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JP2006246007

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DESCRIPTION JP2006246007
The present invention provides a signal processing device and a microphone array system for a
microphone array that can pick up voice in a low frequency band using a small microphone
array. A delay unit (411-1 to 411-M) for adding a delay to a plurality of voice signals respectively
output from a plurality of microphones constituting a microphone array, and the plurality of
voices to which each delay is added A signal processing apparatus (4) having an adder (412) for
summing signals, a harmonic structure detection unit (421) for detecting a harmonic structure of
voice included in the voice signal, and a detected harmonic And a filter unit (422) for selectively
passing a predetermined frequency component based on the structure. [Selected figure] Figure 2
Microphone array signal processing apparatus and microphone array system
[0001]
The present invention relates to a signal processing apparatus and a microphone array system
for a microphone array in which a plurality of microphones are arranged in an arbitrary space.
[0002]
Conventionally, a microphone array in which a plurality of microphones are arranged in an
arbitrary space is configured, and after delay is added to the signals received by each
microphone, the directional processing is performed by performing array processing for
summing the signals. Has been proposed (see Patent Document 1 and Non-patent Document 1).
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Such array processing is called "delay-and-sum processing" or "DS (Delay-and-Sum) processing".
[0003]
Here, the principle of the DS processing is approximately as follows. Generally, as shown in FIG.
13, the microphone array system includes a microphone array consisting of M (M is a natural
number of 2 or more) microphones MICi (i is a natural number of 1 to M) and an audio signal xsi
output from each microphone. A delay unit which loads the delay amount Di to (t), and an adder
which sums the delayed voice signal xsi (t-Di). For the sake of simplicity, the microphone array
acting as a sound receiver is an equally spaced linear array microphone array in which M
microphones are arrayed at equal intervals on a straight line. Correct the time difference of voice
arriving at each microphone from the target direction (direction in which directional
characteristics are desired) ?L by giving an appropriate delay amount Di to the output voice
signal xsi (t) of each microphone, and make it in phase Can. On the other hand, voices coming
from directions other than the target direction ? L are not in phase by the above delay
operation. Therefore, when the delayed audio signal xsi (t-Di) is added, although the in-phased
signal is emphasized, the emphasizing effect is small for the non-in-phased signal. As a result,
directivity characteristics with high sensitivity to the target direction ?L are formed.
[0004]
According to Non-Patent Document 1, the directivity characteristic of the microphone array
system by the DS processing as described above can be expressed as follows. First, the amplitude
ratio between the array processing output y (t) and the array input xi (t), that is, the array gain G
is expressed by the following equation (1) and equation (2).
[0005]
G = | sin (?M / 2) / sin (? / 2) |... Equation (1) where: ? = 2?fd (sin ? L-sin ?) / c equation
(2) f: Frequency of audio signal d: Microphone spacing ?L: Target direction ?: Direction of voice
arrival c: Sound speed
[0006]
The directivity characteristic until the array gain G becomes zero (or sufficiently low gain) across
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the target direction ?L is called a main lobe, and the condition that the array gain G becomes
zero initially is the above equation ( 1) ? M / 2 = ? иииииииииии Equation (3).
When ? L = 0 (the target direction is the front of the microphone array), the angle ? 1 (main
lobe width) at which the array gain first becomes zero is expressed as follows from the above
equations (2) and (3) Ru. ? 1 = sin <?1> (c / fdM) (4) From the above equation (4), the main
lobe width decreases as the frequency f, the microphone distance d, and the number M of
microphones increase. I understand.
[0007]
According to Non-Patent Document 1, in general, regarding the directivity characteristics of the
DS microphone array, although the following can be said, they have common properties even if
they have an array shape other than the linear arrangement. (1) If the number of microphones M
and the microphone distance d are selected large and the array length Md is increased, sharp
directivity in the target direction can be realized. (2) The width of the main lobe is frequency
dependent (the higher the frequency, the sharper). (3) If the microphone distance d is d <c / 2f,
spatial folding of the main lobe does not occur.
[0008]
The applicant did not find by the time of filing the prior art documents related to the present
invention other than the prior art documents specified by the prior art document information
described in the present specification. Japanese Patent Application Laid-Open No. 9-140000
Japanese Patent Laid-Open No. 6-202627 Japanese Patent Laid-Open No. 9-251044 Togao
Ohga, Yoshio Yamazaki, Yutaka Kanada "Sound System and Digital Processing", Institute of
Electronics, Information and Communication Engineers (March 25, 1995 Issue), p. 181-186
[0009]
Due to the nature of the DS microphone system as described above, in order to obtain sharp
directivity characteristics even in a low frequency band, it is necessary to increase the entire
array length, which hinders miniaturization of the microphone array. In addition, when a small
microphone array is used, the directivity characteristic can not be made sharp enough, and there
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is a problem that a voice signal in a low frequency band is buried in another voice signal (noise)
from the periphery. The Therefore, an object of the present invention is to provide a signal
processing apparatus and a microphone array system for a microphone array that can collect
voice in a low frequency band even using a small microphone array.
[0010]
In order to achieve the above object, a signal processing apparatus for microphone array
according to the present invention comprises: delay means for adding a delay to each of a
plurality of audio signals respectively output from a plurality of microphones constituting the
microphone array; A predetermined frequency component is selected based on the detected
harmonic structure, and addition means for summing the plurality of audio signals added thereto,
detection means for detecting the harmonic structure of the sound included in the audio signal,
and And filter means for passing through. In the present invention, with respect to directivity
characteristics determined by the array length and frequency of the microphone array, with
respect to sufficiently high frequency components, necessary directivity characteristics are
obtained by delay-sum processing by the delay means and the addition means, while low
frequency components are obtained. Focuses on the harmonic structure of the audio signal, and
the filter means removes frequency components not related to the audio signal.
[0011]
Here, the detection means may detect the harmonic structure based on, for example, a basic pitch
extracted from the audio signal, but the temporal change of the spectrum of the audio signal, for
example, the spectrum of each harmonic structure The harmonic structure of the audio signal
coming from one sound source may be specified based on the timing of appearance or peak.
[0012]
Also, the filter means is realized by, for example, a comb filter that selectively passes frequency
components (basic pitch and harmonic components) of integral multiples of the basic pitch of the
audio signal among the audio signals output from the addition means. can do.
Therefore, for example, a high pass filter for passing high frequency components of the output of
the adding means, a comb filter for passing predetermined frequency components based on the
harmonic structure, and an output of the high pass filter By comprising output means that adds
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the output of the comb filter and the output, it is possible to selectively pass a predetermined
frequency component based on the detected harmonic structure.
[0013]
Further, the microphone array signal processing apparatus according to the present invention
further includes determination means for determining a sound source, and a predetermined
frequency component based on a harmonic structure of an audio signal coming from an arbitrary
sound source determined by the determination means. May be passed selectively.
[0014]
At this time, the discrimination of the sound source by the discrimination means can be
performed based on the harmonic structure of the audio signal and the frequency characteristic
of the delay-sum process by the delay means and the addition means.
For example, comparing the harmonic structure spectrum of the voice signal from the sound
source before and after the delay-and-summing process, when the sound source is located in the
target direction (center of directivity characteristics of the microphone array), both are almost
the same While the tendency is shown, when the sound source deviates from the target direction,
both tend to be different. Therefore, the sound source can be determined by comparing the
spectrum before and after the delay-and-sum processing for each harmonic structure.
[0015]
Further, in the microphone array system according to the present invention, there is provided a
microphone array comprising a plurality of microphones spatially arranged, and a signal
processing for microphone array processing audio signals respectively outputted from the
microphones constituting the microphone array. In the microphone array system including the
device, any one of the above-described signal processing devices for microphone array is used as
the signal processing device for the microphone array.
[0016]
According to the present invention, selectivity can be enhanced and noise can be suppressed
even for low frequency components for which a sharp directional characteristic could not be
realized conventionally, so that voices of low frequency bands can be obtained without increasing
the array length. It is possible to provide a signal processing device and a microphone array
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system for microphone array that enable sound collection.
[0017]
Hereinafter, embodiments of the present invention will be described with reference to the
drawings.
[0018]
First Embodiment FIG. 1 is a view showing an outline of a microphone array system according to
a first embodiment, and FIG. 2 is a view showing a configuration of a signal processing device of
the microphone array system.
This microphone array system, as shown in FIG. 1, includes M microphones 1-1 to 1-M
constituting the microphone array, and amplifiers 2-1 to 2- which respectively amplify audio
signals output from the respective microphones. M, A / D converters 3-1 to 3-M for A / D
converting amplified audio signals, and a signal processing device 4 for performing digital signal
processing on the A / D converted audio signals And consists of
The signal processing device 4 may also be realized by a computer having a storage device such
as a CPU (central processing unit), a ROM storing a program for controlling the signal processing
device, etc. and a RAM storing various calculation results by the CPU. It is possible.
Also, a dedicated signal processor (DSP) may be used instead of a general purpose CPU.
[0019]
As shown in FIG. 2, the signal processing device 4 includes a delay sum (DS) processing unit 41
and a filter processing unit 42. Among them, the DS processing unit 41 comprises delay units
411-1 to 411-M for adding a delay to each A / D converted audio signal, and an adder 412 for
adding the outputs of these delay units. The basic configuration and operation are the same as
those of the conventional DS processing unit.
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[0020]
The filter processing unit 42 is a filter that performs filter processing based on the harmonic
structure of the audio signal after DS processing output from the DS processing unit 41.
Specifically, the harmonic structure detection unit (pitch extraction unit) 421 and a filter unit
422. Here, the pitch extraction unit 421 extracts the basic pitch from the speech signal after DS
processing output from the DS processing unit 41 by a known pitch extraction method. In
addition, about the extraction method of a well-known pitch, for example, refer to patent
document 2, patent document 3 grade | etc.,. On the other hand, the filter unit 422 acts as a kind
of comb filter for passing only frequency components of integral multiples of the basic pitch
extracted by the pitch extraction unit 421 to a low frequency band, and other high values. For
the frequency band, it is a digital filter that is passed as it is. The frequency band to be operated
as a comb filter may be a frequency band in which sufficient directivity characteristics can not be
obtained by the DS processing. This band can be determined naturally according to the array
length of the microphone array.
[0021]
In the conventional microphone array system, if the array length can not be made sufficiently
large, if the frequency band is low, DS processing can not obtain sufficiently sharp directivity
characteristics. In the processed audio signal, in addition to the audio signal to be picked up, the
audio signal often includes wide band noise such as air conditioning and noise of a projector. On
the other hand, a voice to be picked up generally has a harmonic structure including a basic pitch
(basic frequency) and harmonic components that are integral multiples of the basic pitch.
Therefore, in the present embodiment, first, the pitch extraction unit 421 extracts the basic pitch
(basic frequency) included in the audio signal after DS processing output from the DS processing
unit 41, and the filter unit 422 extracts this basic pitch. By multiplying the basic pitch by an
integer, it is possible to detect a harmonic structure and to remove broadband noise by
performing filtering based on this harmonic structure.
[0022]
Next, the configuration of the above-described filter unit 422 will be described in detail with
reference to FIG. As illustrated in FIG. 3, in the signal processing device 4, the filter processing
unit 42 includes a pitch extraction unit 421, a comb filter 422 a, and a high pass filter (HPF) 422
b that extracts high frequency components from the output of the DS processing unit 41. It can
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be comprised from the adder 422c which adds the output of the comb filter 422a, and the output
of HPF422b. Here, the comb filter 422a is configured to pass frequency components that are
integral multiples of the basic pitch extracted by the pitch extraction unit 421. Therefore, only
the harmonic structure component of the audio signal output from the DS processing unit 41 is
output from the comb filter 422a. Such a comb filter 422a may be configured by a digital filter or
may be implemented in the frequency domain. On the other hand, the HPF 422 b is configured to
pass only the signal component in the high frequency band where sufficient directivity
characteristics can be obtained by the DS processing. Therefore, low frequency components
including wide band noise and the like in the audio signal output from the DS processing unit 41
are cut by the HPF 422 b, and only signal components in high frequency band where sufficient
directivity characteristics can be obtained are output.
[0023]
By adopting such a configuration, the microphone array system according to the present
embodiment performs only DS processing for high frequency components, and for low frequency
band signals for which sharp directivity characteristics can not be obtained by DS processing. ,
Filter processing based on harmonic structure. In particular, for high frequency components, the
output of the DS processing unit 41 is supplied by the HPF 422 b, so that it is possible to avoid
the loss of audio signals whose main energy is distributed in relatively high frequency bands
such as unvoiced consonants, for example. .
[0024]
As a modification of the present embodiment, as shown in FIG. 4, a low pass filter (LPF) 422d is
provided downstream of the comb filter 422a, and the output of the comb filter 422a is passed
through the LPF 422d and then the adder is added. It may be supplied to 422c. Note that such an
LPF 422 d may be provided in the front stage of the comb filter 422 a. At this time, it is more
desirable that the pass band of the LPF 422 d be a low frequency band in which a sufficient
directional characteristic can not be obtained by the DS processing, and the LPF 422 d and the
HPF 422 b complement each other. This makes it possible to suppress the deterioration of the
sound quality.
[0025]
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Second Embodiment Next, a second embodiment of the present invention will be described with
reference to FIG. In the first embodiment, the output of the DS processing unit 41 is used as the
input of the pitch extraction unit 421, and the basic pitch is extracted from the audio signal after
the DS processing. However, in the present embodiment, the DS processing is performed. The
basic pitch is extracted based on the previous signal. FIG. 5 is a diagram showing the
configuration of the signal processing device 4 in the microphone array system according to the
present embodiment. As shown here, the pitch extraction unit 421 may extract a basic pitch from
an audio signal after A / D conversion of an arbitrary one of M microphones constituting the
microphone array, Although not shown, a microphone for basic pitch extraction may be provided
separately from the microphone array. In the present embodiment, the configuration of the
microphone array excluding the signal processing device 4 is the same as that of the first
embodiment described above (see FIG. 1). Further, each component of the signal processing
device 4 is also the same as that of the first embodiment.
[0026]
Third Embodiment Next, a third embodiment of the present invention will be described with
reference to FIGS. 6 to 9. The same reference numerals are used for the same components as
those of the above-described conventional technology and the first embodiment, and the
description thereof will be appropriately omitted. In the microphone array system according to
the third embodiment of the present invention, even if the microphone array detects voices from
a plurality of sound sources as a result of failing to obtain sufficiently sharp directivity
characteristics, It comprises means for discriminating the sound source from the incoming
direction. FIG. 6 shows the configuration of the signal processing device 4 of the microphone
array system according to the present embodiment. In the present embodiment, the signal
processing device 4 includes a filter processing unit 52 including a pitch extraction unit 421, a
determination unit 521, and a filter unit 422. Among them, the pitch extraction unit 421 extracts
the basic pitch from the audio signal (in the present embodiment, the output signal of the DS
processing unit 41) as described in the first embodiment.
[0027]
The discrimination unit 521 compares the signal before and after DS processing for each
harmonic structure obtained from the basic pitch extracted by the pitch extraction unit 421, and
the voice having the basic pitch arrives from the target direction (?L). The basic pitch of the
voice coming from the target direction is output to the filter unit 422. The principle of the
discrimination of the sound source will be described later. The filter unit 422 acts as a kind of
04-05-2019
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comb filter for passing only frequency components of integer multiples of the basic pitch given
by the judging unit 521 for low frequency bands, and for the other high frequency bands. Is a
digital filter that is passed as it is. The characteristic is the same as that of the filter unit 422 in
the first embodiment.
[0028]
Next, the discrimination process of the sound source in the discrimination unit 521 will be
described with reference to FIGS. 7A to 9. (1) Direction of sound source and frequency
characteristic of DS processing The target direction ?L of the microphone array can be
determined by appropriately controlling each delay amount Di in DS processing, and the
directivity characteristic has frequency dependency , As described above (see, for example,
formulas (1) to (4) and the like). FIGS. 7A and 7B both show the frequency characteristics of the
audio signal after DS processing, and the former represents the case where the sound source is in
the target direction ?L and the latter represents the case where the sound source is out of the
target direction ?L. When the sound source is in the target direction ?L, a substantially flat
frequency characteristic is obtained over the entire frequency domain (FIG. 7A). On the other
hand, when the sound source deviates from the target direction ?L, although the flat
characteristic is exhibited in the low frequency region, a plurality of specific frequencies (these
The frequency varies depending on the number of microphones M, the microphone spacing d,
and the deviation ? from the target direction of the sound source. ) And the gain tends to be
small as a whole (FIG. 7B). Therefore, when the sound source is in the target direction ?L, the
sound before the DS processing is compared with the signal after the DS processing in the
frequency domain with respect to the sound coming from a certain sound source, at each peak
frequency constituting the harmonic structure. When the sound source deviates from the target
direction ?L, the level becomes different depending on the peak frequency while the level
becomes substantially equal.
[0029]
(2) Discrimination of sound source based on harmonic structure In a real environment, since a
plurality of signals from various sound sources are mixed, the above-mentioned specific sound
source can be obtained by simply comparing the signals before and after DS processing. It is
almost impossible to find such differences in frequency characteristics. Therefore, in the present
embodiment, focusing on the point that each sound source has a unique harmonic structure, the
signal before DS processing and DS processing are performed only for the position of the
harmonic series forming one harmonic structure. Compare with the later signal. Thus, if the
04-05-2019
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harmonic components are generated from the same sound source, the frequency characteristics
of the DS processing appear for those frequency components. Therefore, it becomes possible to
distinguish a plurality of sound sources by comparing the frequency characteristics of the DS
processing for each harmonic structure.
[0030]
A method of determining a sound source based on such a harmonic structure will be described
with reference to FIGS. 8 to 9. FIG. 8 is a diagram showing an example of a Fourier spectrum of
speech from a specific sound source. The horizontal axis is frequency, and the vertical axis is
intensity. As shown here, in general, speech existing in the natural world has a harmonic
structure, so its Fourier spectrum has peaks appearing at equal intervals at frequencies that are
integer multiples of the basic pitch (natural frequency). 9A and 9B show the difference between
the audio signal before DS processing and the audio signal after DS processing for the harmonic
component of the harmonic structure shown in FIG. 8 (hereinafter, simply ?envelope? the
frequency characteristic of the DS processing for the harmonic component) It is said. FIG. Among
these, FIG. 9A is an envelope when the sound source is in the target direction ?L, and FIG. 9B is
an example of an envelope when the sound source is deviated from the target direction ?L. In
the former case it can be seen that it takes approximately the same value for all overtone
components (i.e. it is flat), while in the latter case it takes different values, especially in the high
frequency range. Therefore, by obtaining the frequency characteristic by DS processing for each
harmonic structure having different basic pitches, it is possible to determine from the feature
whether or not the sound source having the harmonic structure is in the target direction ?L.
[0031]
As described above, in the present embodiment, the determination unit 521 determines the
sound source based on the harmonic structure, and only the harmonic structure of the sound
source in the target direction ?L can be applied to the filter unit 422, which is low. Also in the
frequency band, it is possible to extract an audio signal from the target direction ?L out of audio
signals from a plurality of sound sources collected by the microphone array.
[0032]
In the present embodiment, although it has been described that the discrimination is performed
based on the signal after one DS processing in which the target direction is ?L, other DS
processing in which the target directions are different is simultaneously performed. The same
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determination may be made for the processed signal.
In this case, when the sound source is in the target direction ?L, it is apparent that the envelope
based on the characteristics of the DS processing having different target directions does not
become flat. Therefore, it is possible to further improve the determination accuracy by acquiring
two or more envelopes having different target directions and actively utilizing the information
that the envelopes do not become flat.
[0033]
Further, in the present embodiment, as a method of specifying a harmonic structure for each
sound source from a signal in which sounds from a plurality of sound sources are mixed, the
pitch extraction unit 421 includes each sound signal by a known pitch extraction method. The
basic pitch to be generated may be extracted, but the harmonic structure of the voice coming
from one sound source may be specified based on the temporal change of the spectrum of the
voice signal. FIG. 10 is a diagram showing an example of temporal change of the spectrum of the
audio signal. The vertical axis represents frequency, and the horizontal axis represents time. In
FIG. 10, it is shown that the frequency spectra of speech from different sources (eg, speaker A,
speaker B) appear at different times with their harmonic structures. Here, speaker A starts
talking at time t1, and speaker B starts talking at time t2. Thus, in the harmonic structure
detection unit 421, the harmonic structure of each sound source is specified based on the
temporal change of the spectrum of the audio signal, for example, the appearance of the
spectrum indicating the harmonic structure, the timing of its peak, etc. You may
[0034]
Further, as a modification of the present embodiment, as shown in FIG. 11, the pitch extraction
unit 421 may be configured to extract the basic pitch based on the signal before the DS
processing. Further, a comb filter 422a may be provided instead of the filter unit 422, and the
output thereof may be added to the output of the HPF 422b.
[0035]
Fourth Embodiment The configuration of a signal processing apparatus according to a fourth
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embodiment of the present invention is shown in FIG. This signal processing apparatus omits the
filter section 422a and the HPF 422b in the filter processing section 52 of the signal processing
apparatus 4 shown in FIG. A sound source direction discrimination device is configured by
combining the filter processing unit 52 ? with the DS processing unit 41. In such a sound source
direction discrimination device, the signals before and after the DS processing are compared for
each harmonic structure obtained from the basic pitch extracted by the harmonic structure
detection unit 421, and the voice having the basic pitch is the target direction. It is determined
whether it has arrived from (?L). Therefore, even when there are a plurality of speakers, if the
harmonic structures of the sounds emitted from these speakers are different, it is possible to
specify the direction for each speaker. Although not shown, at this time, the target direction (?L)
at that time may be calculated from the delay amounts D1 to DM of the DS processing unit 41,
and this may be output.
[0036]
Further, in the present embodiment, the harmonic structure of the audio signal collected by the
microphone is specified using the harmonic structure detection unit 421. However, as a
modification, in place of the harmonic structure detection unit 421, A storage means such as a
memory is provided, the harmonic structure of the target sound source is stored, and the
directivity of the microphone array is changed to specify the target sound source direction. In
addition, if it is determined whether the sound source is in front of the microphone array, the
delay units 411-1 to 411 -M of the DS processing unit 41 become unnecessary.
[0037]
It is a figure showing an outline of a microphone array system concerning a 1st embodiment of
the present invention. It is a figure showing composition of a signal processing device of a
microphone array system concerning a 1st embodiment of the present invention. It is a figure
showing the signal processing device of the microphone array system concerning a 1st
embodiment of the present invention. It is a figure which shows the modification of the signal
processing apparatus of the microphone array system concerning the 1st Embodiment of this
invention. It is a figure which shows the structure of the signal processing apparatus of the
microphone array system concerning the 2nd Embodiment of this invention. It is a figure which
shows the structure of the signal processing apparatus of the microphone array system
concerning the 3rd Embodiment of this invention. It is a figure showing the frequency
characteristic of the audio | voice signal (when a sound source exists in object direction (theta) L)
after DS process. It is a figure showing the frequency characteristic of the audio | voice signal
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(when a sound source is not in target direction (theta) L) after DS process. It is a figure which
shows an example of the Fourier spectrum of an audio | voice. It is a figure which shows the
frequency characteristic (when a sound source exists in object direction (theta) L) of DS process
about a harmonic component. It is a figure which shows the frequency characteristic (when a
sound source is not in the objective direction (theta) L) of DS process about a harmonic
component. It is a figure showing an example of the time change of the spectrum of a speech
signal. It is a figure which shows the modification of the signal processing apparatus of the
microphone array system concerning the 3rd Embodiment of this invention. It is a figure which
shows the structure of the signal processing apparatus of the microphone array system
concerning the 4th Embodiment of this invention. It is a figure for demonstrating the
conventional microphone array system.
Explanation of sign
[0038]
1-1 to 1-M: microphone, 2-1 to 2-M: amplifier, 3-1 to 3-M: A / D converter, 4: signal processing
device, 41: delay sum processing unit, 411-1 411-M: delay unit, 412: adder, 42, 52, 52 ', ... filter
processing unit, 421: harmonic structure extraction unit (pitch extraction unit), 422: filter unit,
422a ... comb filter, 422b ... HPF , 422c... Adder, 422 d... LPF, 521.
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