JP5151985B2 - Speaker array and microphone array - Google Patents

Description

  The present invention relates to a technique for improving the directivity of a speaker array and a microphone array, and more particularly to a technique for improving the directivity in a low sound range.

A sound field is formed only in a specific direction by using a speaker array or microphone array formed by arranging a plurality of transducers such as speakers and microphones linearly at a predetermined interval, or only sound coming from a specific direction is received. The technology of picking up sound is widely used.
By the way, in this kind of speaker array or microphone array, it is desirable that the same directivity characteristic can be realized over a wide band from a high sound range to a low sound range. However, the directivity characteristics in the low sound range improve as the array length of the speaker array or microphone array (value obtained by multiplying the number of transducers by the arrangement interval of the transducers) increases (see Non-Patent Document 1). In order to ensure sufficient directivity in the sound range, there has been a problem that the apparatus size of the speaker array and microphone array becomes large.
Thus, various techniques for solving the above-described problems have been proposed in the past, and an example thereof is the technique disclosed in Non-Patent Document 2. This Non-Patent Document 2 describes a Dolph whose cross section in the spatial frequency direction in the two-dimensional frequency plane has a ripple characteristic such as a stop band in the amplitude characteristics of a digital filter connected to each transducer constituting a speaker array or a microphone array. -A technology that expands the band that can give the same directivity to the low frequency range by setting the filter coefficient of each digital filter so that it becomes the amplitude characteristic (or its approximate characteristic) of the Chebyshev filter .
Toshiro Oga, Yoshio Yamazaki, Yutaka Kaneda “Acoustic System and Digital Signal Processing” The Institute of Electronics, Information and Communication Engineers 1993-05, p176-186 Yasushi Matsumoto, Kiyoshi Nishikawa “Design Method for Directional Array Speakers with Constant Sidelobe” IEICE Technical Report 2004-74 p13-18

However, ripples with ripple characteristics such as stopbands usually exist in regions other than non-physical regions (regions where the normalized spatial frequency is greater than the normalized time frequency in the two-dimensional frequency plane). If a large amplitude is given to the ripple such as the stop band for the purpose of improvement, there is a problem that the amplitude level of the side lobe, which is an originally unnecessary directivity characteristic, increases. Further, when actually using a microphone array or speaker array, it is convenient if the directional main axis can be directed in an arbitrary direction (hereinafter referred to as steering). The technology that makes it possible is not disclosed.
The present invention has been made in view of the above problems, and improves the directivity of the speaker array and microphone array in the low frequency range without increasing the array length and avoids an increase in the amplitude level of the side lobe. It is another object of the present invention to provide a technique that enables the steering of the directional spindle.

  In order to solve the above problems, the present invention is provided with a plurality of speakers arranged linearly at a predetermined interval, and corresponding to each of the plurality of speakers, and a predetermined filter coefficient is preset. And a one-dimensional digital filter that outputs and outputs the input voice data according to the filter coefficient according to the filter coefficient, and supplies the input voice data to each one-dimensional digital filter. In the speaker array for supplying the audio data output from the three-dimensional digital filter to the corresponding speaker and outputting the sound corresponding to the audio data, the filter coefficient set in each one-dimensional digital filter is the one for each one. When the frequency characteristics of a two-dimensional digital filter formed by a two-dimensional digital filter are represented on a two-dimensional frequency plane In the cross section in the spatial frequency direction, the non-physical region has a plurality of ripples in the stop band, and the amplitude of the ripples in the non-physical region is larger than the amplitude of the ripples in the physical region among the plurality of ripples. For the ripples in the region where the spatial frequency is smaller than the predetermined value, the amplitude characteristic in which the bandwidth of the ripple in the time frequency direction is narrowed as the spatial frequency corresponding to the ripple is small. Provided is a speaker array characterized by being a filter coefficient given to a two-dimensional digital filter.

  In order to solve the above-described problem, the present invention provides a plurality of microphones arranged linearly at a predetermined interval and each of the plurality of microphones, and a predetermined filter coefficient is set in advance. And a one-dimensional digital filter that performs a filtering process on the input audio data according to the filter coefficient and outputs the same, and the audio data output from each of the plurality of microphones is associated with the corresponding 1 In the microphone array that outputs the sum signal of the audio data output from each one-dimensional digital filter while being supplied to the one-dimensional digital filter, the filter coefficient set in each one-dimensional digital filter is the one-dimensional digital filter. The frequency characteristics of a two-dimensional digital filter formed by a filter are represented on a two-dimensional frequency plane. In the cross section in the spatial frequency direction, there are a plurality of ripples in the stop band, and the amplitude of the ripple in the non-physical region is larger than the amplitude of the ripple in the physical region among the plurality of ripples, For ripples in a region where the spatial frequency is smaller than a predetermined value among ripples in the non-physical region, the smaller the spatial frequency corresponding to the ripple, the narrower the bandwidth in the time frequency direction of the ripple is. There is provided a microphone array characterized by being filter coefficients that give characteristics to the two-dimensional digital filter.

  In order to solve the above-described problem, the present invention provides a plurality of speakers arranged linearly at a predetermined interval, and a plurality of speakers provided corresponding to each of the plurality of speakers. A one-dimensional digital filter that performs a filtering process and outputs the input audio data to each one-dimensional digital filter, while the audio data output from each one-dimensional digital filter In a speaker array for supplying sound to a speaker and outputting sound corresponding to the sound signal, designation means for designating a steering angle of a directional spindle within a predetermined angle range, and according to the steering angle designated by the designation means Specifying means for specifying for each speaker a delay amount to be added to the sound output from each of the plurality of speakers; Delay means for applying to the audio data output from each of the plurality of one-dimensional digital filters a delay corresponding to the delay amount specified for each speaker by a determining means, and each one-dimensional digital filter When the frequency characteristic of the two-dimensional digital filter formed by each of the one-dimensional digital filters is represented by a two-dimensional frequency plane, (1) a plurality of ripples are provided in the stop band in the cross section in the spatial frequency direction. (2) the amplitude of the ripple in the non-physical region among the plurality of ripples is larger than the amplitude of the ripple in the physical region, and (3) within the angular range of the non-physical region. When the steering shaft is steered at the maximum steering angle, a ripple is formed only in an area where a part of the ripple does not protrude into the physical area. Providing a speaker array to symptoms.

  In order to solve the above-described problem, the present invention provides a plurality of speakers arranged linearly at a predetermined interval, and a plurality of speakers provided corresponding to each of the plurality of speakers. A one-dimensional digital filter that performs a filtering process and outputs the input audio data to each one-dimensional digital filter, while the audio data output from each one-dimensional digital filter A speaker array for supplying to a speaker and outputting a sound corresponding to the sound signal, comprising: designation means for designating a steering angle of a directional spindle within a predetermined angle range, and a filter coefficient of each one-dimensional digital filter Represents a frequency characteristic of a two-dimensional digital filter formed by each one-dimensional digital filter in a two-dimensional frequency plane. (1) A passband is formed so as to form a main lobe in the steering angle direction designated by the designation means, and a first between the boundary between the physical region and the non-physical region and the passband is formed. A second stop zone in the non-physical region, (2) having a plurality of ripples in the stop zone across the first and second stop zones, (3) There is provided a speaker array characterized in that the amplitude of the ripple in the second stop band is larger than the amplitude of the ripple in the first stop band.

  In order to solve the above-described problem, the present invention provides a plurality of microphones arranged linearly at a predetermined interval, and a plurality of microphones corresponding to each of the plurality of microphones. A one-dimensional digital filter that performs filtering and outputs, and supplies audio data output from each of the plurality of microphones to the corresponding one-dimensional digital filter, while outputting from the corresponding one-dimensional digital filter In the microphone array that outputs the sum signal of the audio data, a designation unit that designates the steering angle of the directional main spindle within a predetermined angle range, and the plurality of the plurality of voices according to the steering angle designated by the designation unit The delay amount to be added to the audio data output from each microphone is specified for each microphone. Delay means for giving a delay corresponding to the delay amount specified for each of the microphones by the specifying means to the audio data output from the microphones and inputting it to the corresponding one-dimensional digital filter. And the filter coefficient of each one-dimensional digital filter is expressed as follows: (1) in the spatial frequency direction when the frequency characteristic of the two-dimensional digital filter formed by each one-dimensional digital filter is represented by a two-dimensional frequency plane. The cross section has a plurality of ripples in the blocking area, (2) among the plurality of ripples, the ripple amplitude in the non-physical region is larger than the amplitude of the ripple in the physical region; When the steering spindle is steered at the maximum steering angle within the above-mentioned angle range, a part of the ripple enters the physical area. Forming a ripple only in the absence area, to provide a microphone array according to claim.

  In order to solve the above-described problem, the present invention provides a plurality of microphones arranged linearly at a predetermined interval, and a plurality of microphones corresponding to each of the plurality of microphones. A one-dimensional digital filter that performs filtering and outputs, and supplies audio data output from each of the plurality of microphones to the corresponding one-dimensional digital filter, while outputting from the corresponding one-dimensional digital filter A microphone array that outputs a sum signal of the audio data, and a designating unit that designates the steering angle of the directional main axis within a predetermined angle range, and the filter coefficient of each one-dimensional digital filter includes The frequency characteristics of a two-dimensional digital filter formed by a one-dimensional digital filter are represented on a two-dimensional frequency plane In this case, (1) a pass band is formed so as to form a main lobe in the steering angle direction designated by the designation means, and a first between the boundary between the physical area and the non-physical area and the pass area. A second stop zone in the non-physical region, (2) having a plurality of ripples in the stop zone across the first and second stop zones, (3) There is provided a microphone array characterized in that the amplitude of the ripple in the second stop band is larger than the amplitude of the ripple in the first stop band.

  According to the present invention, it is possible to improve the directivity of the speaker array and the microphone array in the low sound range without increasing the array length, and to avoid an increase in the level of the side lobe. The main shaft can be steered.

1 is a block diagram illustrating an electrical configuration of a speaker array 100 according to a first embodiment of the present invention. 3 is a diagram illustrating an example of amplitude characteristics of a two-dimensional digital filter of the speaker array 100 using a two-dimensional frequency plane. FIG. It is the figure which showed a part of the same amplitude characteristic with the equal amplitude characteristic figure. It is the figure which plotted the amplitude characteristic of the speaker array 100 for every predetermined angle. It is the figure which plotted the directional characteristic of the speaker array 100 for every predetermined frequency. It is the figure which showed the relationship between the frequency of the acoustic beam output from the speaker array 100, and the main lobe width of the acoustic beam. design method sectional characteristic at f ₁ ≧ f _l is disclosed in Non-Patent Document 2 is a diagram for explaining the. f 1 _<is a diagram for explaining a method of designing the cross-sectional characteristics at f _l is disclosed in Non-Patent Document 2. It is a figure for demonstrating the design method of the cross-sectional characteristic which concerns on this embodiment. It is a figure for showing the characteristic of the design result of the one-dimensional filter by the equiripple filter design program of Parks & McClellan. It is a figure which shows the design example of the one-dimensional filter by the equiripple filter design program of Parks & McClellan, and the design example of the one-dimensional filter of Dolph-Chebyshev characteristic. It is a figure which shows the electrical structure of the microphone array 200 which concerns on 2nd Embodiment of this invention. It is a block diagram showing the electric constitution of the speaker array 500 which concerns on 3rd Embodiment of this invention. It is the figure which showed the amplitude characteristic and directivity characteristic of the same speaker array. It is the figure which showed the amplitude characteristic and directional characteristic of the speaker array which has not provided the use restriction in a non-physical area. design method sectional characteristic at f ₁ ≧ f _l is disclosed in Non-Patent Document 2 is a diagram for explaining the. f 1 _<is a diagram for explaining a method of designing the cross-sectional characteristics at f _l is disclosed in Non-Patent Document 2. It is a figure for demonstrating the design method of the cross-sectional characteristic which concerns on this embodiment. It is a figure for showing the characteristic of the design result of the one-dimensional filter by the equiripple filter design program of Parks & McClellan. It is a figure which shows the design example of the one-dimensional filter by the equiripple filter design program of Parks & McClellan, and the design example of the one-dimensional filter of Dolph-Chebyshev characteristic. It is a figure which shows the electrical structure of the microphone array 200 which concerns on 4th Embodiment of this invention. It is a figure which shows the electrical structure of the speaker array 300 which concerns on 5th Embodiment of this invention. It is a figure which shows the electrical structure of the speaker array 400 which concerns on 6th Embodiment of this invention. It is a figure which shows the relationship between a two-dimensional frequency plane and a cross section. It is a figure which shows an example of a cross-section design result. It is a figure showing an example of the amplitude characteristic of the two-dimensional digital filter of the speaker array 400 on a two-dimensional frequency plane. It is an equiamplitude characteristic diagram of the same amplitude characteristic. 4 is a diagram illustrating directional characteristics of the speaker 400. FIG. It is a figure which shows the frequency characteristic concerning modification (3), (6).

Explanation of symbols

  100, 300, 400, 500 ... speaker array, 110-i (i: natural number of 1 to n) ... speaker, 120-i ... one-dimensional digital filter, 130-i ... delay circuit, 140 ... U / I section, 150 ... Control unit, 160 ... Storage unit, 200 ... Microphone array, 210-i ... Microphone

Hereinafter, the best mode for carrying out the present invention will be described with reference to the drawings.
(A. First embodiment)
(A-1: Configuration)
FIG. 1 is a block diagram illustrating a configuration example of a speaker array 100 according to the first embodiment of the present invention. As shown in FIG. 1, the speaker array 100 includes transducers (speakers in this embodiment) 110-1 and 110-2 that are linearly arranged at predetermined intervals (in this embodiment, a constant interval D). ... 110-n and the same number of one-dimensional digital filters 120-1, 120-2 ... 120-n as the speakers.

In the speaker array 100 of FIG. 1, an audio signal (analog signal) supplied from an external sound source (not shown) is converted into digital data (hereinafter referred to as audio data) by an A / D converter (not shown). Audio data is supplied to each of the one-dimensional digital filters 120-i (i: natural numbers of 1 to n: the same applies hereinafter). In the present embodiment, a case where an analog audio signal is supplied from an external sound source to the speaker array 100 will be described, but it is of course possible that digital audio data is supplied from the external sound source to the speaker array 100. In this way, when digital audio data is supplied from an external sound source, it is needless to say that the speaker array 100 does not need to be provided with the A / D converter.
In each of the one-dimensional digital filters 120-i in FIG. 1, filter coefficients characteristic of the speaker array according to the present invention are set in advance. Each of these one-dimensional digital filters 120-i performs a filtering process corresponding to the filter coefficient on the audio data delivered from the A / D converter and outputs the result.
The audio data output from each one-dimensional digital filter 120-i is converted into an audio signal by a D / A converter (not shown), and the speaker 110- corresponding to the one-dimensional digital filter 120-i. i. As a result, a sound corresponding to the sound signal supplied from the D / A converter is emitted from each of the speakers 110-i. In the present embodiment, the case where the speaker 110-i emits a sound corresponding to an analog audio signal will be described. However, the speaker 110-i emits a sound corresponding to digital audio data. Needless to say, it is not necessary to provide the D / A converter.
The above is the configuration of the speaker array 100.

As described above, the hardware configuration of the speaker array 100 according to the present embodiment is not different from the hardware configuration of the conventional speaker array. However, in the speaker array 100 according to the present embodiment, filter coefficients characteristic of the speaker array according to the present invention are preset in each of the one-dimensional digital filters 120-i. A characteristic amplitude characteristic of the speaker array according to the present invention is given to the formed two-dimensional digital filter, and a characteristic directivity characteristic of the speaker array according to the present invention is realized by the amplitude characteristic. .
Hereinafter, the amplitude characteristic of the two-dimensional digital filter formed by the one-dimensional digital filter 120-i and the directivity characteristic realized by the amplitude characteristic will be described with reference to the drawings. In the following, it is assumed that each of the speakers 110-i has an ideal characteristic (that is, a characteristic that the directivity characteristic does not depend on the frequency of the output sound). In the following, it is assumed that the speaker arrangement interval D = 0.068 [m], the sampling frequency f _s = 7451 [Hz], the number of FIR taps = 241, and n (the number of speakers) = 15.

(A-2: Amplitude characteristics and directivity characteristics of a two-dimensional digital filter)
2 to 6 are diagrams showing the amplitude characteristics of the two-dimensional digital filter of the speaker array 100 and the directivity characteristics realized by the amplitude characteristics.
FIG. 2 is a diagram showing the amplitude characteristics of the two-dimensional digital filter formed by the one-dimensional digital filter 120-i on a two-dimensional frequency plane. FIG. 3 shows a part of the amplitude characteristics shown in FIG. FIG. 6 is a diagram showing an equiamplitude characteristic diagram in which the normalized time frequency f ₁ is in the range of 0 to 0.5 and the normalized spatial frequency f ₂ is in the range of 0 to 0.5. The normalized time frequency is a value obtained by normalizing the time frequency by the reciprocal of the time sampling interval, and the normalized spatial frequency is obtained by normalizing the spatial frequency by the reciprocal of the speaker arrangement interval D. Value.
As is clear by referring to FIGS. 2 and 3, a plurality in the speaker array 100 according to the present embodiment, the normalized time-frequency f ₁ is lower region (e.g., region f ₁ is 0 to 0.1) And a ripple in the non-physical area is given a large amplitude (in this embodiment, “1”). For a ripple in the physical area, the amplitude of the ripple is in the non-physical area. Lower than the inner ripple. As is clear from FIG. 2 and FIG. 3, since ripples in the non-physical region are equal ripples having substantially the same amplitude, the amplitude characteristics shown in FIG. 2 and FIG. Called a characteristic.

In addition, for ripples in the non-physical region, for a ripple in a region where the normalized spatial frequency f ₂ is equal to or greater than a predetermined value (in FIG. 3, a region where f ₂ ≧ 0.25), a normalized time frequency while the bandwidth of the direction is substantially constant, the normalized spatial frequency f ₂ is about ripple area is less than the predetermined value, the bandwidth of as normalized time-frequency direction value of the normalized spatial frequency f2 becomes smaller It is narrower. The reason why the normalized spatial frequency f ₂ is that narrowing the bandwidth of the above for the ripple given is less than the value area, the more normalized time-frequency direction value of the normalized spatial frequency f2 becomes smaller, It is as follows.
The width of the main lobe depends on the number of ripples in the non-physical region and its amplitude (for example, the larger the number of ripples in the non-physical region in the total number of ripples in the stop region, the narrower the passband, the main lobe width. Is generally known (see Non-Patent Document 2). As for the ripple in the region where the normalized spatial frequency f ₂ is less than the predetermined value as described above, by reducing the bandwidth in the normalized time frequency direction as the value of the normalized spatial frequency f ₂ decreases, Increased number of ripples in non-physical regions available in regions where the normalized time frequency f ₁ is low (eg, regions where f ₁ is 0 to 0.1) compared to keeping the bandwidth constant This is because it is expected that the main lobe width can be further reduced. In the present embodiment, a case where the predetermined value (the value of the normalized spatial frequency that divides the non-physical area) for the normalized spatial frequency f ₂ is 0.25 will be described. The value is 0.25. It is not limited to this, and it may be determined as appropriate according to the directivity characteristic required for the speaker array 100 (that is, the amplitude characteristic that realizes the directivity characteristic).

FIG. 4 shows the amplitude characteristics shown in FIG. 2 in the plane including each speaker 110-i and the observation point of the sound output from the speaker array 100, and the direction perpendicular to the arrangement direction of the speakers 110-i is 0 degree. It is the figure shown as a frequency characteristic about the angle (angle (phi) of FIG. 1) of the observation point direction seen from the center of the speaker arrangement | sequence in the case. FIG. 4 shows frequency characteristics for φ = 0 °, 12.5 °, 20 °, 60 °, and 90 °.
As is apparent from FIG. 4, the amplitude level of the acoustic beam output from the speaker array 100 according to the present embodiment is φ = 12.5 ° when the frequency is higher than a certain value. For the direction, the amplitude level is reduced by about 6 [dB] compared to the direction of φ = 0 °, and for the directions of φ = 20 °, 60 °, and 90 °, the amplitude level is lower than the direction of φ = 0 °. It can be seen that it has decreased by about 20 [dB].
FIG. 5 is a diagram showing the amplitude characteristics shown in FIG. 2 as directivity characteristics at several frequencies (371.09473 Hz, 745.882864 Hz, 1491.6553 Hz, 22333.8447 Hz, and 3354.4053 Hz).
As apparent from FIGS. 4 and 5, according to the speaker array 100 of the present embodiment, the side lobe level is maintained while keeping the main lobe width of the acoustic beam constant for frequencies above a certain value. It can be seen that it can be kept substantially constant (-20 dB in this case).

FIG. 6 shows the width of a region where the amplitude of the acoustic beam is reduced by 6 [dB] compared to the main lobe width (φ = 0 ° direction) for a plurality of types of speaker arrays including the speaker array 100 according to the present embodiment. It is the figure which plotted angle) for every frequency.
Specifically, graphs L1, L2, and L3 in FIG. 6 are plots of the main lobe width of the speaker array 100 according to the present embodiment for each frequency, and the graph L1 shows φs = 20 ° and a non-physical region. When the amplitude of the ripple is set to 1, the graph L2 is φs = 20 ° and the amplitude of the ripple in the non-physical region is set to 2. When the amplitude of the ripple in the non-physical region is set to 2, the graph L3 is the amplitude of the ripple in the non-physical region and φs = 40 ° Is a graph in which the main lobe width of the speaker array 100 is plotted for each frequency.
A graph L4 in FIG. 6 is a graph plotting the main lobe width in the case of rectangular in-phase driving, and the graph L5 is a plot of the main lobe width of the speaker array designed by the design method disclosed in Non-Patent Document 2. It is a graph.
In the graphs L6, L7, and L8 in FIG. 6, the amplitude characteristics in which the bandwidth in the normalization time frequency direction of the ripple in the non-physical region is substantially constant are set in the two-dimensional digital filter. The graph L6 plots the main lobe width of the speaker array for each frequency, and the graph L6 shows that when φs = 40 ° and the amplitude of the ripple in the non-physical region is set to 1, the graph L7 shows φs = 20 ° and When the ripple amplitude of the non-physical region is set to 2, the graph L8 is a graph plotting the main lobe width for each frequency when φs = 20 ° and the ripple amplitude of the non-physical region is set to 1.

As apparent from FIG. 6, according to the speaker array 100 according to the present embodiment, a conventional speaker array such as a speaker array related to rectangular in-phase driving or a speaker array related to the design method disclosed in Non-Patent Document 2 is used. Compared to a speaker array in which the amplitude characteristics in which the bandwidth in the frequency direction of the ripple in the non-physical region and the ripple in the non-physical region is approximately constant are set in the two-dimensional digital filter It can be seen that the main lobe width can be reduced.
Further, as apparent from FIG. 6, in the speaker array 100 according to the present embodiment, for example, when a certain value (for example, 80 °) is defined as the main lobe width, the main lobe width is used. The lower limit of the frequency of the acoustic beam that can be output (that is, the lower end band f _L of the directional speaker array: see Non-Patent Document 2 for details) is reduced compared to the case where rectangular in-phase driving is performed. I know that.

  Specifically, when φs = 20 ° (that is, the main lobe width is 20 °) and the ripple amplitude of the non-physical region is set to 1 (FIG. 6: graph L1), the case where rectangular in-phase driving is performed The lower end of the band of the speaker array is reduced by 14.3%. When φs = 20 ° and the ripple amplitude of the non-physical region is set to 2 (FIG. 6: graph L2), it is reduced by 19.1%, and φs = 40 ° (that is, the main lobe width is 40 °) and the ripple amplitude of the non-physical region is set to 1 (FIG. 6: graph L3), the reduction is 40.58%. That is, according to the speaker array 100 according to the present embodiment, it is possible to reduce the lower end of the band (that is, to improve the directivity in the low sound range) as compared with the conventional rectangular in-phase driving speaker array. Yes.

As described above, in the speaker array 100 according to the present embodiment, amplitude characteristics as shown in FIG. 2 (that is, amplitude characteristics satisfying the following three requirements) are set in the two-dimensional digital filter. This makes it possible to improve the directivity of the speaker array in the low sound range without increasing the array length and to avoid an increase in the sidelobe level.
The above three requirements are:
When the frequency characteristics are expressed in a two-dimensional frequency plane,
(A) having a plurality of ripples in the stop band in the cross section in the spatial frequency direction;
(B) The amplitude of the ripple in the non-physical region among the plurality of ripples is larger than the amplitude of the ripple in the physical region;
(C) Of the ripples in the non-physical region, for the ripples in the region where the normalized spatial frequency is smaller than a predetermined value, the smaller the spatial frequency corresponding to the ripple, the smaller the bandwidth in the time frequency direction of the ripple. Being narrow,
It is.
Next, the design of a two-dimensional digital filter that satisfies the above three requirements (that is, calculation of filter coefficients to be set for each one-dimensional digital filter 120-i) will be described.

(A-3: Two-dimensional digital filter design)
In Non-Patent Document 2 described above, when the amplitude characteristics of a two-dimensional digital filter formed by a group of one-dimensional digital filters connected to each speaker are viewed on a two-dimensional frequency plane, the output of the speaker array is sufficient. It is disclosed that the frequency characteristic when observed at a distant observation point is an amplitude characteristic distributed on a straight line represented by the following Equation 1 in a two-dimensional frequency plane.
(Expression 1) f ₂ = f ₁ · D · sin (φ) / (c · T)
In Equation 1, f ₁ is a normalized time frequency, f ₂ is a normalized spatial frequency, D is a transducer interval, T is a time sampling period, and c is a sound velocity.

Therefore, the directivity characteristic of the speaker array at a certain non-standardized time frequency f is on a straight line defined by the standardized time frequency f ₁ = f · T corresponding to the non-standardized time frequency f in the two-dimensional frequency plane. It can be said that they are distributed according to the relationship represented by the following formula 2.
(Expression 2) φ = sin ⁻¹ {(f ₂ · c · T) / (f ₁ · D)}

That is, if the two-dimensional digital filter can be designed so that the desired directivity at each non-standardized time frequency f is distributed on the straight line f ₁ = f · T in the relationship shown by the above equation 2, A desired directivity characteristic can be obtained. Non-Patent Document 2, the normalized spatial frequency direction of the two-dimensional frequency plane as described above (i.e., f ₂ direction) side by side one-dimensional filter characteristic section of setting the target characteristic of the two-dimensional digital filter, its A method for obtaining FIR filter coefficients by applying a two-dimensional Fourier series approximation to a target characteristic is disclosed.

More specifically, Non-Patent Document 2 describes, as design conditions for a speaker array formed of (N ₂ +1) speakers, an acoustic beam center φ ₀ and beam end angles (φ _{s +} , φ _s− ) and A design procedure for a two-dimensional digital filter in the case where a size (amplitude) δ of an equal ripple side lobe is given is disclosed. In the following, φ ₀ = 0 °, φ _{s +} = φ _s , φ _s− = −φ _s (that is, the acoustic beam is symmetric with respect to its center (φ ₀ = 0 °) axis). To do.

In the design procedure disclosed in Non-Patent Document 2, first, as shown in FIG. 7B, a two-dimensional frequency plane is represented by M ₁ (this embodiment) in the range of f ₁ = −0.5 to 0.5. in, _{M 1} is assumed to be an even number) is divided, the Dolph-Chebyshev characteristic cross-section at each frequency _{f 1 = k 1 / M 1} (k 1 = -M 1 / 2~M 1/2 integer) Design and target fan filter characteristics by juxtaposing them in the following procedure.

Specifically, first, the characteristics of a Dolph-Chebyshev filter of order N ₂ and stop band ripple size δ are designed, and the stop band end frequency f _st is a straight line with φ = φ _s (ie, f ₂ = f ₁ · D · sin (φ _s ) / (c · T)), the frequency f ₁ is obtained. Next, at the cross-sectional position of f ₁ ≧ f _l, the cross-sectional characteristics at f ₁ = f _l are shown in FIG. 7A so that the stop band edge is positioned on the straight line of φ = φ _s. Arranged in _two directions.
On the other hand, for the cross-sectional position of f ₁ <f ₁ , as shown in FIG. 8A, a Dolph-Chebyshev filter in which the stop band ripple is gradually increased from δ to a predetermined allowable value δ _L as shown in FIG. Put the characteristic (f2 = −0.5 to 0.5). However, the stop band ripple is determined so that the stop band end f _st is located on the straight line φ = φ _s in any cross section. Then, as shown in FIG. 8 (b), the value of the frequency _{f 1} the characteristics of the stop band ripple [delta] _L is first placed as _{f u,} the cross section of the f 1 _{<f u} in _f 1 = _{f u} Put the same property as the property. Note that f _L in FIG. 8B is the lower end of the band of the speaker array, and is a value determined by the following Equation 3.
(Expression 3) f _L = c · T · f _c / sin (φ _s )
However, f _c in Equation 3, which is half the frequency characteristic of the stop band ripple [delta] _L shown in FIG. 8 (a).
Thereafter, a filter coefficient to be set for each one-dimensional digital filter is calculated by performing a two-dimensional inverse discrete Fourier transform on the target amplitude characteristic of the fan filter set in this way.

In contrast, in the two-dimensional digital filter design of the speaker array 100 according to this embodiment, the M ₁ split two-dimensional frequency plane, as shown in FIG. _{9 (b), f 1 ≧} f l in the stopband placing the small one-dimensional filter ripples across the while keep the cross-sectional characteristics, the one-dimensional filter of large ripple f 1 _{<f l} in the non-physical area only section. Two amplitude characteristics shown in FIG. 9A are amplitude characteristics of a one-dimensional filter placed on each cross section. As can be seen from the comparison of the two amplitude characteristics, by increasing the ripple in the non-physical region, the frequency range occupied by the ripple is widened, and conversely the passband is narrowed.
Therefore, in the design of the two-dimensional digital filter according to the present embodiment, the ripple amplitude in the non-physical region is placed at the cross-sectional position of the time frequency in the lower sound range until the ripple amplitude reaches the predetermined maximum value, If it is reached, the design is repeated so that the non-physical region is filled with a ripple having a predetermined maximum amplitude (or an amplitude smaller than that) at each f ₁ of a lower frequency, and the cross-sectional characteristics at each f ₁ Set as. Such a design _is in a region where the f 1 _{<f u} is realized by a value greater than _f 2 _{(= f st)} to _{f 1 · D · sin (φ} ) / (c · T) .

  In this embodiment, in order to design a one-dimensional filter having two-step stopband equiripple characteristics as shown in FIG. 9A, a program for designing a filter according to the Parks & McClellan equiripple filter design algorithm is used. . Here, the equiripple filter design algorithm of Parks & McClellan is an algorithm for designing a filter so that a desired frequency response and an actual frequency response are optimized by using a Remez exchange algorithm and a weighted Chebyshev approximation theory. A filter designed according to this algorithm is sometimes referred to as a minimax filter because it is optimal in minimizing the maximum error between the desired frequency response and the actual frequency response. A filter designed according to this algorithm is also known as an equiripple filter because it exhibits equiripple in its frequency response. In this embodiment, the case where the Parks & McClellan equiripple filter design algorithm is used for the design of a one-dimensional filter with two-step stopband equiripple characteristics will be described. However, other FIR filter design algorithms may be used. Of course.

  FIG. 10 is a diagram showing parameters given to the program and characteristics of the design results. As shown in FIG. 10, in the present embodiment, three approximate bands (pass band, stop band 1, stop band 2) are set, and the target amplitude (1 for each approximate band) is set as a parameter for defining each approximate band. , 0, 0), error ripple (respectively δ1≈0, δ2 = δ, δ3 = δn) and weight (respectively w1, w2, w3), and repeated approximation under the condition of δ1w1 = δ2w2 = δ3w3 By defining the filter coefficient, a one-dimensional filter is designed.

FIG. 11 is a diagram in which a one-dimensional filter designed in accordance with Parks &McClellan's equiripple filter design algorithm and a design example of a one-dimensional filter having a Dolph-Chebyshev characteristic are shown. As is apparent from FIG. 11, in the former one-dimensional filter, it is understood that the width of the pass band is narrower than that of the latter by increasing the ripple in the stop band 2. Thus, the effect of narrowing the width of the pass band becomes more remarkable as the number of ripples in the non-physical area occupies the total number of ripples in the stop area. Theoretically, for ripples in the non-physical region, it is possible to set the amplitude as much as possible, but in practice it is necessary to set the upper limit appropriately, for example, as the upper limit “1” (that is, a value equal to the amplitude of the pass band), “2” (a value twice the amplitude of the pass band), or the like may be set.
By performing a two-dimensional inverse discrete Fourier transform on the target amplitude characteristic of the two-dimensional digital filter designed in this way, a filter coefficient to be set for each of the one-dimensional digital filters forming the two-dimensional digital filter is calculated. Is done. By setting the filter coefficient thus calculated in each one-dimensional digital filter 120-i, the amplitude characteristics shown in FIG. 2 are given to the two-dimensional digital filter formed by the one-dimensional digital filter. Become.

(A-4: Effects of the first embodiment)
As described above, the characteristics in the physical region directly affect the directivity, while the characteristics in the non-physical region do not directly affect the directivity. Therefore, in the speaker array 100 according to the present embodiment, By using a one-dimensional filter having a two-stage stopband ripple characteristic, the width of the main lobe can be reduced as the final filter coefficient characteristic while keeping the sidelobe level low particularly in the low band. Is possible.

In addition, according to the present embodiment, by adjusting the one-dimensional filter optimally according to f ₁ , the width of the main lobe is kept constant while keeping the influence of the side lobe low even in a band lower than the conventional band. Is possible. As described above, since the width of the main lobe depends on the number of ripples in the non-physical region and the amplitude thereof, the amplitude set for the ripple in the non-physical region so that the necessary directivity can be obtained according to f _1. By adjusting the number, it becomes possible to make the width of the main lobe constant in a lower band than in the prior art.

Moreover, a relatively high region time-frequency (e.g., the region in the non-patent document 2 defined by f _l ≦ f ₁₎ for, without increasing the amplitude of the ripples in the non-physical area the width of the main lobe sufficient Since it can be narrowed, for example, the Dolph-Chebyshev characteristic disclosed in Non-Patent Document 2 may be used in place of the ripple characteristic of two stages of the stop band. In addition, for such a region, if the width of the main lobe is set so as not to depend on the time frequency as disclosed in Non-Patent Document 2, the conventional method is combined with the improvement in characteristics in the low region according to the present embodiment. It becomes possible to obtain directivity characteristics that do not depend on time frequency in a wider band.

(B. Second Embodiment)
Next, a microphone array 200 according to the second embodiment of the present invention will be described.
FIG. 12 is a diagram illustrating a configuration example of a microphone array 200 according to the second embodiment of the present invention. As apparent from a comparison between FIG. 12 and FIG. 1, the configuration of the microphone array 200 is different from the configuration of the speaker array 100 in that the speaker 110-i (i: natural number of 1 to n) is replaced. The microphone 210-i (i: a natural number from 1 to n) that outputs an audio signal corresponding to the collected audio is provided.
In the microphone array 200, the audio signal output from the microphone 210-i is converted into audio data by an A / D converter (not shown) and input to the one-dimensional digital filter 120-i. Then, the filter processing described above is performed by each one-dimensional digital filter 120-i, and the filtered audio data output from each one-dimensional digital filter is added by an adder (not shown), and a sum as a result of the addition is added. A signal is output. In the second embodiment, it is needless to say that the A / D converter is not necessary when digital audio data is output from the microphone 210-i.

  Now, in the microphone array, the time-frequency characteristic of the plane wave arriving from the angle φ direction shown in FIG. 12 is one-dimensionally connected to each microphone (microphone 210-i in the present embodiment) constituting the microphone array. When the amplitude characteristics of the digital filter group are viewed on a two-dimensional frequency plane, it is generally known that the digital filter group is distributed on the straight line represented by Equation 2 described above. Therefore, by setting the filter coefficient described in the first embodiment to each of the one-dimensional digital filters 120-i, the directivity characteristics of the microphone array 200 are the same as those in the first embodiment. It is possible to obtain an effect (that is, an effect of improving the directivity in the low sound range of the microphone array and avoiding an increase in the sidelobe level without increasing the array length).

(C. Third Embodiment)
(C-1: Configuration)
FIG. 13 is a block diagram illustrating a configuration example of a speaker array 500 according to the third embodiment of the present invention. 13 has the same number of delay circuits 130-1, 130-2,... 130 as the speakers 110-1, 110-2,... 110-n, in addition to the structure of the speaker array 100 shown in FIG. -N, a user interface (hereinafter, “U / I”) unit 140, and a control unit 150. As shown in FIG. 13, in this embodiment, the array surface of the speaker array 500 is formed by linearly arranging the speakers 110-i (i: 1 to n) at a predetermined interval D. Hereinafter, the center position of the array surface is referred to as “C” (see FIG. 13). In FIG. 13, the same constituent elements as those in FIG. 1 are denoted by the same reference numerals, and the description thereof is omitted.

  Each of the one-dimensional digital filters 120-i in FIG. 13 has a frequency characteristic having a directional main axis in a direction perpendicular to the array plane passing through the point C shown in FIG. Filter coefficients given to 500 are set in advance. Each of these one-dimensional digital filters 120-i applies a filter process corresponding to the filter coefficient to the audio data delivered from the A / D converter, and a delay circuit corresponding to the one-dimensional digital filter 120-i. Output to 130-i.

The delay circuit 130-i delays the audio data input from the one-dimensional digital filter 120-i by a delay amount set by the control unit 150 to be described later, and outputs it. The delay circuit 130-i is for realizing steering of the directional spindle by giving the delay to the audio data input from the one-dimensional digital filter 120-i. Note that the steering of the directional spindle can be realized by applying such delay processing as described in “Kiyoshi Nishikawa,“ Broadband multi-beam forming method using delay processing and two-dimensional digital filter ”, IEICE Transactions, A Vol. .J87-A No.12 pp.1480-1490 December 2004 ”(hereinafter referred to as“ references ”).
Here, the delay amount in the delay circuit 130-i may be a delay amount in a sampling interval unit, or may be a delay amount in a smaller unit (ie, a unit less than the sampling interval). (A method for realizing a delay of a unit smaller than the sampling interval is described in detail in Non-Patent Document 1 and the above-mentioned reference). In the present embodiment, each delay circuit 130-i is configured with an FIR filter in order to realize delay processing that is not limited to the sampling interval unit. In the present embodiment, the case where the delay circuit 130-i is configured by an FIR filter will be described, but it may be configured by an IIR filter.
The audio data output from each of the delay circuits 130-i is converted into an audio signal by a D / A converter (not shown), and corresponds to the one-dimensional digital filter 120-i and the delay circuit 130-i. It is supplied to the speaker 110-i.
As a result, a sound corresponding to the sound signal supplied from the D / A converter is emitted from each of the speakers 110-i. In the present embodiment, the case where the speaker 110-i emits a sound corresponding to an analog audio signal will be described. However, the speaker 110-i emits a sound corresponding to digital audio data. Needless to say, it is not necessary to provide the D / A converter.

The U / I unit 140 includes, for example, a display unit configured by a liquid crystal panel and a drive circuit thereof, and an operation unit configured by a plurality of operators such as a numeric keypad.
The U / I unit 140 controls the steering angle of the directional main axis of the speaker array 500 within a predetermined angle range (FIG. 13: −φ _m degrees to φ _m degrees) under the control of the control unit 150. A screen for causing the user to designate the image is displayed on the display means, and data indicating the steering angle input by the user who has visually recognized the screen is operated by appropriately operating the operation element is delivered to the control unit 150. That is, the U / I unit 140 functions as a designation unit that allows the user to designate the steering angle of the pointing spindle.

The control unit 150 specifies a delay amount corresponding to the steering angle transmitted from the U / I unit 140 for each delay circuit 130-i and sets the delay amount to each delay circuit 130-i. More specifically, the control unit 150 includes a CPU (Central Processing Unit), a ROM (Read Only Memory), and a RAM (Random Access Memory) (all of which are not shown). A table and a control program are stored.
The delay amount management table, the predetermined angle range (i.e., -.phi _m degree to [phi] _m of the angular range) in association with each of the angle of the predetermined angle units in the speaker array on the steering angle direction A delay amount to be set in each of the delay circuits 130-i when steering the 500 pointing spindles is written. The delay amount stored in the delay amount management table is set so that the difference in delay amount between adjacent delay circuits becomes τ shown in the following formula 1. In the following equation 1, ρ = D / (c · T): D is the transducer interval, T is the time sampling period, c is the speed of sound, and φ ₀ is the steering angle.
(Equation 1) τ = ρ · sin (φ ₀ )
On the other hand, the control program stored in the ROM includes the data delivered from the U / I unit 140 and the content stored in the delay amount management table by causing the CPU to execute the control program using the RAM as a work area. And a process for setting the delay amount corresponding to the steering angle represented by the data to each of the delay circuits 130-i.
The above is the configuration of the speaker array 500.

As described above, the hardware configuration of the speaker array 500 according to the present embodiment is not different from the hardware configuration of the conventional speaker array.
However, in the speaker array 500 according to the present embodiment, each of the one-dimensional digital filters 120-i achieves frequency characteristics having a directional main axis in the direction where the steering angle is 0 degrees and is characteristic of the speaker array according to the present invention. Since a typical filter coefficient is set in advance, a characteristic amplitude characteristic of the speaker array according to the present invention is given to the two-dimensional digital filter formed by the one-dimensional digital filter. A characteristic directivity characteristic of the speaker array is realized.

Hereinafter, the amplitude characteristic of the two-dimensional digital filter formed by the one-dimensional digital filter 120-i and the directivity characteristic realized by the amplitude characteristic will be described with reference to the drawings. In the following, it is assumed that each of the speakers 110-i has an ideal characteristic (that is, a characteristic that the directivity characteristic does not depend on the frequency of the output sound). In the following, the speaker arrangement interval D = 0.068 [m], the sampling frequency f _s (= 1 / T) = 7451 [Hz], the number of FIR taps = 241, and n (the number of speakers) = 15. Shall.

(C-2: Amplitude characteristics and directivity characteristics of a two-dimensional digital filter)
FIG. 14 is a diagram illustrating the amplitude characteristics of the speaker array 500 and the directivity characteristics realized by the amplitude characteristics. 14A and 14B are realized by an amplitude characteristic and a directivity characteristic (that is, a two-dimensional digital filter) when the steering angle specified via the U / I unit 140 is 0 degree. 14 (c) and 14 (d) are diagrams showing the amplitude characteristic and the directivity when the steering angle is φ ₀ degrees.

More particularly, FIG. 14 (a), the main portion of the amplitude characteristic of the two-dimensional digital filter formed by the one-dimensional digital filters 120-i (specifically, the normalized time-frequency f ₁ is 0-0 in the range of .5, and a diagram normalized spatial frequency f ₂ is expressed in equal amplitude characteristic diagram range) of 0 to 0.5. The normalized time frequency is a value obtained by normalizing the time frequency by the reciprocal of the time sampling interval, and the normalized spatial frequency is obtained by normalizing the spatial frequency by the reciprocal of the speaker arrangement interval D. Value.

As is clear by referring FIG. 14 (a), a plurality in the speaker array 500 according to the present embodiment, the normalized time-frequency _{f 1} is lower region (e.g., region _{f 1} is 0 to 0.1) Ripple is provided. Among these plural ripples, a ripple in the non-physical area is given a large amplitude (in this embodiment, “1”), while a ripple in the physical area has an amplitude within the non-physical area. It is kept lower than the ripple. As apparent from FIG. 14A, the ripples in the non-physical region are equal ripples having substantially the same amplitude, and therefore the amplitude characteristics shown in FIG. Called a characteristic.

In addition, in the non-physical region, ripple is performed only in a region where the normalized spatial frequency f ₂ is equal to or greater than a predetermined value (0.25 in FIG. 14A) determined according to the maximum steering angle φ _m . Is provided. The reason is as follows.

  The width of the main lobe depends on the number of ripples in the non-physical region and the amplitude thereof (for example, the larger the number of ripples in the non-physical region in the total number of ripples in the stop region, the narrower the passband is. It is generally known that the width can be reduced (see Non-Patent Document 2). For this reason, it is possible to use a ripple as shown in FIG. 15A, for example, without imposing a restriction that a ripple is not provided in a region where the normalized spatial frequency value is less than a predetermined value. It is expected that the number of ripples in the physical area will increase and the main lobe width can be further reduced.

However, if a ripple is provided as shown in FIG. 15 (a) without imposing the above limitation, it is provided in the non-physical region in FIG. 15 (a) when steering of the pointing spindle is performed. As shown in FIG. 15C, the ripple may protrude into the physical area. This is explained in detail in the above-mentioned reference, but in the amplitude characteristic diagram shown in FIG. 15A, the characteristic that was on the straight line f ₂ = f ₁ · ρ · sin (φ) (non-patent document) 2) moves on the straight line f ₂ = f ₁ · ρ · (sin (φ) −sin (φ ₀ )) after steering with the steering angle φ ₀ , so that the physical area and the non-physical area a boundary straight line _{f 2} = ± _{f 1} · _ρ and the line _{f 2} = ripple was between _{± f 1 · ρ · (sin} (φ) + sin (φ 0)) is the sign of phi _{0, f 2} If either> 0 side or f ₂ <0 side, it comes from coming out in the physical area. When such an event occurs, the amplitude characteristics are disturbed in the 90-degree direction as shown in FIG.

In contrast, in the speaker array 500 according to this embodiment, since the normalized spatial frequency f ₂ of the non-physical area have limitations imposed of providing a ripple only area is the predetermined value or more, Even if steering of the directional spindle is performed, ripples in the non-physical region do not protrude into the physical region (see FIG. 14C), and the disturbance of the amplitude characteristic in the 90-degree direction that appears in FIG. It does not occur (see FIG. 14D). In the present embodiment, the non-physical area, normalized spatial frequency f ₂ is, a case has been described in imposing the constraint that providing ripple only the maximum steering angle to a predetermined value or more determined in accordance with the area, a main In other words, the ripples in the non-physical area may be provided only in the area that does not protrude into the physical area when steering is performed at the maximum steering angle.

As described above, in the speaker array 500 according to the present embodiment, the two-dimensional digital filter includes an amplitude characteristic as illustrated in FIG. 14A (that is, an amplitude characteristic that satisfies the following three requirements). By setting the filter coefficient to achieve the above, the directivity in the low frequency range of the speaker array is improved without increasing the array length, the increase in the side lobe level is avoided, and the amplitude characteristic is improved. It is possible to steer the directional spindle without causing disturbance. The above three requirements are as follows.
When the frequency characteristic is represented by a two-dimensional frequency plane, (a-1) having a plurality of ripples in the stop band in the cross section in the spatial frequency direction, and (a-2) in the non-physical region among the plurality of ripples. (A-3) Among the non-physical areas, when the steering of the directional spindle is performed at the maximum steering angle, a part of the ripples is larger than the amplitude of the ripples in the physical area. The ripple is formed only in an area that does not protrude into the physical area.
Next, the design of a two-dimensional digital filter that satisfies the above three requirements (that is, calculation of filter coefficients to be set for each one-dimensional digital filter 120-i) will be described.

(C-3: Design of two-dimensional digital filter)
In Non-Patent Document 2 described above, when the amplitude characteristics of a two-dimensional digital filter formed by a group of one-dimensional digital filters connected to each speaker are viewed on a two-dimensional frequency plane, the output of the speaker array is sufficient. It is disclosed that the frequency characteristic when observed at a distant observation point is an amplitude characteristic distributed on a straight line represented by the following formula 2 in a two-dimensional frequency plane.
(Expression 2) f ₂ = f ₁ · D · sin (φ) / (c · T)
In Equation 2, f ₁ is a normalized time frequency, f ₂ is a normalized spatial frequency, D is a transducer interval, T is a time sampling period, and c is a sound velocity.

Therefore, the directivity characteristic of the speaker array at a certain non-standardized time frequency f is on a straight line defined by the standardized time frequency f ₁ = f · T corresponding to the non-standardized time frequency f in the two-dimensional frequency plane. It can be said that they are distributed according to the relationship expressed by the following equation (3).
(Expression 3) φ = sin ⁻¹ {(f ₂ · c · T) / (f ₁ · D)}

That is, if the two-dimensional digital filter can be designed so that the desired directivity at each non-standardized time frequency f is distributed on the straight line f ₁ = f · T according to the relationship expressed by the above equation 3, A desired directivity characteristic can be obtained. Non-Patent Document 2, the normalized spatial frequency direction of the two-dimensional frequency plane as described above (i.e., f ₂ direction) side by side one-dimensional filter characteristic section of setting the target characteristic of the two-dimensional digital filter, its A method for obtaining FIR filter coefficients by applying a two-dimensional Fourier series approximation to a target characteristic is disclosed.

More specifically, Non-Patent Document 2 describes, as design conditions for a speaker array formed of (N ₂ +1) speakers, an acoustic beam center φ ₀ and beam end angles (φ _{s +} , φ _s− ) and A design procedure for a two-dimensional digital filter in the case where a size (amplitude) δ of an equal ripple side lobe is given is disclosed. In the following, φ ₀ = 0 °, φ _{s +} = φ _s , φ _s− = −φ _s (that is, the acoustic beam is symmetric with respect to its center (φ ₀ = 0 °) axis). To do.

In the design procedure disclosed in Non-Patent Document 2, first, as shown in FIG. 16B, a two-dimensional frequency plane is represented by M ₁ (this embodiment) in the range of f ₁ = −0.5 to 0.5. in, _{M 1} is assumed to be an even number) is divided, the Dolph-Chebyshev characteristic cross-section at each frequency _{f 1 = k 1 / M 1} (k 1 = -M 1 / 2~M 1/2 integer) Design and target fan filter characteristics by juxtaposing them in the following procedure.

Specifically, first, the characteristics of a Dolph-Chebyshev filter of order N ₂ and stop band ripple size δ are designed, and the stop band end frequency f _st is a straight line with φ = φ _s (ie, f ₂ = f ₁ · D · sin (φ _s ) / (c · T)), the frequency f ₁ is obtained. Next, at the cross-sectional position of f ₁ ≧ f _l, the cross-sectional characteristics at f ₁ = f ₁ are as follows, as shown in FIG. 16A, so that the stop band edge is positioned on the straight line of φ = φ _s. Arranged in _two directions.
On the other hand, for cross-sectional position of _{f 1 <f l, Dolph-} Chebyshev filter as shown in FIG. 17 (a), the stop band ripple, which increased gradually the amplitude from [delta] to a predetermined allowable value [delta] _L Put the characteristic (f2 = −0.5 to 0.5). However, the stop band ripple is determined so that the stop band end f _st is located on the straight line φ = φ _s in any cross section. Then, as shown in FIG. 17 (b), the value of the frequency _{f 1} the characteristics of the stop band ripple [delta] _L is first placed as _{f u,} the cross section of the f 1 _{<f u} in _f 1 = _{f u} Put the same property as the property. In addition, fL of _FIG.17 (b) is a band lower end of a speaker array, and is a value decided by the following formula 4.
(Expression 4) f _L = c · T · f _c / sin (φ _s )
However, f _c in Equation 4, which is half the frequency characteristic of the stop band ripple [delta] _L shown in FIG. 17 (a).
Thereafter, a filter coefficient to be set for each one-dimensional digital filter is calculated by performing a two-dimensional inverse discrete Fourier transform on the target amplitude characteristic of the fan filter set in this way.

On the other hand, in the design of the two-dimensional digital filter of the speaker array 500 according to the present embodiment, in order to satisfy the requirements (a-1) and (a-2) among the above three requirements, FIG. In the two-dimensional frequency plane divided into M ₁ as shown in FIG. ₁ , when f ₁ ≧ f _l , a one-dimensional filter having a small ripple is set as a cross-sectional characteristic over the entire stop band, while when f ₁ <f _l , only in the non-physical region A large ripple one-dimensional filter is placed on the cross section. Two amplitude characteristics shown in FIG. 18A are amplitude characteristics of a one-dimensional filter placed on each cross section. As can be seen from the comparison of the two amplitude characteristics, by increasing the ripple in the non-physical region, the frequency range occupied by the ripple is widened, and conversely the passband is narrowed.

  In this embodiment, in order to design a one-dimensional filter having two-step stopband equiripple characteristics as shown in FIG. 18A, a program for designing a filter according to the Parks & McClellan equiripple filter design algorithm is used. . Here, the equiripple filter design algorithm of Parks & McClellan is an algorithm for designing a filter so that a desired frequency response and an actual frequency response are optimized by using a Remez exchange algorithm and a weighted Chebyshev approximation theory. A filter designed according to this algorithm is sometimes referred to as a minimax filter because it is optimal in that it minimizes the maximum error between the desired frequency response and the actual frequency response. A filter designed according to this algorithm is also known as an equiripple filter because it exhibits equiripple in its frequency response. In this embodiment, the case where the Parks & McClellan equiripple filter design algorithm is used for the design of a one-dimensional filter having two-stage stopband equiripple characteristics will be described. However, other FIR filter design algorithms may be used. Of course.

  FIG. 19 is a diagram showing the parameters given to the program and the characteristics of the design results. As shown in FIG. 19, in this embodiment, three approximate bands (pass band, stop band 1, and stop band 2) are set, and the target amplitude (1 for each approximate band) is set as a parameter for defining each approximate band. , 0, 0), error ripple (respectively δ1≈0, δ2 = δ, δ3 = δn) and weight (respectively w1, w2, w3), and repeated approximation under the condition of δ1w1 = δ2w2 = δ3w3 By defining the filter coefficient, a one-dimensional filter is designed.

  FIG. 20 is a diagram in which a one-dimensional filter designed in accordance with Parks & McClellan's equiripple filter design algorithm and a design example of a one-dimensional filter having a Dolph-Chebyshev characteristic are shown. As is apparent from FIG. 20, in the former one-dimensional filter, it is understood that the width of the pass band is narrower than that of the latter because the ripple in the stop band 2 is increased. Thus, the effect of narrowing the width of the pass band becomes more remarkable as the number of ripples in the non-physical area occupies the total number of ripples in the stop area. Theoretically, for ripples in the non-physical region, it is possible to set the amplitude as much as possible, but in practice it is necessary to set the upper limit appropriately, for example, as the upper limit “1” (that is, a value equal to the amplitude of the pass band), “2” (a value twice the amplitude of the pass band), or the like may be set.

As described above, the filter coefficients that satisfy the requirements (a-1) and (a-2) (that is, the filter coefficients that realize the ripple characteristics of two stages of stopbands) are calculated. Furthermore, since it is necessary to satisfy the requirement (a-3), such a design is realized by adjusting (larger) f _st that is the condition of the stop band 1 for each f ₁ from a value determined by the specification. ing. The value of _fe is as specified, and the design is such that the smaller the value f1, the smaller the value.
By performing a two-dimensional inverse discrete Fourier transform on the target amplitude characteristic of the two-dimensional digital filter designed in this way, a filter coefficient to be set for each of the one-dimensional digital filters forming the two-dimensional digital filter is calculated. Is done. By setting the filter coefficients calculated in this way in each one-dimensional digital filter 120-i, the amplitude characteristics shown in FIG. 14A are given to the two-dimensional digital filter formed by these one-dimensional digital filters. Will be.

(C-4: Effects of the third embodiment)
As described above, the characteristics of the physical area directly affect the directivity, while the characteristics in the non-physical area do not directly affect the directivity. Therefore, in the speaker array 500 according to the present embodiment, By using a one-dimensional filter having a two-stage stopband ripple characteristic, the width of the main lobe can be reduced as the final filter coefficient characteristic while keeping the sidelobe level low particularly in the low band. Is possible.

  In addition, according to the present embodiment, since the amplitude characteristic that imposes the use restriction determined in accordance with the maximum steering angle in the non-physical region is set in the two-dimensional digital filter, the steering angle is smaller than the maximum steering angle. Even when the steering is performed, the amplitude characteristic is not disturbed in the direction of 90 degrees. That is, according to the speaker array 500 according to the present embodiment, it is possible to steer the pointing main shaft in any direction within an angle range determined according to the maximum steering angle.

Incidentally, a relatively high region time-frequency (e.g., the region in the non-patent document 2 defined by f _l ≦ f ₁₎ for the width of the main lobe without a large amplitude of ripples in the non-physical area sufficient Since it can be narrowed, for example, the Dolph-Chebyshev characteristic disclosed in Non-Patent Document 2 may be used in place of the ripple characteristic of two stages of the stop band. In addition, for such a region, if the width of the main lobe is set so as not to depend on the time frequency as disclosed in Non-Patent Document 2, the conventional method is combined with the improvement in characteristics in the low region according to the present embodiment. It becomes possible to obtain directivity characteristics that do not depend on time frequency in a wider band.

(D. Fourth Embodiment)
Next, a microphone array 200 according to a fourth embodiment of the present invention will be described.
FIG. 21 is a diagram illustrating a configuration example of a microphone array 200 according to the fourth embodiment of the present invention. As apparent from a comparison between FIG. 21 and FIG. 13, the configuration of the microphone array 200 is different from the configuration of the speaker array 500 in that the audio corresponding to the collected audio is used instead of the speaker 110-i. This is the point that a microphone 210-i that outputs a signal is provided.

  In the microphone array 200, the audio signal output from the microphone 210-i is converted into audio data by an A / D converter (not shown), and a delay corresponding to the steering angle is given by the delay circuit 130-i. To the one-dimensional digital filter 120-i. Then, the filter processing described above is performed by each one-dimensional digital filter 120-i, and the filtered audio data output from each one-dimensional digital filter is added by an adder (not shown), and a sum as a result of the addition is added. A signal is output. In the fourth embodiment as well, it goes without saying that the A / D converter is not necessary when digital audio data is output from the microphone 210-i.

  In the microphone array, the time-frequency characteristic of the plane wave arriving from the direction of the angle φ shown in FIG. 21 is one-dimensionally connected to each microphone (microphone 210-i in the present embodiment) constituting the microphone array. When the amplitude characteristics of the digital filter group are viewed on a two-dimensional frequency plane, it is generally known that the digital filter group is distributed on the straight line represented by Equation 3 described above.

  Therefore, the microphone array 200 is set by setting the filter coefficients described in the above-described third embodiment (that is, the filter coefficients satisfying the above-described three requirements) in each of the one-dimensional digital filters 120-i. As in the third embodiment described above, the directivity in the low sound range of the microphone array can be improved and the side lobe level can be prevented from increasing without increasing the array length. There are effects such as.

  In addition, in the microphone array 200 according to the present embodiment, by setting the delay amount corresponding to the steering angle input via the U / I unit 140 to each delay circuit 130-i, steering of the directional spindle is performed. Even if such steering is performed, the amplitude characteristic in the 90 degree direction is disturbed because the restriction of the requirement (a-3) described above is imposed on the amplitude characteristic of the two-dimensional digital filter. Will not occur.

(E. Fifth Embodiment)
Next, a speaker array 300 according to a fifth embodiment of the present invention will be described.
FIG. 22 is a diagram illustrating a configuration example of a speaker array 300 according to the fifth embodiment of the present invention. As apparent from a comparison between FIG. 22 and FIG. 13, the configuration of the speaker array 300 is different from the configuration of the speaker array 500 in that it has a delay circuit 130-i (i: a natural number of 1 to n). It is a point that has not been done.

In the speaker array 300, as the filter coefficient of the one-dimensional digital filter 120-i, the filter coefficient satisfying the above three requirements is set in advance as the filter coefficient when the steering angle is 0 degree. Then, when a steering angle is input via the U / I unit 140, the control unit 150 calculates a delay amount corresponding to the steering angle, and satisfies the following three requirements according to the delay amount. And rewriting the filter coefficient of the one-dimensional digital filter 120-i with the result of calculation, the steering of the directional spindle is realized.
When the frequency characteristic is represented by a two-dimensional frequency plane, (b-1) a passband is formed so as to form a main lobe in the designated steering angle direction, and the boundary between the physical region and the non-physical region, Forming a first stop zone with the passband, while forming a second stop zone in the non-physical region; (b-2) over the first and second stop zones; (B-3) the amplitude of the ripple in the second stop band is larger than the amplitude of the ripple in the first stop band, and within the first stop band and the second stop band. Each of which is equiripple.
A filter coefficient calculation method that satisfies the above three requirements will be described in detail later.

  As described above, the speaker array 300 according to the fifth embodiment does not include the delay circuit 130-i that gives a delay corresponding to the steering angle to the audio signal supplied to each of the speakers 110-i. Although different from the array 500, the one-dimensional digital filter 120-i of the speaker array 300 has both the function of the one-dimensional digital filter of the speaker array 500 and the function of the delay circuit 130-i. 300 can achieve the same operational effects as the speaker array 500.

  The above is the configuration of the speaker array 300 according to the fifth exemplary embodiment of the present invention. Of course, the microphone array 200 may be configured by replacing each speaker 110-i of the speaker array 300 with the microphone 210-i as in the microphone array 200 for the speaker array 500 described above. According to the microphone array configured as described above, the same operational effects as those of the microphone array 200 described above can be obtained.

(F. Sixth Embodiment)
Next, a speaker array 400 according to a sixth embodiment of the present invention will be described.
FIG. 23 is a diagram illustrating a configuration example of a speaker array 400 according to the sixth embodiment of the present invention. As apparent from a comparison between FIG. 23 and FIG. 22, the configuration of the speaker array 400 is different from the configuration of the speaker array 300 in that the storage unit 160 is included.

  The storage unit 160 is, for example, a hard disk, and is associated with each of a plurality of predetermined angle units within a predetermined steering range, and the one-dimensional digital filter 120-is used when the directional spindle is steered to that angle. Stored are filter coefficients to be set for each of i (filter coefficients that satisfy the requirements (b-1), (b-2), and (b-3) described above). That is, the storage unit 160 stores filter coefficients designed in accordance with specifications in which the directional main shaft is steered at each angle in association with each of a plurality of predetermined angles within a predetermined steering range.

  In the speaker array 400, when the steering angle is input via the U / I unit 140, the control unit 150 reads out the filter coefficient corresponding to the steering angle from the storage unit 160, and the one-dimensional digital filter 120-. Set to each of i. As a result, steering of the directional spindle is realized.

Next, a design method for filter coefficients stored in the storage unit 160 will be described.
In the design of the filter coefficient of the speaker array 400 according to the present embodiment, a steered main lobe is formed while using the non-physical region along a line of φ = ± 90 degrees that is a boundary line between the non-physical region and the physical region. By doing so, the spread of directivity due to the use limitation of the non-physical area is avoided.
Specifically, as shown in FIG. 24, first, five approximate bands (stop band 2 ′, stop band 1 ′, pass band, stop band 1, and stop band 2) are set. Each approximate band is defined by the following three types of parameters as shown in FIG. That is, the target amplitude (each of 0, 0, 1, 0, 0) and the error ripple (δ ₃ = δ _n , δ ₂ = δ, δ ₁ ≈0, δ ₂ = δ, δ ₃ = δ, respectively, of each approximate band _n ) and weights (w ₃ , w ₂ , w ₁ , w ₂ , w ₃ , respectively). Then, for each approximate band, approximation by a predetermined filter coefficient design algorithm is repeatedly executed under the condition that the product of the error ripple and the weight is constant (that is, the condition of δ ₁ w ₁ = δ ₂ w ₂ = δ ₃ w ₃ ). Thus, the one-dimensional filter is designed by determining the filter coefficient. As the filter coefficient design algorithm, the Parks & McClellan equiripple filter design algorithm described above is used.

In the present embodiment, the design is such that the non-physical region is filled with a ripple having a specified maximum amplitude or less at each f _{1 at} a low frequency, and is set as a cross-sectional characteristic at each f ₁ . The design is such that the increase in the main lobe width at a low frequency is reduced. Specifically, such a design is realized by appropriately adjusting (increasing) the values of fst and fst ′ which are the conditions of the stop band 1 and the stop band 1 ′ for each f ₁ from the values determined by the specifications. doing. (Fe and fe 'are the values as specified, and the design decreases as f1 decreases).

  FIG. 26 is a diagram showing the amplitude characteristics realized by the filter coefficients designed as described above on a two-dimensional frequency plane. FIG. 27 shows a part of the amplitude characteristics shown in FIG. FIG. 6 is an equiamplitude characteristic diagram showing a range in which the normalized time frequency f1 is −0.5 to 0.5 and the normalized spatial frequency f2 is −0.5 to 0.5. FIG. 28 is a diagram showing the amplitude characteristics shown in FIG. 26 as directivity characteristics at several frequencies (371.09473 Hz, 745.88264 Hz, 1491.6553 Hz, 22333.8447 Hz and 3354.4053 Hz). As is apparent from FIG. 28, it can be seen that the main axis of steering is steered in the range of 330 to 340 degrees (that is, in the direction of −30 to −20 degrees). Since the frequency at which the grating is generated by the steering is lowered, the grating lobe appears in the directivity shown in FIG. 28. However, it is usually necessary to use only the band where the grating does not occur. Will not occur.

  The above is the configuration of the speaker array 400 according to the sixth embodiment of the present invention. In addition, like the microphone array 200 with respect to the speaker array 500 described above, a microphone array capable of steering a directional main axis in an arbitrary direction is configured by replacing each speaker 110-i of the speaker array 300 with a microphone 210-i. Of course, it is good. According to the microphone array configured as described above, the directional main axis can be steered in an arbitrary direction. Of course, like the microphone array 200 described above, the speaker array is not lengthened. In addition, it is possible to improve the directivity of the microphone array in the low sound range and to avoid an increase in the sidelobe level.

(C. deformation)
Although the embodiment of the present invention has been described above, it is needless to say that the embodiment described above may be modified as described below.
(1) In the above-described embodiment, the case where an acoustic beam symmetric with respect to the central axis of the passband is described, but an asymmetric acoustic beam with respect to the symmetric axis can also be formed.

(2) In the above-described embodiment, the case where the speaker 110-i and the microphone 210-i have ideal characteristics has been described. However, since transducers such as speakers and microphones generally have directivity characteristics according to frequency, amplitude characteristics to be applied to the two-dimensional digital filter (that is, set to each one-dimensional digital filter 120-i). The power filter coefficient) may be determined in consideration of the directivity characteristics for each frequency of the transducer. This can be achieved by applying a method similar to the method disclosed in “Kiyoshi Nishikawa, Takaya Osaki“ Directive Array Speaker Using Two-Dimensional Digital Filter ”(1995)”, for example.

(3) In the above-described embodiment, an equiripple having an amplitude equal to or larger than the amplitude of the passband is provided for the nonphysical region in the stopband, and the physical region A ripple characteristic such as a two-step stop band in which a ripple having an amplitude smaller than a ripple (“δ = 0.1” in the above-described embodiment) is provided, and the ripple in the non-physical area is a space corresponding to the ripple. The case has been described where the amplitude characteristic in which the bandwidth in the time frequency direction of the ripple becomes narrower as the frequency is smaller is given to the two-dimensional digital filter. However, the ripples in the non-physical area are not necessarily equal ripples. For example, as shown in FIG. 29, a ripple having a larger amplitude than the passband in a stopband in the non-physical region (stopband 2 in FIG. 29), an amplitude smaller than the ripple, and within the physical region The stop band multistage ripple characteristic may be provided with a ripple having a larger amplitude than the ripple in the stop band (stop band 1 in FIG. 29). In short, the frequency characteristics of the two-dimensional digital filter of the speaker array or the microphone array according to the present invention are that a plurality of ripples are provided in the stop band, and the amplitude of the ripple in the non-physical region is larger than the amplitude of the ripple in the physical region. The ripple in the non-physical region has only to have a frequency characteristic in which the bandwidth in the time frequency direction of the ripple is narrower as the spatial frequency corresponding to the ripple is smaller.

(4) In the above-described embodiment, a case has been described in which filter coefficients characteristic of the speaker array according to the present invention are preset in each one-dimensional digital filter forming the two-dimensional digital filter. The filter coefficient may be sequentially calculated and set each time such a speaker array or microphone array is used. In this way, for example, when the speaker array or the microphone array according to the present invention is installed and used in an acoustic space such as a concert hall, the directivity characteristics according to the acoustic characteristics of the acoustic space such as the size and shape of the acoustic space. Can be set appropriately.
Further, the filter coefficient set for each one-dimensional digital filter may be given from the outside of the speaker array or the microphone array. Specifically, a filter that sets a communication means such as a NIC (Network Interface Card) in the speaker array or the microphone array and a filter coefficient acquired via the communication network using the communication means in each one-dimensional digital filter. Coefficient setting means may be provided, and for example, reading means for reading data from a computer-readable recording medium such as a CD-ROM (Compact Disk-Read Only Memory) is provided instead of the communication means. Of course, the filter coefficient may be written and distributed on the recording medium, and the filter coefficient read by the reading means may be set in each one-dimensional digital filter by the filter coefficient setting means.

(5) In the above-described embodiment, the aspect in which the acoustic beam that is symmetric with respect to the central axis of the pass band is formed when the steering angle is 0 degree has been described. However, the acoustic beam that is asymmetric with respect to the central axis is formed. It is also possible to do.

(6) In the embodiment described above, an equiripple having an amplitude equal to or greater than the amplitude of the passband is provided in the non-physical region of the stopband, and the physical region includes a non-physical region. This is ripple characteristics such as a two-step stop band in which a ripple having an amplitude smaller than ripple (“δ = 0.1” in the above-described embodiment) is provided, and further, a restriction according to the steering angle (the requirement (a) -3) or (b-1)) has been described as being placed in a non-physical area.
However, the ripples in the non-physical area are not necessarily equal ripples. For example, as shown in FIG. 29, a ripple having a larger amplitude than the passband in a stopband in the non-physical region (stopband 2 in FIG. 29), an amplitude smaller than the ripple, and within the physical region The stop band multistage ripple characteristic may be provided with a ripple having a larger amplitude than the ripple in the stop band (stop band 1 in FIG. 29). In short, the frequency characteristics of the two-dimensional digital filter of the speaker array or the microphone array according to the present invention are that a plurality of ripples are provided in the stop band, and the amplitude of the ripple in the non-physical region is larger than the amplitude of the ripple in the physical region. Any frequency characteristic may be used as long as it is large and the ripples in the non-physical region are arranged so that they do not protrude into the physical region when steering the pointing spindle.

(7) In the above-described embodiment, a case has been described in which filter coefficients characteristic of the speaker array according to the present invention are set in advance in each one-dimensional digital filter forming the two-dimensional digital filter. The filter coefficient may be sequentially calculated and set each time such a speaker array or microphone array is used. In this way, for example, when the speaker array or the microphone array according to the present invention is installed and used in an acoustic space such as a concert hall, the directivity characteristics according to the acoustic characteristics of the acoustic space such as the size and shape of the acoustic space. Can be set appropriately.
Further, the filter coefficient set for each one-dimensional digital filter may be given from the outside of the speaker array or the microphone array. Specifically, a filter that sets a communication means such as a NIC (Network Interface Card) in the speaker array or the microphone array and a filter coefficient acquired via the communication network using the communication means in each one-dimensional digital filter. Coefficient setting means may be provided, and for example, reading means for reading data from a computer-readable recording medium such as a CD-ROM (Compact Disk-Read Only Memory) is provided instead of the communication means. Of course, the filter coefficient may be written and distributed on the recording medium, and the filter coefficient read by the reading means may be set in each one-dimensional digital filter by the filter coefficient setting means.

  According to the present invention, it is possible to improve the directivity of the speaker array and the microphone array in the low sound range without increasing the array length, and to avoid an increase in the level of the side lobe. The main shaft can be steered.

Claims (6)

A plurality of speakers arranged in a straight line at a predetermined interval, and provided corresponding to each of the plurality of speakers, a predetermined filter coefficient is set in advance, and the input voice data includes the filter coefficient. A one-dimensional digital filter that performs a corresponding filtering process and outputs the input audio data to each one-dimensional digital filter, while corresponding to the audio data output from each one-dimensional digital filter In the speaker array for supplying to the speaker and outputting sound corresponding to the sound data,
The filter coefficient set for each one-dimensional digital filter is:
When the frequency characteristics of the two-dimensional digital filter formed by each of the one-dimensional digital filters are represented on a two-dimensional frequency plane, the cross section in the spatial frequency direction has a plurality of ripples in the stop band, and the plurality Of the ripples in the non-physical region among the ripples in the non-physical region, the ripples in the region of the non-physical region where the spatial frequency is less than a predetermined value among the ripples in the non-physical region A loudspeaker array characterized by a filter coefficient that gives the two-dimensional digital filter an amplitude characteristic in which the smaller the spatial frequency corresponding to the ripple is, the narrower the bandwidth in the time frequency direction of the ripple is. A plurality of microphones arranged in a straight line at a predetermined interval and a plurality of microphones are provided corresponding to each of the plurality of microphones, and a predetermined filter coefficient is set in advance. A one-dimensional digital filter that performs a corresponding filtering process and outputs the audio data output from each of the plurality of microphones to the corresponding one-dimensional digital filter. In the microphone array that outputs the sum signal of the audio data output from
The filter coefficient set for each one-dimensional digital filter is:
When the frequency characteristics of the two-dimensional digital filter formed by each of the one-dimensional digital filters are represented on a two-dimensional frequency plane, the cross section in the spatial frequency direction has a plurality of ripples in the stop band, and the plurality Of the ripples in the non-physical region among the ripples in the non-physical region, the ripples in the region of the non-physical region where the spatial frequency is less than a predetermined value among the ripples in the non-physical region A microphone array having filter coefficients that give the two-dimensional digital filter amplitude characteristics such that the smaller the spatial frequency corresponding to ripple is, the narrower the bandwidth in the time frequency direction of the ripple is. A plurality of speakers arranged linearly at a predetermined interval, and a one-dimensional digital filter provided corresponding to each of the plurality of speakers, which performs a predetermined filter process on the input audio data and outputs it. And supplying the input audio data to each one-dimensional digital filter, while supplying the audio data output from each one-dimensional digital filter to the corresponding speaker to output the sound corresponding to the audio signal. In the speaker array,
A designation means for designating a steering angle of the directional spindle within a predetermined angle range;
A specifying unit that specifies, for each speaker, a delay amount to be added to the sound output from each of the plurality of speakers according to the steering angle specified by the specifying unit;
Delay means for giving a delay corresponding to the delay amount specified for each speaker by the specifying means to audio data output from each of the plurality of one-dimensional digital filters;
With
The filter coefficient of each one-dimensional digital filter is:
When the frequency characteristic of the two-dimensional digital filter formed by each one-dimensional digital filter is represented by a two-dimensional frequency plane,
(1) having a plurality of ripples in the stop band in the cross section in the spatial frequency direction;
(2) The amplitude of the ripple in the non-physical region among the plurality of ripples is larger than the amplitude of the ripple in the physical region;
(3) Of the non-physical region, forming a ripple only in a region where a part of the ripple does not protrude into the physical region when steering of the directional spindle is performed at the maximum steering angle within the angle range,
Speaker array. A plurality of speakers arranged linearly at a predetermined interval, and a one-dimensional digital filter provided corresponding to each of the plurality of speakers, which performs a predetermined filter process on the input audio data and outputs it. And supplying the input audio data to each one-dimensional digital filter, while supplying the audio data output from each one-dimensional digital filter to the corresponding speaker to output the sound corresponding to the audio signal. In the speaker array,
A designation means for designating a steering angle of the directional spindle within a predetermined angle range;
With
The filter coefficient of each one-dimensional digital filter is:
When the frequency characteristic of the two-dimensional digital filter formed by each one-dimensional digital filter is represented by a two-dimensional frequency plane,
(1) A passband is formed so as to form a main lobe in the steering angle direction designated by the designation means, and a first stopband is formed between the boundary between the physical region and the non-physical region and the passband. Forming a second stop zone in the non-physical region,
(2) having a plurality of ripples in the stop band across the first and second stop areas;
(3) the ripple amplitude of the second stop band is larger than the ripple amplitude of the first stop band;
Speaker array. A plurality of microphones arranged linearly at a predetermined interval, and a one-dimensional digital filter provided corresponding to each of the plurality of microphones, which performs a predetermined filter process on the input audio data and outputs it. A microphone array for supplying audio data output from each of the plurality of microphones to the corresponding one-dimensional digital filter, and outputting a sum signal of the audio data output from each of the one-dimensional digital filters,
A designation means for designating a steering angle of the directional spindle within a predetermined angle range;
A specifying unit that specifies, for each microphone, a delay amount to be added to audio data output from each of the plurality of microphones according to the steering angle specified by the specifying unit;
A delay unit that adds a delay corresponding to the delay amount specified for each microphone by the specifying unit to the audio data output from each microphone and inputs the audio data to the corresponding one-dimensional digital filter;
With
The filter coefficient of each one-dimensional digital filter is:
When the frequency characteristic of the two-dimensional digital filter formed by each one-dimensional digital filter is represented by a two-dimensional frequency plane,
(1) having a plurality of ripples in the stop band in the cross section in the spatial frequency direction;
(2) The amplitude of the ripple in the non-physical region among the plurality of ripples is larger than the amplitude of the ripple in the physical region;
(3) Of the non-physical region, forming a ripple only in a region where a part of the ripple does not protrude into the physical region when steering of the directional spindle is performed at the maximum steering angle within the angle range,
A microphone array characterized by A plurality of microphones arranged linearly at a predetermined interval, and a one-dimensional digital filter provided corresponding to each of the plurality of microphones, which performs a predetermined filter process on the input audio data and outputs it. A microphone array for supplying audio data output from each of the plurality of microphones to the corresponding one-dimensional digital filter, and outputting a sum signal of the audio data output from each of the one-dimensional digital filters,
A designation means for designating a steering angle of the directional spindle within a predetermined angle range;
With
The filter coefficient of each one-dimensional digital filter is:
When the frequency characteristic of the two-dimensional digital filter formed by each one-dimensional digital filter is represented by a two-dimensional frequency plane,
(1) A passband is formed so as to form a main lobe in the steering angle direction designated by the designation means, and a first stopband is formed between the boundary between the physical region and the non-physical region and the passband. Forming a second stop zone in the non-physical region,
(2) having a plurality of ripples in the stop band across the first and second stop areas;
(3) the ripple amplitude of the second stop band is larger than the ripple amplitude of the first stop band;
A microphone array characterized by

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