIG. 16, in the acoustic output element array 200, sound reaches the ears of the listeners 261 and 262 at a sufficient volume, but the sound is low at the positions of the sound receiving elements 263 and 267.

  The present invention is not limited to the embodiment described above. For example, in each of the above-described embodiments, the example in which the filter coefficient of the digital filter 111 is set by pre-calculation is shown. However, when the distance between the acoustic output element and the control point changes dynamically and the distance information can be obtained dynamically, the filter coefficient of the digital filter 111 is set dynamically, and the directivity is changed. It is good also as changing dynamically.

  In each of the above embodiments, each acoustic output element constituting the acoustic output element array has the same performance, but some of the acoustic output elements constituting the acoustic output element array are different from other acoustic output elements. May have performance. For example, some of the sound output elements constituting the sound output element array may be double sound source elements, and the other sound output elements may be single sound source elements.

  In each of the above embodiments, each acoustic output element is arranged so that each acoustic signal emission surface of each acoustic output element constituting the acoustic output element array faces the same direction, but with respect to the axis of the acoustic output element array. As long as an axially symmetric directional characteristic can be realized, some of the sound output elements may face different directions from other sound output elements.

  In addition, the various processes described above are not only executed in time series according to the description, but may be executed in parallel or individually according to the processing capability of the apparatus that executes the processes or as necessary. Needless to say, other modifications are possible without departing from the spirit of the present invention.

  The present invention can be used, for example, in audio equipment, communication equipment, and the like that reproduce audio signals.

FIG. 1 is a configuration diagram showing a configuration of a speaker array in which a plurality of speakers shown in Patent Document 2 are arranged in a straight line. FIG. 2 is a diagram for illustrating the configuration of the speaker array. FIG. 3 is a diagram for explaining the arrangement of the sound output elements and the control points constituting the sound output element array of the first embodiment. FIG. 4 is a configuration diagram for explaining the configuration of the acoustic output element array of the first embodiment. FIG. 5 is a graph illustrating the measurement result of the directivity of the acoustic output element array according to the first embodiment. FIG. 6 is a diagram illustrating an example of directivity characteristics of the sound output array according to the second embodiment. FIG. 7 is a diagram illustrating an example of directivity characteristics of the sound output array according to the second embodiment. FIG. 8 is a diagram illustrating an example of directivity characteristics of the sound output array according to the second embodiment. FIG. 9 is a diagram illustrating a frequency characteristic of sound pressure representing the effect of the correction filter F (ω). FIG. 10 is a configuration diagram for explaining the configuration of the acoustic output element array of the third embodiment. FIG. 11 is a diagram for explaining the arrangement of acoustic output elements and control points that constitute the acoustic output element array of the third embodiment. FIG. 12 is a graph illustrating experimental results of directivity characteristics of the acoustic output array according to the third embodiment. FIGS. 13A and 13B are diagrams schematically showing the relationship between the frequency characteristics of the reproduction sound signal and the interference sound signal. FIG. 13A shows a case where the interference sound signal is white noise, and FIG. 13B shows a case where the interference sound signal is pink noise. FIG. FIG. 14 is a configuration diagram for explaining a configuration of a modified example of the third embodiment. FIG. 15 is a configuration diagram showing the configuration of the acoustic output element array of the fourth embodiment. FIG. 16 is a diagram in which the directivity characteristics of the acoustic output element array according to the fourth embodiment and the arrangements of the acoustic output elements, the sound receiving elements, and the listener are overlapped.

Claims (15)

A plurality of acoustic output elements arranged on the same axis and arranged in a direction in which each directional characteristic is axially symmetric with respect to the axis;
A first digital filter group having the same number of digital filters as the sound output elements to which the reproduced sound signal is input; and a second digital filter group having the same number of digital filters as the sound output elements to which the interference sound signal is input; An adder group for adding output signals of the filters of the first digital filter group and the second digital filter group, and the plurality of acoustic output elements of the first digital filter group and the second digital filter group Each supply signal generated by adding the corresponding output signals to each other, and a signal supply unit that supplies the corresponding acoustic output element,
Each acoustic output element outputs an acoustic signal corresponding to each of the supplied supply signals,
The first digital filter group includes each acoustic signal output from each acoustic output element from each acoustic output element to each control point when approximating that each acoustic output element to each control point is a free space. In the case of convolution with each approximate transfer function and mixing at each control point, the convolution mixed signal becomes non-zero at the control point in the sound reproduction region set in the axial direction in which the sound output elements are arranged. , A first digital filter group in which a filter coefficient is set to zero at a control point of the sound cutoff region excluding the sound reproduction region,
The second digital filter group includes each acoustic signal output from each acoustic output element from each acoustic output element to each control point when approximating that each acoustic output element to each control point is a free space. In the case of convolution with each approximate transfer function and mixing at each control point, the convolution mixed signal becomes zero at the control point of the sound reproduction region set in the axial direction in which the sound output elements are arranged, A second digital filter group in which a filter coefficient is set to be non-zero at a control point in the sound blocking area excluding the sound reproduction area;
Each approximate transfer function is a function determined by each distance from each acoustic output element to each control point.
An acoustic output element array. The acoustic output element array according to claim 1,
The acoustic output element array, wherein the disturbing sound signal has a frequency amplitude characteristic having the same tendency as the reproduced signal. The acoustic output element array according to claim 1,
An acoustic output element array comprising: an interference sound generation unit that receives the reproduced sound signal as input and generates an interference sound having the same time envelope characteristic as the reproduced sound signal and inputs the generated interference sound to the second digital filter group . The acoustic output element array according to any one of claims 1 to 3,
Each of the control points is set on only one point of a set of points on the axis on which the sound output elements are arranged and / or axisymmetric with respect to the axis.
An acoustic output element array. The acoustic output element array according to any one of claims 1 to 4,
The approximate transfer function from at least some of the acoustic output elements to any control point is such that the distance between the acoustic output element and the control point is r, the speed of sound is c, the imaginary unit is j, and the frequency is (α / r) · exp (−j · ω · r / c) where ω is assumed, the number of Napier is exp, and α ≠ 0 is a constant.
An acoustic output element array. The acoustic output element array according to any one of claims 1 to 5,
At least some of the acoustic output elements are dual sound source elements that output a first acoustic signal from the first part and output a second acoustic signal whose phase is inverted from that of the first acoustic signal from the second part. ,
In the approximate transfer function from the acoustic output element that is the dual sound source element to an arbitrary control point, the distance between the first part and the control point is r _F, and the distance between the second part and the control point is (Α _F / r) in the case where r _R is _R r, sound velocity is c, imaginary unit is j, frequency is ω, Napier number is exp, and α _F > 0 and α _R > 0 are constants. _{F) · exp (-j · ω} · r F / c) - is a _{(α R / r R) ·} exp (-j · ω · r R / c),
An acoustic output element array. The acoustic output element array according to any one of claims 1 to 6,
The first and second digital filter groups are digital filters in which a filter coefficient that changes a value of the convolution mixed signal according to a frequency is set at at least some control points of the sound reproduction region.
An acoustic output element array. The acoustic output element array according to any one of claims 1 to 7,
Each of the sound output elements outputs at least a part of the sound signal corresponding to each of the supplied supply signals toward the sound reproduction region side,
An acoustic output element array. The acoustic output element array according to any one of claims 1 to 8,
A sound receiving element for voice communication disposed in the sound blocking region;
The output signal output from the sound receiving element is transmitted to the communication partner device,
The signal supply unit supplies each of the supply signals generated by inputting a transmission signal transmitted from the communication partner device to the first digital filter group to the acoustic output element.
An acoustic output element array. The acoustic output element array according to claim 9, wherein
The axis on which the plurality of acoustic output elements are arranged is an axis perpendicular to the ground,
Each of the acoustic output elements is arranged with the respective acoustic signal emission surfaces facing in the direction opposite to the ground.
An acoustic output element array. For a plurality of acoustic output elements arranged on the same axis and arranged in a direction in which each directional characteristic is axially symmetric with respect to the axis,
A first digital filter group having the same number of digital filters as the sound output elements to which the reproduced sound signal is input; and a second digital filter group having the same number of digital filters as the sound output elements to which the interference sound signal is input; A signal supply unit including an adder group that adds output signals of the filters of the first digital filter group and the second digital filter group includes the plurality of the first digital filter group and the second digital filter group. Supplying each supply signal generated by adding the respective output signals corresponding to the acoustic output elements to the corresponding acoustic output elements;
Each acoustic output element has a process of outputting an acoustic signal corresponding to each of the supplied supply signals,
The first digital filter group includes each acoustic signal output from each acoustic output element from each acoustic output element to each control point when approximating that each acoustic output element to each control point is a free space. In the case of convolution with each approximate transfer function and mixing at each control point, the convolution mixed signal becomes non-zero at the control point in the sound reproduction region set in the axial direction where the sound output elements are arranged. , A first digital filter group in which a filter coefficient is set to zero at a control point of the sound cutoff region excluding the sound reproduction region,
The second digital filter group includes each acoustic signal output from each acoustic output element from each acoustic output element to each control point when approximating that each acoustic output element to each control point is a free space. In the case of convolution with each approximate transfer function and mixing at each control point, the convolution mixed signal becomes zero at the control point of the sound reproduction region set in the axial direction in which the sound output elements are arranged, A second digital filter group in which a filter coefficient is set to be non-zero at a control point in the sound blocking area excluding the sound reproduction area;
Each approximate transfer function is a function determined by each distance from each acoustic output element to each control point.
A sound output method characterized by the above. The sound output method according to claim 11,
The sound output method, wherein the disturbing sound signal has a frequency amplitude characteristic having the same tendency as the reproduced signal. The sound output method according to claim 11,
The interference sound generation unit includes an interference sound generation process in which the reproduction sound signal is input, the interference sound having the same time envelope characteristic as the reproduction sound signal is generated and input to the second digital filter group. Sound output method. The sound output method according to any one of claims 11 to 13,
Each of the control points is set on only one point of a set of points on the axis on which the sound output elements are arranged and / or axisymmetric with respect to the axis.
A sound output method characterized by the above. The sound output method according to any one of claims 11 to 14,
The approximate transfer function from at least some of the acoustic output elements to any control point is such that the distance between the acoustic output element and the control point is r, the speed of sound is c, the imaginary unit is j, and the frequency is (α / r) · exp (−j · ω · r / c) where ω is assumed, the number of Napier is exp, and α ≠ 0 is a constant.
A sound output method characterized by the above.

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