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JP2007135199

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DESCRIPTION JP2007135199
An object of the present invention is to improve the directivity of a speaker array or a
microphone array in the low frequency range without increasing the array length, and to avoid
an increase in the level of a side lobe. A frequency characteristic of a two-dimensional digital
filter formed by the one-dimensional digital filters is expressed by a two-dimensional frequency
plane in a one-dimensional digital filter connected to each of a plurality of speakers forming a
speaker array. , In a cross section in the spatial frequency direction, an amplitude having a
plurality of ripples in the stop band and an amplitude of a ripple in a non-physical region among
the plurality of ripples being larger than an amplitude of a ripple in the physical region Set filter
coefficients that give characteristics to the two-dimensional digital filter. [Selected figure] Figure
2
スピーカアレイおよびマイクロホンアレイ
[0001]
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
range.
[0002]
A sound field is formed only in a specific direction using a speaker array or a microphone array
formed by arranging a plurality of transducers such as speakers and microphones linearly at
predetermined intervals, or only sound coming from a specific direction Techniques for picking
04-05-2019
1
up sound are generally widespread.
By the way, in this type of speaker array or microphone array, it is desirable that the same
directivity characteristic can be realized over a wide band from the high tone range to the low
tone range. However, the directivity characteristic of the low-pitched sound region improves as
the array length of the speaker array or the microphone array (the value obtained by multiplying
the number of transducers by the arrangement interval of the transducers) increases (see NonPatent Document 1). In order to ensure sufficient directivity in the sound range, there has been a
problem that the device size of the speaker array or the microphone array becomes large.
Therefore, various techniques for solving the problems as described above have conventionally
been proposed, and an example thereof is the technique disclosed in Non-Patent Document 2. In
this non-patent document 2, regarding the amplitude characteristics of digital filters connected to
each of the transducers constituting the speaker array and the microphone array, the cross
section in the spatial frequency direction on the two-dimensional frequency plane is a ripple
characteristic such as a stop band. -A technique is disclosed for extending the band that can give
the same directional characteristic to the bass side by setting the filter coefficient of each digital
filter so as to be the amplitude characteristic (or its approximate characteristic) of the Chebyshev
filter . Jiro Oga, Yoshio Yamazaki, Yutaka Kanada "Acoustic Systems and Digital Signal
Processing" The Institute of Electronics, Information and Communication Engineers 1993-05
p176-186 Koji Matsumoto, Kiyoshi Nishikawa "Design Method of Directional Array Loudspeakers
with Constant Sidelobe Amount" The Institute of Electronics, Information and Communication
Engineers Academic Bulletin 2004-74 p13-18
[0003]
However, the ripple of the stop band equal ripple characteristic is a non-physical region (in a
two-dimensional frequency plane, a region of | f2 |> ρ | f1 |. However, since it usually exists in a
region other than ρ = D / cT, T: sampling interval, D: speaker interval, c: sound velocity, f1:
normalized time frequency, f2: normalized space frequency) If a large amplitude is given to the
ripples in the stop zone and the like in order to improve the property, there is a problem that the
amplitude level of the side lobe, which is an originally unnecessary directivity characteristic, is
increased. The present invention has been made in view of the above problems, and it is possible
to improve the directivity of the speaker array and the microphone array in the low frequency
range without increasing the array length, and to avoid an increase in the amplitude level of the
side lobes. The aim is to provide a technology that makes it possible to
[0004]
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2
In order to solve the above problems, according to the present invention, a plurality of speakers
linearly arranged at predetermined intervals and a plurality of the speakers are provided
corresponding to each other, and predetermined filter coefficients are set in advance. And a onedimensional digital filter for applying the filter processing according to the filter coefficient to
the input audio data and outputting the one-dimensional digital filter, and the audio data
obtained by performing the digital conversion on the input audio signal is each one-dimensional
A speaker array for supplying an audio signal obtained by performing analog conversion on
audio data output from each of the one-dimensional digital filters while supplying to a digital
filter to the corresponding speaker and outputting audio corresponding to the audio signal The
filter coefficients set in each one-dimensional digital filter are formed by the one-dimensional
digital filters. When the frequency characteristic of the two-dimensional digital filter to be
displayed is represented by a two-dimensional frequency plane, a cross section in the spatial
frequency direction has a plurality of ripples in the stop band, and The present invention
provides a speaker array characterized in that it is a filter coefficient for providing the twodimensional digital filter with an amplitude characteristic in which the amplitude of the ripple in
the physical region is larger than the amplitude of the ripple in the physical region.
[0005]
Further, in order to solve the above problems, according to the present invention, a plurality of
microphones linearly arranged at predetermined intervals and a plurality of microphones are
provided corresponding to each of the plurality of microphones, and predetermined filter
coefficients are preset. And one-dimensional digital filter for applying the filtering process
according to the filter coefficient to the input audio data and outputting it, and performing digital
conversion on the audio signal output from each of the plurality of microphones In the
microphone array that supplies the obtained audio data to the corresponding one-dimensional
digital filter, and outputs the sum signal of the audio data output from the one-dimensional
digital filters, the one-dimensional digital filters are set. The filter coefficients of the twodimensional digital filter formed by each one-dimensional digital filter are When the wave
number characteristics are represented by a two-dimensional frequency plane, the cross section
in the spatial frequency direction has a plurality of ripples in the stop band, and among the
plurality of ripples, the amplitude of the ripple in the non-physical region is the physical region
The present invention provides a microphone array characterized in that it is a filter coefficient
for providing the two-dimensional digital filter with an amplitude characteristic that is larger
than the amplitude of the internal ripple.
[0006]
According to the present invention, it is possible to improve the directivity of the speaker array
or the microphone array in the low frequency range without increasing the array length, and to
avoid an increase in the side lobe level.
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3
[0007]
The best mode for carrying out the present invention will be described below with reference to
the drawings.
(A.
First Embodiment (A-1: Configuration) FIG. 1 is a block diagram showing a configuration example
of a speaker array 100 according to a first embodiment of the present invention.
As shown in FIG. 1, the speaker array 100 includes transducers (in the present embodiment,
speakers) 110-1 and 110-2 linearly arranged at predetermined intervals (in the present
embodiment, a constant interval D). .. 110-n, and the same number of one-dimensional digital
filters 120-1, 120-2.
[0008]
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 one-dimensional digital filter 120-i
(i: natural number of 1 to n: the same applies hereinafter). In each of the one-dimensional digital
filters 120-i of FIG. 1, filter coefficients characteristic to the speaker array according to the
present invention are set in advance. Each of these one-dimensional digital filters 120-i performs
the filtering process according to the filter coefficient to the audio data delivered from the A / D
converter and outputs it. Then, audio data output from each of the one-dimensional digital filters
120-i is converted into an audio signal by a D / A converter (not shown), and the speaker 110corresponding to the one-dimensional digital filters 120-i. It is supplied to i. As a result, from
each of the speakers 110-i, an audio corresponding to the audio signal supplied from the D / A
converter is emitted. The above is the configuration of the speaker array 100.
[0009]
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4
As described above, the hardware configuration of the speaker array 100 according to the
present embodiment is the same as the hardware configuration of the conventional speaker
array. However, in the speaker array 100 according to the present embodiment, filter coefficients
characteristic to the speaker array according to the present invention are preset in each of the
one-dimensional digital filters 120-i. A characteristic amplitude characteristic is imparted to the
speaker array according to the present invention to the two-dimensional digital filter to be
formed, and the directional characteristic 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). Also, in the following, it is assumed that the speaker
arrangement interval D = 0.068 [m], the sampling frequency fs = 6087 [Hz], the number of FIR
taps = 61, n (the number of speakers) = 15.
[0010]
(A-2: Amplitude Characteristics and Directional Characteristics of Two-Dimensional Digital Filter)
FIGS. 2 to 6 are diagrams showing the amplitude characteristics of the two-dimensional digital
filter of the speaker array 100 and the directional characteristics realized by the amplitude
characteristics. FIG. 2 is a diagram showing the amplitude characteristic of the two-dimensional
digital filter formed by the one-dimensional digital filter 120-i in a two-dimensional frequency
plane, and FIG. 3 is a part of the amplitude characteristic shown in FIG. In the diagram, the
normalized time frequency f1 is in the range of 0 to 0.5, and the normalized spatial frequency f2
is in the range of 0 to 0.5) in the form of equal amplitude characteristics. 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 arrangement interval D of the speakers. It is a value. As
apparent from FIG. 2 and FIG. 3, in the speaker array 100 according to the present embodiment,
in the region where the normalized time frequency f1 of the stop band is low (for example, the
region where f1 is 0 to 0.1) A plurality of ripples are provided, and a large amplitude ("1" in this
embodiment) is given to the ripples in the non-physical region among the plurality of ripples, and
the ripples in the physical region are , Its amplitude is kept lower than the ripple in the nonphysical region. As apparent from FIG. 2 and FIG. 3, since the ripples in the non-physical region
are equal ripples having substantially the same amplitude, the amplitude characteristics shown in
FIG. 2 and FIG. It is called a characteristic.
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[0011]
FIG. 4 shows that the amplitude characteristic shown in FIG. 2 is 0 degrees in the direction
perpendicular to the arrangement direction of the speakers 110-i in the plane including the
speakers 110-i and the observation points of the sound outputted from the speaker array 100. It
is the figure shown as a frequency characteristic about the angle (angle phi of Drawing 1) of the
direction of the observation point seen from the center of a speaker arrangement in the case. In
FIG. 4, frequency characteristics for φ = 0 °, 24 °, 40 °, 70 ° and 90 ° are shown. As
apparent from FIG. 4, the amplitude level of the acoustic beam output from the speaker array
100 according to the present embodiment is in the direction of φ = 24 ° when the frequency is
higher than a certain value. The amplitude level decreases by about 6 [dB] compared to the φ =
0 ° direction, and the amplitude level is about 20 compared to the φ = 0 ° direction for the φ
= 40 °, 70 ° and 90 ° directions. It can be seen that [dB] is decreasing. FIG. 5 is a diagram
showing the amplitude characteristics shown in FIG. 2 as directivity characteristics at several
frequencies (202.10742 Hz, 404.21484 Hz, 499.32422 Hz, 998.64844 Hz, 1997.2969 Hz and
2995.9453 Hz). is there. As apparent from FIG. 4 and FIG. 5, according to the speaker array 100
according to the present embodiment, the level of the side lobe is maintained while the main lobe
width of the acoustic beam is kept constant for frequencies above a certain value. It can be seen
that it is possible to keep V.sub.2 substantially constant (in this case -20 dB).
[0012]
FIG. 6 compares the main lobe width (φ = 0 ° direction) of the speaker array 100 according to
the present embodiment and the speaker array of the conventional rectangular in-phase drive
(in-phase drive by signals subjected to rectangular window processing) It is the figure which
plotted the angle which represents the width | variety of the area | region where the amplitude of
an acoustic beam falls 6 [dB] for every frequency. As apparent from FIG. 6, according to the
speaker array 100 according to the present embodiment, it is possible to narrow the main lobe
width in the low frequency range as compared with the conventional rectangular in-phase drive
speaker array. It is understood that 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 °) as
the main lobe width is determined, the main lobe width is used. Lower limit of the frequency of
the acoustic beam (ie, the lower end fL of the band of the directional speaker array: refer to NonPatent Document 2 for details) is reduced compared to the conventional rectangular common
mode drive. It is understood that More specifically, when the ripple amplitude of the non-physical
region is set to “1” (FIG. 6: gain 1), the lower end of the band of the speaker array is 20
compared to the conventional rectangular in-phase driving. In the case where the amplitude of
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6
the ripple in the non-physical region is set to “2” (FIG. 6: gain 2), the reduction is 32.8%. 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, improve the directivity of the bass region) compared to
the conventional rectangular in-phase drive speaker array. There is.
[0013]
As described above, in the speaker array 100 according to the present embodiment, when the
frequency characteristics of the two-dimensional digital filter are represented by the twodimensional frequency plane, in the cross section in the spatial frequency direction, An amplitude
characteristic having a plurality of ripples, and 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 (in the present embodiment, the blocking shown in FIG. 2) By setting the (two-stage etc.
ripple characteristics), it is possible to improve the directivity of the speaker array or the
microphone array in the low frequency range and to avoid an increase in the side lobe level
without lengthening the array length. It has become. Next, the design of a two-dimensional digital
filter (that is, calculation of filter coefficients to be set to each one-dimensional digital filter 120-i)
for realizing a two-step ripple characteristic such as a two-step stop band as shown in FIG.
[0014]
(A-3: Design of Two-Dimensional Digital Filter) In the non-patent document 2 described above,
the amplitude characteristics of the two-dimensional digital filter formed by the group of onedimensional digital filters connected to each speaker are two-dimensional frequency It is
disclosed that the frequency characteristics when the output of the speaker array is observed at a
sufficiently distant observation point when viewed in a plane are amplitude characteristics
distributed on a straight line represented by the following equation 1 in a two-dimensional
frequency plane It is done. Where f1 is the normalized time frequency, f2 is the normalized
spatial frequency, D is the transducer interval, and T is the time sampling period. , C is the speed
of sound.
[0015]
Therefore, the directivity characteristic of the speaker array at a certain non-normalized time
frequency f is, on a two-dimensional frequency plane, on a straight line defined by the
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standardized time frequency f1 = f · T corresponding to the non-normalized time frequency f It
can be said that it is distributed by the relationship shown by the following equation 2. (Equation
2) φ = sin <-1> {(f2 · c · T) / (f1 · D)}
[0016]
That is, if it is possible to design a two-dimensional digital filter so that the desired directivity
characteristics at each non-normalized time frequency f are distributed on the straight line f1 = f ·
T in the relationship represented by the above equation 2, it is desired as a result The directional
characteristics of are obtained. In Non-Patent Document 2, as described above, the target
characteristics of the two-dimensional digital filter are set by arranging the one-dimensional filter
characteristics on the cross section in the normalized spatial frequency direction (that is, the f2
direction) of the two-dimensional frequency plane. A method is disclosed for obtaining FIR filter
coefficients by applying a two-dimensional Fourier series approximation to the property.
[0017]
More specifically, Non-Patent Document 2 describes the acoustic beam center φ 0 and beam end
angles (φ s +, φ s −) and equal ripple side lobes as a design condition of a speaker array formed
of (N 2 +1) speakers. A design procedure for a two-dimensional digital filter is disclosed where
the magnitude (amplitude) δ is given. In the following, it is assumed that φ0 = 0 °, φs + = φs,
and φs − = − φs (that is, the acoustic beam is symmetrical with respect to the center (φ0 = 0
°) axis).
[0018]
In the design procedure disclosed in Non-Patent Document 2, first, as shown in FIG. 7B, the twodimensional frequency plane is M1 (in this embodiment, in the range of f1 = −0.5 to 0.5). Design
the Dolph-Chebyshev characteristic on the cross section at each frequency f1 = k1 / M1 (k1 = M1 / 2 to M1 / 2) and divide them in parallel by the following procedure. To achieve the target
fan filter characteristics. The following description is based on the area of f1 ≧ 0.
[0019]
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Specifically, first, the characteristics of a Dolph-Chebyshev filter of order N 2 and stopband ripple
size δ are designed, and the stopband end frequency fst is a straight line with φ = φs (ie, f 2 = f
1 · D · sin A frequency fl is obtained when it matches the straight line represented by (φs) / (c ·
T). Then, at the cross-sectional position of f1 ≧ fl, the cross-sectional characteristics at f1 = fl are
expanded in the f2 direction so that the blocking region end is positioned on the straight line of
φ = φs, as shown in FIG. Deploy. On the other hand, for the cross-sectional position of f1 <fl, as
shown in FIG. 8A, a Dolph-Chebyshev filter characteristic (f2 in which the stopband ripple is
gradually increased from δ to a predetermined tolerance value δL) =-0.5 to 0.5). However, the
stop zone ripple is determined so that the stop zone end fst is located on the straight line of φ =
φs in any cross section. Then, as shown in FIG. 8 (b), assuming that the value of the frequency f1
at which the characteristic of the stop band ripple δL is initially placed is fu, the same
characteristic as the cross-sectional characteristic of f1 = fu is set at f1 <fu . Note that fL in FIG.
8B is the lower end of the band of the speaker array, and is a value determined by the following
Equation 3. (Equation 3) fL = c · T · fc / (D sin (φ s)) where fc is the half amplitude frequency of
the Dolph-Chebyshev filter characteristic of the stopband ripple δL shown in FIG. It is.
Thereafter, by applying two-dimensional inverse discrete Fourier transform to the target
amplitude characteristics of the fan filter set in this manner, filter coefficients to be set in each
one-dimensional digital filter are calculated.
[0020]
On the other hand, in the design of the two-dimensional digital filter of the speaker array 100
according to the present embodiment, in the two-dimensional frequency plane divided by M1 as
shown in FIG. While a one-dimensional filter with small ripples is used as the cross-sectional
characteristic, when f1 <fl, one-dimensional filters with large ripples are placed in the cross
section only in the non-physical region (shaded area in the figure). The two amplitude
characteristics shown in FIG. 9A are the amplitude characteristics of a one-dimensional filter
placed on each cross section. As can be seen from the comparison of these two amplitude
characteristics, by increasing the ripple in the non-physical region, the frequency range occupied
by the ripple is broadened, and conversely the passband is narrowed. Therefore, in the design of
the two-dimensional digital filter according to the present embodiment, the cross-sectional
position of the time frequency of the bass region is further placed until the ripple amplitude in
the non-physical region reaches a predetermined maximum value.
[0021]
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9
In the present embodiment, in order to design a one-dimensional filter with two-stage stop
characteristics with stop band and the like as shown in FIG. 9A, a program for performing filter
design in accordance with Parks & McClellan's equal ripple filter design algorithm is used. . Here,
Parks & McClellan's equiripple filter design algorithm is an algorithm for designing a filter so as
to optimize a desired frequency response and an actual frequency response using Remez
exchange algorithm and weighted Chebyshev approximation theory. Filters designed according to
this algorithm are sometimes referred to as minimax filters because they are optimal in that they
minimize the maximum error between the desired frequency response and the actual frequency
response. Also, filters designed according to this algorithm are also known as equal ripple filters
because they exhibit equal ripple in their frequency response. Although this embodiment
describes the case of using the Parks & McClellan equiripple filter design algorithm for designing
a one-dimensional filter with two-stage stopband etc. ripple characteristics, other FIR filter design
algorithms may be used. Of course.
[0022]
FIG. 10 is a diagram showing the parameters given to the program and the characteristics of the
design result. As shown in FIG. 10, in the present embodiment, three approximate bands (pass
band, stop band 1 and stop band 2) are set, and target amplitudes of the respective approximate
bands (one each) are defined as parameters defining the respective approximate bands. , 0, 0,
error ripples (δ1 0 0, δ2 = δ, δ3 = δn) and weights (w1, w2 and w3 respectively) and
iterative approximation under the condition δ1w1 = δ2w2 = δ3w3 The one-dimensional filter
is designed by defining the filter coefficient.
[0023]
FIG. 11 is a diagram in which a one-dimensional filter designed according to the Parks &
McClellan equiripple filter design algorithm and a design example of a one-dimensional filter
with Dolph-Chebyshev characteristics are described together. As apparent from FIG. 11, in the
former one-dimensional filter, the width of the passband is narrower than that of the latter by
increasing the ripple of the stopband 2. The effect of narrowing the width of the passband in this
manner becomes more remarkable as the property in which the number of ripples in the nonphysical region accounts for the total number of ripples in the stoppage region. Although it is
theoretically possible to set the amplitude of the ripple in the non-physical region to any large
value, practically it is necessary to set the upper limit value appropriately, for example, as the
upper limit value It is sufficient to set "1" (that is, a value equal to the amplitude of the passband),
"2" (value of twice the amplitude of the passband), or the like. By applying the two-dimensional
04-05-2019
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inverse discrete Fourier transform to the target amplitude characteristic of the two-dimensional
digital filter designed in this way, the filter coefficients to be set for each one-dimensional digital
filter forming the two-dimensional digital filter are calculated Be done. By setting the filter
coefficients calculated in this manner to each one-dimensional digital filter 120-i, the twodimensional digital filter formed by these one-dimensional digital filters is given the amplitude
characteristic shown in FIG. Become.
[0024]
(A-4: Effects of First Embodiment) As described above, the characteristics of the physical area
directly affect the directional characteristics, while the characteristics in the non-physical area do
not directly affect the directional characteristics. Therefore, in the speaker array 100 according
to the present embodiment, the level of the side lobe is kept low particularly in the low band as a
characteristic of the final filter coefficient by using a one-dimensional filter with two-stage stop
band etc. ripple characteristics. As it is, it is possible to reduce the width of the main lobe.
[0025]
Further, according to the present embodiment, by adjusting the one-dimensional filter optimally
according to f1, the width of the main lobe can be made constant while keeping the influence of
the side lobe low even in a band lower than the conventional one. It will be possible.
As described above, since the width of the main lobe depends on the number of ripples in the
non-physical area and the amplitude thereof, the amplitude to be set to the ripple of the nonphysical area so as to obtain necessary directivity characteristics according to f1 By adjusting the
number, it is possible to make the width of the main lobe constant in a lower band than in the
past.
[0026]
Further, in a region where the time frequency is relatively high (for example, a region defined by
fl f f1 in Non-Patent Document 2), the width of the main lobe is sufficiently narrowed without
increasing the amplitude of the ripple in the non-physical region. Since it is possible to use the
Dolph-Chebyshev characteristic disclosed in Non-Patent Document 2, for example, it may be used
in place of the stop zone two-step ripple characteristic. In addition, 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 for such
04-05-2019
11
a region, the characteristic improvement in the low range according to the present embodiment
can be achieved. It becomes possible to obtain directivity characteristics independent of time
frequency in a wider band.
[0027]
(B. Second Embodiment Next, a microphone array 200 according to a second embodiment of
the present invention will be described. FIG. 12 is a view showing an example of the arrangement
of a microphone array 200 according to the second embodiment of the present invention. As
apparent from comparison between FIG. 12 and FIG. 1, the point that the configuration of the
microphone array 200 is different from the configuration of the speaker array 100 is replaced
with the speaker 110-i (i: natural number of 1 to n) This is a point provided with a microphone
210-i (i: natural number of 1 to n) for outputting an audio signal corresponding to the collected
voice. In the microphone array 200, an 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 above-described filter processing is performed by each onedimensional 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 the addition result A signal is
output.
[0028]
Now, in the microphone array, the time-frequency characteristics of the plane wave coming from
the direction of the angle φ shown in FIG. 12 are one-dimensional connected to the respective
microphones (microphones 210-i in this embodiment) constituting the microphone array. It is
generally known that when the amplitude characteristics of the digital filter group are viewed in
a two-dimensional frequency plane, they are distributed on the straight line represented by the
above-mentioned equation (2). Therefore, by setting the filter coefficients described in the first
embodiment to each of the one-dimensional digital filters 120-i, the directivity characteristic of
the microphone array 200 is the same as that in the first embodiment described above. It is
possible to obtain an effect (that is, an effect of improving the directivity of the microphone array
in the low frequency range and avoiding an increase in the side lobe level without increasing the
array length).
[0029]
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12
(C. Modifications> Although the embodiment of the present invention has been described
above, it goes without saying that modifications as described below may be added to the abovedescribed embodiment. (1) In the embodiment described above, the case of forming the acoustic
beam symmetrical with respect to the central axis of the passband has been described, but it is
also possible to form the acoustic beam asymmetric with respect to the symmetry axis.
[0030]
(2) In the embodiment described above, 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 directional characteristics by frequency, the amplitude
characteristics to be applied to the two-dimensional digital filter (that is, set to each onedimensional digital filter 120-i The to-be-filtered factor may be determined in consideration of
the directivity characteristics of the transducer according to frequency. Such a thing is possible,
for example, by applying the same method as the method disclosed in “Seiki Nishikawa, Takaya
Osaki“ Directive Array Speaker Using a Two-Dimensional Digital Filter ”(1995)”.
[0031]
(3) In the embodiment described above, equal ripples with large amplitude are provided in the
non-physical region of the blocking region, and the physical region has a smaller amplitude than
ripples in the non-physical region (in the above embodiment, “δ = A description has been given
of the case where the amplitude characteristics of the two-step stop ripple characteristics such as
providing a ripple having 0.1 ") are given to the two-dimensional digital filter. However, ripples in
non-physical areas do not necessarily have to be equal ripples. For example, as shown in FIG. 13,
a ripple having an amplitude larger than that of the passband in the non-physical region stop
region (stop region 2 in FIG. 13) and an amplitude smaller than the ripple and within the physical
region It may be a multi-pass band ripple characteristic such as a multi-pass band in which a
ripple having an amplitude larger than the ripple of the non-blocking band (the blocking area 1
in FIG. 13) is provided. The point is that the frequency characteristics of the two-dimensional
digital filter of the speaker array and the microphone array according to the present invention
provide a plurality of ripples in the stop band and the ripple amplitude in the non-physical region
is higher than the ripple amplitude in the physical region It is sufficient if the frequency
characteristic is large.
04-05-2019
13
[0032]
(4) In the above embodiment, a case is described in which filter coefficients characteristic to 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 coefficients may be sequentially
calculated and set each time the speaker array or the microphone array is used. Thus, 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, directivity characteristics
according to the acoustic characteristics of the acoustic space, such as the size and shape of the
acoustic space Can be set properly. In addition, filter coefficients to be set to the respective onedimensional digital filters may be given from the outside of the speaker array or the microphone
array. Specifically, for the speaker array or the microphone array, for example, a communication
unit such as a network interface card (NIC) and a filter for setting the filter coefficients acquired
via the communication network using the communication unit to the respective one-dimensional
digital filters A coefficient setting means may be provided, and a reading means for reading data
from a computer-readable recording medium such as a CD-ROM (Compact Disk-Read Only
Memory) may be provided instead of the communication means. Of course, the filter coefficients
may be written on the recording medium and distributed, and the filter coefficients read by the
reading unit may be set in each one-dimensional digital filter by the filter coefficient setting unit.
[0033]
It is a block diagram showing the electric constitution of the speaker array 100 concerning a 1st
embodiment of the present invention. It is a figure which expressed an example of the amplitude
characteristic of the two-dimensional digital filter of the speaker array 100 using the twodimensional frequency plane. It is the figure which showed a part of same amplitude
characteristic with the equal amplitude characteristic figure. It is the figure which plotted the
amplitude characteristic of the same speaker array 100 for every predetermined angle. It is the
figure which plotted the directivity of the same 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 same speaker array 100, and the main lobe width of the acoustic beam. It
is a figure for demonstrating the design method currently disclosed by the nonpatent literature 2
about the cross-sectional characteristic in f1> = fl. It is a figure for demonstrating the design
method currently disclosed by the nonpatent literature 2 about the cross-sectional characteristic
in f1 <fl. 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 Parks & McClellan equal ripple filter design program. FIG.
8 is a diagram showing a design example of a one-dimensional filter and a design example of a
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one-dimensional filter with Dolph-Chebyshev characteristics by Parks & McClellan's equal ripple
filter design program. It is a figure which shows the electric constitution of the microphone array
200 which concerns on 2nd Embodiment of this invention. It is a figure which shows the
frequency characteristic which concerns on a modification (3).
Explanation of sign
[0034]
100 ... speaker array, 110-i (i: natural number of 1 to n) ... speaker, 120-i ... one-dimensional
digital filter, 200 ... microphone array, 210-i ... microphone.
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