WO2008041530A1 - Réseau de haut-parleurs et réseau de microphones - Google Patents

Réseau de haut-parleurs et réseau de microphones Download PDF

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Publication number
WO2008041530A1
WO2008041530A1 PCT/JP2007/068452 JP2007068452W WO2008041530A1 WO 2008041530 A1 WO2008041530 A1 WO 2008041530A1 JP 2007068452 W JP2007068452 W JP 2007068452W WO 2008041530 A1 WO2008041530 A1 WO 2008041530A1
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WIPO (PCT)
Prior art keywords
dimensional digital
ripple
digital filter
filter
amplitude
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PCT/JP2007/068452
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English (en)
Japanese (ja)
Inventor
Kiyoshi Nishikawa
Takashi Kushida
Original Assignee
National University Corporation Kanazawa University
Yamaha Corporation
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Application filed by National University Corporation Kanazawa University, Yamaha Corporation filed Critical National University Corporation Kanazawa University
Priority to JP2008537462A priority Critical patent/JP5151985B2/ja
Publication of WO2008041530A1 publication Critical patent/WO2008041530A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/403—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers loud-speakers

Definitions

  • the present invention relates to a technique for improving the directivity of a speaker array and a microphone array, and particularly to a technique for improving directivity in a low sound range.
  • a sound field is formed only in a specific direction by using a speaker array or a microphone array formed by arranging a plurality of transducers such as speakers and microphones in a straight line at a predetermined interval.
  • Technology that picks up only audio is generally popular.
  • the same directivity characteristic can be realized over a wide band from a high sound range to a low sound range.
  • 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 (non- In order to ensure sufficient directivity in the low frequency range, there is a problem that the device size of the spin force array and the microphone array becomes large.
  • Non-Patent Document 2 describes the amplitude characteristics of a digital filter connected to each transducer constituting a speaker array and a microphone array! /, And a cross section in the spatial frequency direction on a two-dimensional frequency plane.
  • the filter coefficient of each digital filter By setting the filter coefficient of each digital filter so that it becomes the amplitude characteristic (or its approximate characteristic) of the Dolph-Chebyshev filter that has a ripple characteristic such as a stop band, the band that can give the same directivity characteristic is obtained.
  • a technique for extending to the low frequency side is disclosed.
  • Non-Patent Document 1 Toshiro Oga, Yoshio Yamazaki, Yutaka Kaneda "Acoustic System and Digital Signal Processing" The Institute of Electronics, Information and Communication Engineers 1993-05 pl 76 ⁇ 186
  • Non-patent document 2 Yasushi Matsumoto, Kiyoshi Nishikawa "Installation of a directional array speaker with a constant sidelobe amount Measurement Method "IEICE Technical Report 2004— 74 pi 3 ⁇ ; 18
  • ripples with a ripple characteristic such as a stop band usually exist also in regions other than the non-physical region (the region 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 in order to improve the directivity in the sound range, there is a problem that the amplitude level of the side lobe, which is an originally unnecessary directivity characteristic, increases. Also, 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). It does not disclose the technology to be used.
  • the present invention has been made in view of the above problems, and improves the directivity in the low frequency range of the spin force array and the microphone array without increasing the array length, and reduces the amplitude level of the side mouth.
  • the purpose is to provide a technology that makes it possible to avoid the increase and steer the directional spindle.
  • the present invention provides a plurality of speakers arranged linearly at a predetermined interval, and is provided corresponding to each of the plurality of speakers, and a predetermined filter coefficient is preset. And a one-dimensional digital filter that 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. Filter coefficients set in the respective one-dimensional digital filters in the speaker array for supplying the audio data output from the respective one-dimensional digital filters to the corresponding speakers and outputting sounds corresponding to the audio data. Represents the frequency characteristics of the two-dimensional digital filter formed by each one-dimensional digital filter in a two-dimensional frequency plane.
  • the amplitude of the ripples in the non-physical region among the plurality of ripples is larger than the amplitude of the ripples in the physical region.
  • a speaker array characterized by a filter coefficient that gives the two-dimensional digital filter an amplitude characteristic in which the bandwidth in the time frequency direction of the ripple becomes narrower as the spatial frequency corresponding to the ripple becomes smaller To do.
  • the present invention provides a plurality of microphones arranged linearly at a predetermined interval, and a predetermined filter coefficient provided corresponding to each of the plurality of microphones.
  • a predetermined filter coefficient provided corresponding to each of the plurality of microphones.
  • the filter coefficient set for each one-dimensional digital filter is The frequency characteristics of the two-dimensional digital filter formed by each of the one-dimensional digital filters are expressed in a two-dimensional frequency plane.
  • the cross section in the spatial frequency direction has 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.
  • 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.
  • the smaller the spatial frequency corresponding to the ripple the narrower the bandwidth in the time frequency direction of the ripple is.
  • the present invention provides a plurality of speakers arranged linearly at a predetermined interval, and audio data input corresponding to each of the plurality of speakers.
  • a one-dimensional digital filter that outputs a predetermined filter process and supplies the input sound data to each one-dimensional digital filter, while the sound data output from each one-dimensional digital filter In the speaker array for supplying to the corresponding speaker and outputting the sound corresponding to the sound signal, a designation means for designating a steering angle of the directional spindle within a predetermined angle range, and a steering designated by the designation means
  • a specifying unit that specifies, for each speaker, a delay amount to be added to audio output from each of the plurality of speakers according to an angle.
  • the plurality of one-dimensional digital Delay means for adding to the audio data output from each of the Tal filters, and the filter coefficient of each one-dimensional digital filter is the frequency characteristic of the two-dimensional digital filter formed by each one-dimensional digital filter.
  • the cross section in the spatial frequency direction has a plurality of ripples in the stop band
  • the ripple amplitude in the non-physical region among the plurality of ripples is the physical region.
  • the non-physical regions when the steering of the directional spindle is performed at the maximum steering angle within the above-mentioned angular range, some of the ripples are in the physical region.
  • a speaker array characterized in that a ripple is formed only in a region that does not protrude.
  • the present invention provides a plurality of speakers arranged in a straight line at a predetermined interval, and input audio data provided corresponding to each of the plurality of speakers.
  • a one-dimensional digital filter that outputs a predetermined filter process and supplies the input sound data to each one-dimensional digital filter, while the sound data output from each one-dimensional digital filter
  • a speaker array for supplying to the corresponding speaker and outputting a sound corresponding to the sound signal, and a designating unit for designating a steering angle of a directional spindle within a predetermined angle range, and each of the one-dimensional digital
  • the filter coefficient of the filter is the frequency characteristic of the 2D digital filter formed by each 1D digital filter.
  • a pass band is formed so as to form a main lobe in the steering angle direction specified by the specifying means, and a second pass is formed between the boundary between the physical area and the non-physical area and the pass area.
  • a spin force array characterized in that the amplitude of the second stopband ripple is greater than the amplitude of the first stopband ripple.
  • the present invention provides a plurality of microphones arranged linearly at a predetermined interval, and audio data input corresponding to each of the plurality of microphones.
  • a one-dimensional digital filter that performs a predetermined filtering process and outputs the sound data output from each of the plurality of microphones to the corresponding one-dimensional digital filter, while each of the one-dimensional digital filters.
  • a designation unit that designates the steering angle of the directional spindle within a predetermined angle range, and the plurality of microphones according to the steering angle designated by the designation unit Specific means for specifying the delay amount to be added to the audio data output from each of the microphones, and a delay according to the delay amount specified for each microphone by the specifying means from each microphone.
  • Delay means for adding to the corresponding one-dimensional digital filter to be applied to the audio data to be output, and the filter coefficient of each one-dimensional digital filter is a two-dimensional formed by each one-dimensional digital filter (1) Spatial frequency direction when the frequency characteristics of a digital filter are represented in a two-dimensional frequency plane
  • the cross section has a plurality of ripples in the blocking area.
  • the amplitude of the ripple in the non-physical area is larger than the amplitude of the ripple in the physical area.
  • the ripple is formed only in a region where a part of the ripple does not protrude into the physical region when the steering spindle is steered at the maximum steering angle within the angle range.
  • a microphone array is provided.
  • 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 for outputting a sum signal of the audio data, and a designating unit for designating a steering angle of the directional spindle 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 (1)
  • a pass band is formed so as to form a main lobe in the steering angle direction designated by the designating means, and a passband is formed between the boundary between the physical area and the non-physical area and the passband.
  • a mouthphone array is provided.
  • the present invention it is possible to improve the directivity in the low sound range of the speaker array and the microphone array without increasing the array length, and to avoid an increase in the level of the side lobe.
  • the effect is that steering of the directional spindle becomes possible.
  • FIG. 1 is a block diagram showing an electrical configuration of a speaker array 100 according to a first embodiment of the present invention.
  • FIG. 2 is a diagram showing an example of amplitude characteristics of a two-dimensional digital filter of the speaker array 100 using a two-dimensional frequency plane.
  • FIG. 3 is a diagram showing a part of the same amplitude characteristic in an equiamplitude characteristic diagram.
  • FIG. 4 is a diagram in which the amplitude characteristics of the speaker array 100 are plotted for each predetermined angle.
  • FIG. 5 is a diagram in which the directivity characteristics of the speaker array 100 are plotted for each predetermined frequency.
  • FIG. 6 is a diagram showing the relationship between the frequency of the acoustic beam output from the speaker array 100 and the main lobe width of the acoustic beam.
  • FIG. 7 Explains the design method disclosed in Non-Patent Document 2 for cross-sectional characteristics at f ⁇ f.
  • FIG. 8 Explains the design method disclosed in Non-Patent Document 2 for cross-sectional characteristics when f ⁇ f.
  • FIG. 9 is a diagram for explaining a cross-sectional property design method according to the present embodiment.
  • FIG. 10 is a diagram showing characteristics of a design result of a one-dimensional filter by Parks & McClellan's equiripple filter design program.
  • FIG. L l A diagram showing a design example of a one-dimensional filter by the equiripple filter design program of Parks & McClellan and a design example of a one-dimensional filter having a Dolph-Chebyshev characteristic.
  • FIG. 12 is a diagram showing an electrical configuration of a microphone array 200 according to a second embodiment of the present invention.
  • FIG. 13 is a diagram showing an electrical configuration of a speaker array 500 according to the third embodiment of the invention.
  • FIG. 13 is a diagram showing an electrical configuration of a speaker array 500 according to the third embodiment of the invention.
  • FIG. 14 is a diagram showing amplitude characteristics and directivity characteristics of the speaker array 500.
  • FIG. 15 is a diagram showing amplitude characteristics and directivity characteristics of a loudspeaker array in which usage restrictions are not set in non-physical areas.
  • FIG. 16 Explains the design method disclosed in Non-Patent Document 2 for the cross-sectional characteristics when f ⁇ f.
  • FIG. 17 Explains the design method disclosed in Non-Patent Document 2 for cross-sectional characteristics when f ⁇ f.
  • FIG. 18 is a diagram for explaining a cross-sectional property design method according to the present embodiment.
  • FIG. 19 is a diagram showing the characteristics of the design result of a one-dimensional filter by Parks & McClellan's equiripple filter design program.
  • FIG. 20 is a diagram illustrating a design example of a one-dimensional filter by a Parks & McClellan equiripple filter design program and a design example of a one-dimensional filter having a Dolph-Chebyshev characteristic.
  • FIG. 21 is a diagram showing an electrical configuration of a microphone array 200 according to a fourth embodiment of the present invention.
  • FIG. 22 is a diagram showing an electrical configuration of a speaker array 300 according to a fifth embodiment of the present invention.
  • FIG. 23 is a diagram showing an electrical configuration of a speaker array 400 according to a sixth embodiment of the present invention.
  • FIG. 24 is a diagram showing a relationship between a two-dimensional frequency plane and a cross section.
  • FIG. 25 is a diagram showing an example of a cross-sectional design result.
  • FIG. 26 is a diagram showing an example of amplitude characteristics of the two-dimensional digital filter of the speaker array 400 on a two-dimensional frequency plane.
  • FIG. 27 is an equal amplitude characteristic diagram of the same amplitude characteristic.
  • FIG. 28 is a diagram illustrating a directivity characteristic of the speaker 400.
  • FIG. 29 is a diagram showing frequency characteristics according to modification examples (3) and (6).
  • FIG. 1 is a block diagram showing a configuration example of the speaker array 100 according to the first embodiment of the present invention.
  • this speaker array 100 is composed of transducers (speaking force in this embodiment) arranged in a straight line at a predetermined interval (in this embodiment, a constant interval D).
  • a predetermined interval in this embodiment, a constant interval D.
  • 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).
  • the audio data is supplied to each of the one-dimensional digital filters 120-i (i: natural number of 1 to n: hereinafter the same).
  • an analog audio signal is supplied to the speaker array 100 with external sound source power
  • digital audio data is also supplied to the speaker array 100 with external sound source power. In this way, it goes without saying that it is not necessary to provide the A / D converter in the speaker array 100 when audio data in digital format is supplied from an external sound source.
  • 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 output processing on the audio data delivered from the A / D converter according to the filter coefficient.
  • the audio data output from each force of the one-dimensional digital filter 120-i is converted into an audio signal by a ⁇ / A converter (not shown) and corresponds to the one-dimensional digital filter 120-i. Supplied to speaker 110-i.
  • the sound corresponding to the sound signal supplied from the D / A converter is emitted from each of the speakers 110-i. It becomes.
  • the speaker 110-i emits a sound corresponding to an analog audio signal.
  • 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 hardware configuration of the speaker array 100 according to the present embodiment is not different from the hardware configuration of the conventional speaker array.
  • filter coefficients characteristic of the speaker array according to the present invention are preset in each of the one-dimensional digital filters 120-i.
  • the formed two-dimensional digital filter is imparted with the characteristic amplitude characteristic of the speaker array according to the present invention, and the characteristic directivity characteristic of the speaker array according to the present invention is realized by the amplitude characteristic.
  • the amplitude characteristics of the two-dimensional digital filter formed by the one-dimensional digital filter 120-i and the directivity characteristics realized by the amplitude characteristics will be described with reference to the drawings.
  • each speaker 110-i has an ideal characteristic (that is, a characteristic that its directivity does not depend on the frequency of the output sound).
  • the speaker spacing D 0.068 [m]
  • sampling frequency f 745
  • 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 a 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. 2 (specific examples).
  • FIG. 5 is a diagram showing an equiamplitude characteristic diagram of the normalized time frequency f force in the range of 3 ⁇ 4 to 0.5 and the normalized spatial frequency f force in the range of 3 ⁇ 4 to 0.5. Standardized time frequency
  • the number is a value obtained by normalizing the temporal frequency by the reciprocal of the time sampling interval
  • the normalized spatial frequency is a value obtained by normalizing the spatial frequency by the inverse of the speaker arrangement interval D.
  • a plurality of ripples are present in a region where the normalized time frequency is low (for example, a region where ⁇ is 0 to 0.1).
  • a large amplitude in this embodiment, “1”
  • the ripples in the non-physical region are equal ripples with substantially the same amplitude, so the amplitude characteristics shown in FIG. 2 and FIG. This is called a ripple characteristic.
  • the normalized spatial frequency f is greater than or equal to a predetermined value.
  • the normalized spatial frequency f is less than the predetermined value.
  • the bandwidth in the normalized time frequency direction becomes narrower as the normalized spatial frequency f 2 decreases.
  • the normalized spatial frequency f is
  • the reason why the bandwidth in the normalized time frequency direction is narrowed as the value of the normalized spatial frequency f 2 is reduced is as follows. It is.
  • 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, Is generally known (see Non-Patent Document 2). As described above, for ripples in the region where the normalized spatial frequency f is less than the predetermined value,
  • the bandwidth in the normalized time frequency direction becomes narrower as the normalized spatial frequency f becomes smaller.
  • the frequency of Lipnoré in a non-physical region that can be used in a region where the normalized time frequency f is low (for example, a region where f force ⁇ ⁇ 0.1) is compared to the case where the bandwidth is kept constant. This is because the number is expected to increase and the main lobe width can be further reduced.
  • a predetermined value for the normalized spatial frequency f (a non-physical region is divided).
  • the normalized spatial frequency value to be divided is 0.25 will be described. However, such a value is not limited to 0.25. It may be determined appropriately according to the amplitude characteristics).
  • Fig. 4 shows the amplitude characteristics shown in Fig. 2 output from each speaker 110-i and speaker array 100.
  • the angle of the observation point direction (angle ⁇ in Fig. 1) viewed from the center of the speaker arrangement when the direction perpendicular to the arrangement direction of the speakers 110-i is 0 degrees in the plane including the observation point of It is the figure shown as a frequency characteristic.
  • the frequency characteristics for ⁇ 0 °, 12.5 °, 20 °, 60 °, and 90 ° are shown.
  • Figure 5 shows the directional characteristics at the amplitude of the heel force (371.09473Hz, 745. 8276 4Hz, 1491. 6553Hz, 2233. 8447Hz and 3354. 4053Hz). It is a figure.
  • the side lobe level is maintained while keeping the main lobe width of the acoustic beam constant for frequencies of a certain value or more. Can be kept almost constant (in this case-20dB).
  • graph L5 is The speaker array designed by the design method disclosed in Non-Patent Document 2
  • the graphs L6, L7, and L8 in Fig. 6 show two-dimensional digital filters with amplitude characteristics that have two-stage equal ripples in the stop band and a substantially constant bandwidth in the normalized time-frequency direction of ripples in the non-physical domain.
  • the speaker array 100 As apparent from FIG. 6, according to the speaker array 100 according to the present embodiment, the speaker array according to the rectangular in-phase drive, the spin force array according to the design method disclosed in Non-Patent Document 2, and the like. Compared with conventional speaker arrays of the above, and speaker arrays with a two-stage digital filter whose amplitude characteristics have a substantially constant bandwidth in the time domain in the normalization time frequency direction of ripple in the non-physical area with two-step stopband ripple. Thus, it is clear that the main lobe width in the low frequency range can be reduced.
  • 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 f of the directional loudspeaker array f: see Non-Patent Document 2 for details) performs rectangular in-phase drive.
  • the secondary Directivity in the low frequency range of the speaker array without increasing the array length by setting the original digital filter with the amplitude characteristics shown in Fig. 2 (that is, the amplitude characteristics satisfying the three requirements described below). It is possible to prevent the increase of the side lobe level.
  • Non-Patent Document 2 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 shown. It is disclosed that the frequency characteristic when observed at a sufficiently distant observation point is an amplitude characteristic distributed on a straight line represented by the following Equation 1 in a two-dimensional frequency plane.
  • f is the normalized time frequency
  • f is the normalized spatial frequency
  • D is the transformer.
  • T is the time sampling period
  • c is the speed of sound.
  • Non-Patent Document 2 shows a cross section in the normalized spatial frequency direction (ie, f direction) of the two-dimensional frequency plane as described above.
  • a method is disclosed in which FIR filter coefficients are obtained by arranging one-dimensional filter characteristics, setting target characteristics of a two-dimensional digital filter, and applying a two-dimensional Fourier series approximation to the target characteristics.
  • M is an even number
  • the target fan filter characteristics are obtained by juxtaposing in the order of steps.
  • Equation 3 f is the half-value frequency c L of the characteristics of the stopband ripple ⁇ shown in Fig. 8 (a).
  • the filter coefficient to be set for each one-dimensional digital filter is calculated by performing a two-dimensional inverse dispersive Fourier transform on the target amplitude characteristic of the fan filter set in this way.
  • Fig. 9 (a) Place a large ripple 1D filter on the cross section only in the non-physical region.
  • the two amplitude characteristics shown in Fig. 9 (a) are the amplitude characteristics of the one-dimensional filter placed on each cross section. As can be seen from the comparison of these two amplitude characteristics, increasing the ripple in the non-physical region broadens the frequency range occupied by the ripple, and conversely narrows the passband.
  • the amplitude of the ripple in the non-physical region is placed at the cross-sectional position of the time frequency in the lower sound range until the predetermined maximum value is reached.
  • the design is repeated so that the non-physical region is filled with a ripple with a predetermined maximum amplitude (or smaller amplitude) at each frequency f lower than that.
  • a program that performs filter design according to Parks &McClellan's equiripple filter design algorithm is used. is doing.
  • Parks &McClellan's equiripple filter design algorithm is an algorithm that uses the Remez exchange algorithm and the weighted Chebyshev approximation theory to design the filter so that the desired frequency response and the actual frequency response are optimized.
  • a filter designed according to this algorithm It is sometimes called a minimax filter because it is optimal in minimizing the maximum error between the actual 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.
  • the Parks & McClellan equiripple filter design algorithm is used for the design of a one-dimensional filter having two-stage stopband equiripple characteristics, and other FIR filter design algorithms may be used.
  • FIG. 10 is a diagram showing parameters given to the program and characteristics of the design results.
  • the one-dimensional filter is designed by repetitively approximating under the condition of and determining the filter coefficient.
  • 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.
  • the width of the passband is narrower than that of the latter by increasing the ripple in the stopband 2.
  • the effect of narrowing the width of the pass band in this way becomes more remarkable as the number of ripples in the non-physical area occupies the total number of ripples in the stop area.
  • the amplitude can be set as large as possible. 1S Practically, the upper limit value must be set appropriately. You can set 1 "(ie, a value equal to the passband amplitude) or" 2 "(a value twice the passband amplitude)!
  • the filter coefficients calculated in this way for each one-dimensional digital filter 120-i the amplitude shown in Fig. 2 is applied to the two-dimensional digital filter formed by these one-dimensional digital filters. Properties will be imparted.
  • 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, the speaker array according to the present embodiment 100
  • the width of the main lobe can be reduced with the side lobe level kept low especially in the low band as the final filter coefficient characteristic. Is possible.
  • the width of the main lobe is kept constant while keeping the influence of the side lobe low even in a lower band than in the past. It becomes possible.
  • the width of the main lobe depends on the number of ripples in the non-physical region and its amplitude, the amplitude set for the ripple in the non-physical region so that the necessary directivity can be obtained according to f.
  • the number it becomes possible to make the width of the main lobe constant in a lower band than before.
  • a region having a relatively high time frequency (for example, specified as f ⁇ f in Non-Patent Document 2).
  • the width of the main lobe can be made sufficiently narrow without increasing the ripple amplitude in the non-physical region.
  • the Dolph-Chebyshev characteristic disclosed in Patent Document 2 may be used. 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, in combination with the improvement of the 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 than before.
  • FIG. 12 is a diagram showing a configuration example of the microphone array 200 according to the second embodiment of the present invention.
  • the difference between the configuration of the microphone array 200 and the configuration of the speaker array 100 is that the speaker 110—i (i :; natural number of! ⁇ N)
  • a microphone 210-i (i :; natural number of! To n) that outputs an audio signal corresponding to the collected audio is provided.
  • 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.
  • the time-frequency characteristic of the plane wave arriving from the direction of the angle ⁇ shown in Fig. 12 is connected to each microphone (microphone 210-i in this embodiment) constituting the microphone array. It is generally known that the amplitude characteristics of the one-dimensional digital filter group distributed on the straight line represented by Equation 2 described above when viewed on a two-dimensional frequency plane. For this reason, 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. (That is, the effect of improving the directivity in the low frequency range of the microphone array without increasing the array length and avoiding an increase in the sidelobe level) can be obtained.
  • FIG. 13 is a block diagram showing a configuration example of a speaker array 500 according to the third embodiment of the present invention.
  • the configuration of the speaker array 500 shown in FIG. 13 is the same as that of the speaker array 100 shown in FIG. 1, and the same number of delay circuits 130—1, 130 as the number of speakers 1 10—1, 1 10—2,. — 2... 130 — ⁇ , a user interface (hereinafter “U / l”) part 140, and a wholesaler part 150.
  • the array surface of the speaker array 500 is formed by linearly arranging the speakers 110-i (i:;! To n) at a predetermined interval D. .
  • the center position of the array surface is referred to as “C” (see FIG. 13).
  • Each of the one-dimensional digital filters 120-i in FIG. 13 has a frequency having a directional main axis in a direction perpendicular to the array plane (that is, a steering angle of 0 degrees) through the point C shown in FIG. Filter coefficients that give the characteristics to the speaker array 500 are set in advance.
  • Each of these one-dimensional digital filters 120-i applies a filtering process corresponding to the filter coefficient to the audio data delivered from the A / D converter, and corresponds to the one-dimensional digital filter 120-i. Output to delay circuit 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 described later, and outputs the delayed audio data.
  • 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.
  • As for the ability to realize steering of the oriented spindle by applying such delay processing see “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 200 December 2004 ”(hereinafter referred to as“ references ”).
  • the delay amount in the delay circuit 130-i may be a delay amount in a sampling interval unit or a delay amount in a smaller unit (ie, a unit less than the sampling interval).
  • each of the delay circuits 130-i is configured by an FIR filter in order to realize delay processing that is not limited to the sampling interval unit.
  • each delay circuit 130-i is converted into an audio signal by a D / A converter (not shown), and it corresponds to the one-dimensional digital filter 120-i and the delay circuit 130-i. Is supplied to the speaker 110-i.
  • the sound corresponding to the sound signal supplied from the D / A converter is emitted from each of the speakers 110-i.
  • the force S the speed for explaining the case where the speaker 110 i emits a sound corresponding to an analog audio signal is described.
  • One power 1 10 If i emits sound corresponding to digital audio data, it is not necessary to install the D / A converter! / Needless to say! /.
  • the U / I unit 140 includes, for example, a display unit configured with a liquid crystal panel and a drive circuit thereof, and an operation unit configured with a plurality of operators such as a numeric keypad.
  • the U / I unit 140 sets the steering angle of the main spindle of the speaker array 500 within a predetermined angle range (FIG. 13: ⁇ degrees to ⁇ degrees).
  • a screen for allowing the user to specify the position is displayed on the display means, and data indicating the steering angle input by appropriately operating the operation element by the user viewing the screen is delivered to the control unit 150.
  • the U / I unit 140 functions as a designation unit that allows the user to designate the steering angle of the directional spindle.
  • the control unit 150 is for specifying the delay amount corresponding to the steering angle transmitted from the U / I unit 140 for each delay circuit 130-i and setting the delay amount to each delay circuit 130-i.
  • the control unit 150 includes a CPU (Central Processing Unit), a ROM (Re omitted), and a delay amount management table and a control program are stored in the ROM.
  • a speaker array is arranged in the steering angle direction in association with each angle of the predetermined angle unit within the predetermined angle range (that is, an angle mm range of ⁇ degrees to ⁇ degrees).
  • the delay amount stored in this delay amount management table is set so that the difference in delay amount between adjacent delay circuits is ⁇ shown in the following equation (1).
  • p D / (c -T): D is the transducer interval, T is the time sampling period, c is the speed of sound, ⁇ is stearin
  • control program stored in the ROM stores the data delivered from the U / I unit 140 and the contents stored in the delay amount management table by causing the CPU to execute the control program using the RAM as a work area. And the CPU sets the delay amount according to the steering angle represented by the data to each of the delay circuits 130-i. It is for execution.
  • the hardware configuration of the speaker array 500 according to this embodiment is not different from the hardware configuration of the conventional speaker array.
  • each of the one-dimensional digital filters 120-i realizes frequency characteristics having a directional main axis in the direction where the steering angle is 0 degree, and the speaker array according to the present invention. Since characteristic filter coefficients are set in advance, a characteristic amplitude characteristic is given to the speaker array according to the present invention to the two-dimensional digital filter formed by the one-dimensional digital filter. The characteristic directivity of the speaker array is realized.
  • each speaker 110-i has an ideal characteristic (that is, a characteristic that its directivity does not depend on the frequency of the output sound).
  • the speaker spacing D 0.068 [m]
  • sampling frequency f ( 1 /
  • T) 7451 [Hz]
  • number of FIR taps 241
  • n (number of speakers) 15.
  • FIG. 14 is a diagram showing the amplitude characteristics of the speaker array 500 and the directivity characteristics realized by the amplitude characteristics.
  • 14 (a) and 14 (b) are realized by an amplitude characteristic and a directivity characteristic (that is, realized by a two-dimensional digital filter) when the steering angle specified via the U / I unit 140 is 0 degree.
  • Fig. 14 (c) and Fig. 14 (d) show the amplitude when the steering angle is ⁇ degrees.
  • FIG. 14 (a) shows the main part of the amplitude characteristic of the two-dimensional digital filter formed by the one-dimensional digital filter 120-i (specifically, the normalized time frequency f is 0). In the range of ⁇ 0.5 and the normalized spatial frequency f force ⁇ ⁇ 0.5)
  • the normalized time frequency is the time sampling interval.
  • the normalized spatial frequency is a value obtained by normalizing the spatial frequency by the reciprocal of the arrangement interval D of the spin force.
  • the ripple in the non-physical area is given a large amplitude (in this embodiment, “1”), while the ripple in the physical area has an amplitude within the non-physical area. It is kept lower than the ripple.
  • the ripples in the non-physical region are equal ripples with substantially the same amplitude, so the amplitude characteristics shown in FIG. This is called a ripple characteristic.
  • 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 larger the passband It is generally known that the narrow main lobe width can be narrowed (see Non-Patent Document 2). For this reason, there is no ripple in the region where the normalized spatial frequency value is less than the predetermined value, for example, the force to provide a ripple as shown in Fig. 15 (a) without imposing any restrictions. It is expected that the number of ripples in the available non-physical area will increase and the main lobe width can be further reduced.
  • a ripple is provided only in a region of the non-physical region where the normalized spatial frequency f is equal to or greater than the predetermined value.
  • the ripple in the non-physical area does not protrude into the physical area (see Fig. 14 (c)) and appears in Fig. 15 (d).
  • the ripple in the non-physical area does not protrude into the physical area (see Fig. 14 (c)) and appears in Fig. 15 (d).
  • the normalized spatial frequency f force depends on the maximum steering angle.
  • the two-dimensional digital filter includes an amplitude characteristic as shown in FIG. 14 (a) (that is, the three requirements described below).
  • a filter coefficient that realizes an amplitude characteristic that satisfies the above requirements, the directivity in the low frequency range of the speaker array without increasing the array length is improved, and an increase in the side lobe level is avoided. Furthermore, it is possible to steer the directional main shaft without causing disturbance in the amplitude characteristics.
  • the above three requirements are as follows.
  • Non-Patent Document 2 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 shown. It is disclosed that the frequency characteristic when observed at a sufficiently distant observation point is an amplitude characteristic distributed on a two-dimensional frequency plane on a straight line represented by Equation 2 below.
  • f is the normalized time frequency
  • f is the normalized spatial frequency
  • D is the transformer.
  • T is the time sampling period
  • c is the speed of sound.
  • Non-Patent Document 2 shows a cross section in the normalized spatial frequency direction (ie, f direction) of the two-dimensional frequency plane as described above.
  • a method is disclosed in which FIR filter coefficients are obtained by arranging one-dimensional filter characteristics, setting target characteristics of a two-dimensional digital filter, and applying a two-dimensional Fourier series approximation to the target characteristics.
  • Non-Patent Document 2 describes a spin formed by (N + 1) speakers.
  • the acoustic beam center ⁇ As a design condition of the single force array, the acoustic beam center ⁇ , the beam end angle ( ⁇ , ⁇ ) and
  • a design procedure for a two-dimensional digital filter is disclosed in the case where the size (amplitude) ⁇ of an equal ripple side lobe is given.
  • ⁇ 0.
  • ⁇ ⁇
  • ⁇ ⁇
  • f is the half-value frequency c L of the characteristics of the stopband ripple ⁇ shown in Fig. 17 (a).
  • the filter coefficient to be set for each one-dimensional digital filter is calculated by performing a two-dimensional inverse dispersive Fourier transform on the target amplitude characteristic of the fan filter set in this way.
  • FIG. 18 (b) 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. As shown in Fig. 18 (b), in the two-dimensional frequency plane divided into M, a cross-sectional characteristic of a one-dimensional filter with a small ripple is kept in the entire stopband when f ⁇ f, while in a non-physical region when f ⁇ f.
  • Fig. 18 (a) The two amplitude characteristics shown in Fig. 18 (a) are the amplitude characteristics of the one-dimensional filter placed on each cross section. As can be seen from the comparison of these two amplitude characteristics, increasing the ripple in the non-physical region broadens the frequency range occupied by the ripple, and conversely narrows the passband. [0060] In this embodiment, in order to design a one-dimensional filter with two-step stopband equiripple characteristics as shown in Fig. 18 (a), filter design is performed according to Parks &McClellan's equiripple filter design algorithm. You are using a program.
  • Parks &McClellan's equiripple filter design algorithm is an algorithm that uses the Remez exchange algorithm and the weighted Chebyshev approximation theory to design the filter so that the desired frequency response and the actual frequency response are optimized.
  • Filters designed according to this algorithm are sometimes referred to as minimax filters because they are 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.
  • other FIR filter design algorithms are used. Of course! / ⁇ Of course.
  • FIG. 19 is a diagram showing parameters given to the program and characteristics of the design results.
  • the one-dimensional filter is designed by repetitively approximating under the condition of and determining the filter coefficient.
  • FIG. 20 is a diagram in which a one-dimensional filter designed according to Parks &McClellan's equiripple filter design algorithm and a design example of a one-dimensional filter having a Dolph-Chebyshev characteristic are shown.
  • the passband width is narrower than the latter by increasing the ripple in stopband 2.
  • the effect of narrowing the width of the pass band in this way becomes more remarkable as the number of ripples in the non-physical area occupies the total number of ripples in the stop area.
  • the amplitude can be set as large as possible. 1S Practically, the upper limit value must be set appropriately. 1 "(ie a value equal to the passband amplitude) or" 2 "(a value twice the passband amplitude) It should be set! /.
  • filter coefficients that satisfy the requirements (a-1) and (a-2) are calculated.
  • such a condition can be obtained by adjusting (increasing) the f, which is the condition of the stopband 1, for each f from the value determined by the specification.
  • the design is realized.
  • the value of f is as specified, and the design is such that the value decreases as fl becomes smaller.
  • the filter coefficients calculated in this way for each one-dimensional digital filter 120-i the amplitude characteristics shown in Fig. 14 (a) are given to the two-dimensional digital filter formed by these one-dimensional digital filters. Will be.
  • the characteristics of the physical area directly affect the directivity, while the characteristics in the non-physical area do not directly affect the directivity.
  • the width of the main lobe can be reduced with the side lobe level kept low especially in the low band as the final filter coefficient characteristic. Is possible.
  • the amplitude characteristic that imposes the use restriction determined according to the maximum steering angle in the non-physical region is set in the two-dimensional digital filter! Even if steering is performed at a smaller steering angle, the amplitude characteristics will not be disturbed in the direction of 90 degrees. That is, according to the speaker array 500 according to the present embodiment, the pointing main shaft can be steered in any direction within an angle range determined according to the maximum steering angle.
  • the width of the main lobe can be made sufficiently narrow without increasing the ripple amplitude in the non-physical region.
  • the Dolph-Chebyshev characteristic disclosed in Patent Document 2 may be used. Also, In 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, it is possible to improve the characteristics in the low region according to the present embodiment and It becomes possible to obtain directivity characteristics that do not depend on time frequency in a wide band.
  • FIG. 21 is a diagram showing a configuration example of a microphone array 200 according to the fourth embodiment of the present invention.
  • the configuration of the microphone array 200 is different from the configuration of the speaker array 500 in that the sound corresponding to the collected sound is used instead of the speaker 110-i. This is the point that a microphone 210-i that outputs a signal is provided.
  • the audio signal output from the microphone 210-i is converted into audio data by an A / D converter (not shown), and the delay circuit 130-i responds to the steering angle. Is input to the one-dimensional digital filter 120-i.
  • the above-described filter processing 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 the sum as a result of the addition is added.
  • a signal is output.
  • the A / D converter is not necessary when digital audio data is output from the microphone 210-i.
  • the time-frequency characteristics of the plane wave arriving from the angle ⁇ direction shown in Fig. 21 are connected to the respective microphones (microphones 210-i in this embodiment) constituting the microphone array. It is generally known that the amplitude characteristics of the 1D digital filter group distributed on the 2D frequency plane are distributed on the straight line represented by Equation 3 above.
  • the filter coefficients described in the above-described third embodiment that is, the filter coefficients satisfying the above-described three requirements
  • the directional characteristics of the microphone array 200 improve the directivity in the low frequency range of the microphone array without increasing the array length, and avoid increasing the sidelobe level. It becomes possible There is an effect.
  • the microphone array 200 by setting the delay amount corresponding to the steering angle input via the U / I unit 140 to each delay circuit 130-i, the directivity spindle Even if this kind of steering is performed, the amplitude characteristic of the two-dimensional digital filter is restricted by the requirement (a-3) described above, so that There is no disturbance in the amplitude characteristics.
  • FIG. 22 is a diagram showing a configuration example of a speaker array 300 according to the fifth embodiment of the present invention. As is clear from the comparison between FIG. 22 and FIG. 13, the configuration of the speaker array 300 is different from the configuration of the speaker array 500.
  • the delay circuit 130—i (i :; natural number of! To n) It is a point that does not have.
  • the filter coefficient force satisfying the above three requirements S is set in advance as a filter coefficient when the steering angle is 0 degree. Yes. Then, when the steering angle is input via the U / I unit 140, the control unit 150 calculates a delay amount corresponding to the steering angle, and the filter coefficient satisfying 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 direction spindle is realized.
  • a passband is formed so as to form a main lobe in the specified 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 stopband within the non-physical zone;
  • (b-2) crossing the first and second stopbands (B—3) the amplitude of the ripple in the second stop band is larger than the amplitude of the first stop band, and in the first and second stop bands.
  • the speaker array 300 includes 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!
  • the difference between the speaker array 500 and the 1D digital filter 120—i of the speaker array 300 is that the 1D digital filter function of the speaker array 500 and the delay circuit 130—i Therefore, the speaker array 300 can achieve the same operational effects as the speaker force array 500.
  • the microphone array 300 is the configuration of the speaker array 300 according to the fifth exemplary embodiment of the present invention.
  • the microphone array may be configured by replacing each speaker 110-i of the speaker array 300 with the microphone 210-i. 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.
  • FIG. 23 is a diagram showing 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 it has a storage unit 160.
  • the storage unit 160 is, for example, a hard disk and corresponds to each of a plurality of angles in a predetermined angle unit within a predetermined steering range, and the steering main axis is steered to that angle.
  • the control unit 150 reads out the filter coefficient corresponding to the steering angle from the storage unit 160, and Set to each of digital filter 120-i. As a result, steering of the directional spindle is realized.
  • the design of the one-dimensional filter is performed by repeatedly executing approximation using a predetermined filter coefficient design algorithm and determining the filter coefficient.
  • the filter coefficient design algorithm uses the Parks & McClellan equiripple filter design algorithm described above.
  • the design is repeated so that the non-physical region is filled with a ripple having a specified maximum amplitude or lower than each of the low! / And frequency f! /
  • the design is designed to reduce the increase in the main lobe width at low frequencies.
  • this design is realized by appropriately adjusting (increasing) the values of fst and fst 'that are the conditions of stopband 1 and stopband ⁇ for each f from the values determined by the specifications. is doing. (Fe fe 'is a value as specified, and becomes smaller as fl becomes smaller).
  • 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. Is a diagram showing an equiamplitude characteristic diagram in a range in which the normalized time frequency fl is ⁇ 0.5 0.5 and the normalized spatial frequency f 2 is ⁇ 0.5 0.5.
  • Fig. 28 shows the amplitude characteristics shown in Fig. 26 as the directivity characteristics at the frequency of the repulsive force (371. 09473Hz, 745. 82764Hz, 1491. 6553Hz, 2233. 8447Hz and 3354. 4053Hz).
  • FIG. As can be seen from FIG.
  • the main spindle is steered at 330 340 degrees (ie, -30 to 20 degrees). Note that the steering Since the generated frequency is lowered, a grating lobe appears in the directivity shown in FIG. 28. Normally, however, it is sufficient to use only a band where no grating is generated.
  • each microphone 110-i in the speaker array 300 can be replaced with the microphone 210-i, and the microphone array capable of steering the directional main axis in any direction. Of course, it may be configured. According to the microphone array configured as described above, it is possible to steer the directional main axis in an arbitrary direction. Of course, as with the microphone array 200 described above, the speaker array without increasing the array length. In addition to improving the directivity of the microphone array in the low sound range, it is possible to avoid an increase in the sidelobe level.
  • the filter coefficient to be set to i 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” Directional Array Speaker Using Two-Dimensional Digital Filter ”(1995)”, for example.
  • the non-physical region in the stop band! / Is provided with an equal ripple having an amplitude equal to or greater than the amplitude of the pass band, and the physical region.
  • the ripple in the non-physical area corresponds to the ripple.
  • the amplitude characteristics are given to the two-dimensional digital filter, where the smaller the spatial frequency is, the narrower the bandwidth in the time frequency direction of the ripple is, but the ripples in the non-physical region are not necessarily equal. For example, as shown in Fig.
  • a ripple having a larger amplitude than the passband in the stopband in the non-physical region (stopband 2 in Fig. 29) and an amplitude larger than that ripple
  • the stop region is provided with a ripple having a smaller amplitude and a ripple having a larger amplitude than the ripple in the stop region (stop region 1 in FIG. 29) in the physical region.
  • Speaker The frequency characteristics of the two-dimensional digital filter of a ray or microphone array are that multiple ripples are provided in the stop band, and the ripple amplitude in the non-physical region is larger than the ripple amplitude in the physical region, and
  • the ripple in the non-physical region may have a frequency characteristic such that the smaller the spatial frequency corresponding to the ripple, the narrower the bandwidth in the time frequency direction of the ripple is.
  • the filter coefficient may be sequentially calculated and set each time the speaker array or microphone array according to the above is used.
  • the directivity corresponding to the acoustic characteristics of the acoustic space such as the size and shape of the acoustic space is used. It becomes possible to set characteristics appropriately.
  • the filter coefficient set for each one-dimensional digital filter may be given from outside the speaker array or microphone array.
  • communication means such as a NIC (Network Interface Card), for example, and a filter coefficient acquired via the communication network using the communication means are set in each one-dimensional digital filter in the speaker array or the microphone array.
  • a filter coefficient setting means For example, a reading means for reading data from a computer-readable recording medium such as a CD-ROM (Compact Disk-Read Only Memory) is provided in place of the communication means.
  • a computer-readable recording medium such as a CD-ROM (Compact Disk-Read Only Memory) is provided in place of the communication means.
  • the filter coefficient may be written and distributed on a recording medium, and the filter coefficient read by the reading means may be set to each one-dimensional digital filter by the filter coefficient setting means.
  • the above-mentioned restrictions (requirements (a-3) or (b-1)) mentioned above are described in the non-physical area! /.
  • ripples in non-physical areas do not necessarily have to be equiripple. For example, as shown in FIG. 29, in the stop band in the non-physical area (stop band 2 in FIG.
  • the stop band multi-stage 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).
  • the frequency characteristics of the two-dimensional digital filter of the speaker array and microphone-phone array according to the present invention are such that a plurality of ripples are provided in the stop band, and the ripple amplitude in the non-physical region is the amplitude of the ripple in the physical region. And the frequency characteristics are such that ripples in the non-physical region do not protrude into the physical region when steering the steering spindle! / Good!
  • 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. Force Each time the speaker array or microphone array according to the present invention is used, the filter coefficient may be calculated and set sequentially. In this way, for example, when the speaker array or microphone array according to the present invention is installed and used in an acoustic space such as a concert hall, the directivity corresponding to the acoustic characteristics of the acoustic space such as the size and shape of the acoustic space is used. It becomes possible to set characteristics appropriately.
  • the filter coefficients set for each of the above one-dimensional digital filters can be You may make it give from the exterior of a microphone array.
  • communication means such as a NIC (Network Interface Card), for example, and a filter coefficient acquired via the communication network using the communication means are set in each one-dimensional digital filter in the speaker array or the microphone array.
  • a filter coefficient setting means For example, a reading means for reading data from a computer-readable recording medium such as a CD-ROM (Compact Disk-Read Only Memory) is provided in place of the communication means.
  • the filter coefficient may be written and distributed on the recording medium, and the filter coefficient read by the reading unit may be set to each one-dimensional digital filter by the filter coefficient setting unit.
  • the present invention it is possible to improve the directivity in the low sound range of a speaker array or a microphone array without increasing the array length, avoid an increase in the level of side lobes, and further, The effect is that steering of the vehicle becomes possible.

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  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Obtaining Desirable Characteristics In Audible-Bandwidth Transducers (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

L'invention permet d'améliorer la directivité d'un réseau de haut-parleurs et d'un réseau de microphones dans une zone à tonalités basses et d'éviter une augmentation du niveau des lobes secondaires sans augmenter la longueur du réseau. Un coefficient de filtre satisfaisant les trois conditions suivantes est établi pour les filtres numériques monodimensionnels fixés aux haut-parleurs respectifs constituant un réseau de haut-parleurs. Lorsque la caractéristique de fréquence des filtres numériques bidimensionnels formés par les filtres numériques monodimensionnels est exprimée dans un plan de fréquence bidimensionnel, (a) il existe plusieurs ondulations dans une zone interdite dans la section transversale de la direction de fréquence spatiale, (b) parmi ces ondulations, celles dans la zone non physique ont une amplitude supérieure à celles se trouvant dans la zone physique, et (c) parmi les ondulations de la zone non physique, les ondulations se trouvant dans une zone possédant une fréquence spatiale inférieure à une valeur prédéfinie ont une largeur de bande plus étroite dans le sens de la fréquence temporelle des ondulations à mesure que la fréquence spatiale correspondant aux ondulations diminue.
PCT/JP2007/068452 2006-09-25 2007-09-21 Réseau de haut-parleurs et réseau de microphones WO2008041530A1 (fr)

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Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2011041023A (ja) * 2009-08-11 2011-02-24 Kanazawa Univ デジタル音響信号処理装置
JP2016099606A (ja) * 2014-11-26 2016-05-30 ソニー株式会社 信号処理装置、信号処理方法及びプログラム
US11310617B2 (en) 2016-07-05 2022-04-19 Sony Corporation Sound field forming apparatus and method

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Title
NISHIKAWA K. AND TAKABAYASHI S.: "Kotaiiki Shingo ni Taisuru Beam Steering", IEICE TECHNICAL REPORT, THE INSTITUTE OF ELECTRONICS, INFORMATION AND COMMUNICATION ENGINEERS, vol. 93, April 1993 (1993-04-01), pages 33 - 40 *
OTA M. AND NISHIKAWA K.: "Ittei Sidelobe-ryo no Shikosei Array Speaker no Kotaiiki Sekkei", IEICE TECHNICAL REPORT, THE INSTITUTE OF ELECTRONICS, INFORMATION AND COMMUNICATION ENGINEERS, vol. 105, October 2005 (2005-10-01), pages 7 - 12 *

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2011041023A (ja) * 2009-08-11 2011-02-24 Kanazawa Univ デジタル音響信号処理装置
JP2016099606A (ja) * 2014-11-26 2016-05-30 ソニー株式会社 信号処理装置、信号処理方法及びプログラム
US11310617B2 (en) 2016-07-05 2022-04-19 Sony Corporation Sound field forming apparatus and method

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