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JP2009055343

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DESCRIPTION JP2009055343
PROBLEM TO BE SOLVED: To use a plurality of microphones that generate sound signals based
on received sound, to eliminate the influence of individual differences among the microphones,
and to cope with the aging of the characteristics of the microphones. An apparatus, a phase
difference correction method, and a computer program are provided. A sound processing
apparatus (1) converts the plurality of sound signals based on respective sounds received by a
plurality of sound receiving units (14a, 14b) such as microphones into signals on a frequency
axis; Calculation means 122 for calculating the spectral ratio of each sound signal converted to a
signal on the frequency axis by the FFT conversion means 121 and one sound signal converted
based on the spectrum ratio calculated by the calculation means 122 It comprises: calculating
means 123 for calculating the correction value of the phase of the converted other sound signal,
and correction means 124 for correcting the phase of the converted other sound signal based on
the correction value calculated by the calculating means 123. [Selected figure] Figure 3
Sound processing apparatus, phase difference correction method and computer program
[0001]
The present invention provides a sound processing apparatus including a plurality of sound
receiving units that generate sound signals based on received sound, and processing each sound
signal generated by the plurality of sound receiving units, and the sound processing apparatus.
The present invention relates to a phase difference correction method and a computer program
for realizing the sound processing device, and more particularly to a sound processing device for
correcting a phase difference of a sound signal caused by individual differences among a
plurality of sound receiving units, a phase difference correction method and a computer program
About.
04-05-2019
1
[0002]
Various sound processing apparatuses have been developed and put into practical use that
perform processing such as specifying the direction of arrival of sound using a plurality of
microphones.
An example of a sound processing apparatus using a plurality of microphones will be described.
FIG. 11 is a perspective view showing an outline of the sound processing apparatus. In FIG. 11,
reference numeral 1000 denotes a sound processing apparatus using a mobile phone. The sound
processing apparatus 1000 includes a rectangular parallelepiped casing 1001. A first
microphone 1002 is disposed on the front of the housing 1001 to receive voices uttered by the
speaker. A second microphone 1003 is disposed on the bottom of the housing 1001.
[0003]
Sounds come from various directions to the sound processing apparatus 1000, and the sound
processing apparatus 1000 is based on the phase difference corresponding to the time difference
of the sound that has reached the first microphone 1002 and the second microphone 1003.
Identify the direction of arrival of the sound. Then, the sound processing apparatus 1000 forms a
desired directional pattern by performing processing such as suppression of the sound received
by the first microphone 1002 according to the arrival direction.
[0004]
The plurality of microphones used in the sound processing apparatus 1000 as illustrated in FIG.
11 are required to have the same characteristics such as sensitivity. FIG. 12 is a radar chart
showing measurement results of the directional characteristics of the sound processing
apparatus 1000. The radar chart shown in FIG. 12 indicates the signal intensity (dB) after
suppression of the sound received by the first microphone 1002 of the sound processing
apparatus 1000 for each direction of sound arrival. Note that the situation where sound comes
from the front direction where the first microphone 1002 of the case 1001 of the sound
processing apparatus 1000 is disposed is 0 degrees, the situation that sounds from the direction
of the right side face is 90 degrees, the situation coming from the back direction Is 180 degrees,
and the situation coming from the direction of the left side is 270 degrees. In FIG. 12, the solid
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line indicates the state 1 in which the sensitivity of the first microphone 1002 and the second
microphone 1003 is the same, and the broken line indicates the state 2 in which the first
microphone 1002 has higher sensitivity than the second microphone 1003. The alternate long
and short dash line indicates that the sensitivity of the second microphone 1003 is higher than
that of the first microphone 1002. If the desired directivity is in state 1 in which the sensitivities
of the first microphone 1002 and the second microphone 1003 are the same, in states 2 and 3 in
which the sensitivities of the first microphone 1002 and the second microphone 1003 are
different, There are variations in the directional characteristics in the rear.
[0005]
If there are individual differences in the microphones as shown in FIG. 12, the characteristics of
the sound processing apparatus will be affected. However, in microphones generally
manufactured, individual differences such as sensitivity differences exist within certain
standards. Therefore, methods have been proposed in which teacher signals are generated from
positions at equal distances for a plurality of microphones, and adjustment is made so that the
characteristics of the microphones match (for example, see Patent Document 1 and Patent
Document 2). JP-A-2002-99297 JP-A-2004-343700
[0006]
However, in the method of adjusting individual differences in advance based on teacher signals
generated from equidistant positions as disclosed in Patent Document 1 and Patent Document 2,
for each set of microphones included in the sound processing apparatus, That is, since it is
necessary to adjust for each sound processing device, there is a problem that the cost at the time
of production increases. In addition, since it is not possible to cope with individual differences in
aging of the microphones, there is a problem that the characteristics of the microphones differ
after shipment.
[0007]
The present invention has been made in view of the above circumstances, and calculates the
correction value of another sound signal based on one sound signal from the spectral ratio based
on each sound signal, and uses the calculated correction value for the other. Since the sound
signal can be corrected at the time of use of the device by correcting the phase of the sound
signal, a sound processing device capable of coping with aging and suppressing an increase in
production cost. It is an object of the present invention to provide a phase difference correction
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method using a processing device, and a computer program for realizing the sound processing
device.
[0008]
A sound processing apparatus according to a first aspect of the present invention includes a
plurality of sound receiving units that generate a sound signal based on received sound, and the
sound processing apparatus that processes each sound signal generated by the plurality of sound
receiving units. A converting unit for converting a plurality of sound signals based on respective
sounds received by the plurality of sound receiving units into signals on a frequency axis; and
respective sound signals converted to signals on a frequency axis by the converting unit A
calculation unit for calculating the spectral ratio of the second sound signal, and a calculation
unit for calculating a correction value of the phase of the converted other sound signal based on
the converted sound signal based on the spectral ratio calculated by the calculation unit; And a
correction unit configured to correct the phase of another converted sound signal based on the
correction value calculated by the calculation unit.
[0009]
A sound processing apparatus according to a second aspect of the invention is characterized in
that, in the first aspect, the calculation unit is configured to calculate a power spectrum ratio of
each sound signal converted into a signal on a frequency axis. .
[0010]
The sound processing apparatus according to the third invention is characterized in that, in the
second invention, the calculation unit is configured to calculate the correction value based on the
following equation (A).
Pcomp (ω) = α · F {S2 (ω) / S1 (ω)} + β (Equation (A) where ω: angular frequency Pcomp (ω):
correction value of phase S1 (ω): one sound signal Power spectrum S2 (ω): Power spectrum of
another sound signal α, β: constant F (): function
[0011]
The sound processing apparatus according to the fourth invention is characterized in that in the
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second invention, the calculation unit is configured to calculate the correction value based on the
following equation (B).
Pcomp (ω) = [α · F {S1 (ω) / S2 (ω)}] · ω + β (B) where ω: angular frequency Pcomp (ω):
phase correction value S1 (ω): one Power spectrum of sound signal S2 (ω): Power spectrum of
other sound signal α, β: constant F (): function
[0012]
In the sound processor according to the fifth invention, in the third invention or the fourth
invention, the function is a logarithmic function, and the correction unit adds a correction value
to the phase of another sound signal converted. It is characterized in that it is configured.
[0013]
The sound processing apparatus according to the sixth invention is characterized in that, in the
first invention, the calculation unit is configured to calculate the amplitude spectrum ratio of
each sound signal converted into the signal on the frequency axis. .
[0014]
The sound processing apparatus according to a seventh aspect of the present invention is the
sound processing apparatus according to any one of the first to sixth aspects, further comprising
a smoothing section that smooths the temporal change of the correction value calculated by the
calculating section. It is characterized in that the smoothing unit is configured to perform
correction based on the smoothed correction value.
[0015]
A phase difference correction method according to an eighth aspect of the present invention is a
phase difference correction method for correcting, using a computer, phase differences of
respective sound signals generated by a plurality of sound receiving units that generate sound
signals based on received sound. A step of converting a plurality of sound signals based on
respective sounds received by the plurality of sound receiving units into signals on the frequency
axis, and sound signals converted to signals on the frequency axis by the conversion unit
Calculating the correction value of the phase of another converted sound signal on the basis of
the converted one sound signal based on the spectrum ratio calculated by the calculation unit;
And correcting the phase of the converted other sound signal based on the calculated correction
value.
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[0016]
A computer program according to a ninth aspect of the present invention is loaded on a
computer to define a procedure to be executed on the computer, and each of a plurality of sound
receiving units that generate a sound signal based on the sound received is generated. A
computer program for causing the computer to correct the phase difference of the sound signals
in the computer, and converting the plurality of sound signals based on respective sounds
received by the plurality of sound receiving units into signals on the frequency axis in the
computer A step of calculating the spectral ratio of each sound signal converted to a signal on
the frequency axis by the converter, and the converted one sound signal as a reference based on
the spectral ratio calculated by the calculator Execute a procedure of calculating a correction
value of the phase of another sound signal and a procedure of correcting the phase of the
converted other sound signal based on the correction value calculated by the calculation unit And
wherein the door.
[0017]
In the present invention, the correction value of another sound signal based on one sound signal
is calculated from the spectral ratio based on each sound signal generated by the plurality of
sound receiving units, and the calculated correction value is used to calculate the other
correction value. At the time of production, since there is no need to adjust individual differences
for each set of sound receiving units in order to correct the sound signal appropriately by the
sound processing device in use by correcting the phase of the sound signal, it is possible to It is
possible to suppress the increase in costs of the equipment, and even if individual differences
occur in the secular change of each sound receiving unit after shipment, by correcting the sound
signal each time, the difference in characteristics due to the secular change is absorbed. It is
possible.
[0018]
A sound processing apparatus, a phase difference correction method, and a computer program
according to the present invention generate a sound signal based on sound received by each of a
plurality of sound receiving units such as microphones, and generate the generated sound signals
on a frequency axis The spectral ratio of the converted sound signal is calculated, and based on
the calculated spectral ratio, the correction value of the phase of another sound signal converted
based on the converted one sound signal is calculated and calculated. The phase of the converted
other sound signal is corrected based on the correction value.
[0019]
In the present invention, based on the experimental result that the waveform of the microphone
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with low sensitivity leads in phase to the waveform of the microphone with high sensitivity, the
sensitivity is substituted for the spectrum, and the correction value of the phase difference is
calculated according to the spectral ratio.
[0020]
According to this configuration, according to the present invention, it is possible to appropriately
correct the sound signal by the sound processing device in use.
For this reason, since it is not necessary to adjust the sensitivity difference for each set of sound
receiving units at the time of production, it is possible to suppress an increase in the cost at the
time of production, and thus it is possible to obtain excellent effects.
Moreover, according to the present invention, even if the sensitivity difference occurs in the
secular change of each sound receiving unit, it is possible to absorb the difference in the
characteristics due to the secular change by performing the correction of the sound signal each
time. Play.
[0021]
Hereinafter, the present invention will be described in detail based on the drawings showing the
embodiments.
[0022]
Embodiment 1
FIG. 1 is a perspective view showing an example of the outer shape of a sound processing device
according to Embodiment 1 of the present invention.
In FIG. 1, reference numeral 1 denotes a sound processing apparatus according to the present
invention using a computer such as a mobile phone. The sound processing apparatus 1 includes
a rectangular parallelepiped casing 10.
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On the front of the case 10, a first sound receiving unit 14a using a microphone such as a
condenser microphone is disposed to receive voices uttered by the speaker.
Further, on the bottom of the case 10, a second sound receiving unit 14b using a microphone
such as a condenser microphone is disposed.
Sounds come from various directions to the sound processing device 1, and the sound processing
device 1 is based on the phase difference corresponding to the time difference to reach the first
sound receiving unit 14a and the second sound receiving unit 14b. To estimate the direction of
arrival of the sound.
Then, the sound processing device 1 forms a desired directivity characteristic by performing
processing such as suppression of the sound received by the first sound receiving unit 14 a
according to the arrival direction.
In the following description, the first sound receiving unit 14 a and the second sound receiving
unit 14 b will be described as the sound receiving unit 14 unless it is necessary to distinguish
them.
[0023]
FIG. 2 is a block diagram showing an example of a hardware configuration of the sound
processing apparatus 1 according to Embodiment 1 of the present invention.
In FIG. 2, reference numeral 1 denotes a sound processing apparatus according to the present
invention using a computer such as a mobile phone. The sound processing apparatus 1 includes
a control unit 11 such as a CPU for controlling the entire apparatus and a computer program
100 according to the present invention. A recording unit 12 such as a ROM and a RAM for
recording data such as programs and various setting values is provided, and a communication
unit 13 such as an antenna serving as a communication interface and an accessory thereof. In
addition, the sound processing device 1 converts a plurality of sound receiving units 14 and 14
such as microphones and the like that generate an analog sound signal by receiving an external
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sound, a sound output unit 15 such as a speaker, and conversion processing of sound signals. A
sound converter 16 is provided. Furthermore, the sound processing device 1 includes an
operation unit 17 that receives an operation by key input such as alphanumeric characters and
various commands, and a display unit 18 such as a liquid crystal display that displays various
information. In addition, although the form which the sound processing apparatus 1 equips with
the two sound receiving parts 14 and 14 is demonstrated here, you may provide not only this but
three or more sound receiving parts 14, 14, ... in this invention. Then, a computer such as a
mobile phone operates as the sound processing apparatus 1 of the present invention by causing
the control unit 11 to execute various procedures included in the computer program 100 of the
present invention.
[0024]
FIG. 3 is a functional block diagram showing an example of functions of the sound processing
device 1 according to the first embodiment of the present invention. The sound processing
apparatus 1 according to the present invention includes an LPF (a first sound receiving unit 14a
and a second sound receiving unit 14b) to prevent aliasing errors when converting a sound
signal that is an analog signal into a digital signal. And an A / D conversion means 161 for
converting a sound signal which is an analog signal into a digital signal. The first sound receiving
unit 14a and the second sound receiving unit 14b include an amplifier (not shown) that amplifies
a sound signal that is an analog signal. The anti-aliasing filter 160 and the A / D conversion
means 161 are functions realized by the sound conversion unit 16. The anti-aliasing filter 160
and the A / D conversion means 161 may not be built in the sound processing apparatus 1 as the
sound conversion unit 16 but may be mounted on an external sound capturing device together
with the sound receiving units 14, 14 is there.
[0025]
Furthermore, the sound processing apparatus 1 of the present invention comprises a frame
generation unit 120 for generating a frame of a predetermined time length, which is a unit of
processing, from a sound signal, and FFT (Fast Fourier Transformation) processing of the sound
signal. The second sound receiver 14b receives the FFT conversion means 121 for converting to
the above signal, the calculation means 122 for calculating the power spectrum ratio of each
sound signal converted to the signal on the frequency axis, and the spectrum ratio. The
calculating means 123 for calculating the correction value of the phase of the sound signal, the
correcting means 124 for correcting the phase of the sound signal received by the second sound
receiving unit 14b based on the correction value, and the first sound receiving unit 14a And
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sound processing means 125 for performing processing such as suppression of the sound
produced. The frame generation unit 120, the FFT conversion unit 121, the calculation unit 122,
the calculation unit 123, the correction unit 124, and the sound processing unit 125 indicate
functions as software realized by executing various computer programs in the recording unit 12.
However, it may be realized using dedicated hardware such as various processing chips.
[0026]
Next, the theory of the sound processing device 1 according to the first embodiment of the
present invention will be described. The sound processing apparatus 1 according to the present
invention performs the processing by the sound processing unit 125 based on the sound
received by the first sound receiving unit 14a and the second sound receiving unit 14b as the
first sound receiving unit 14a. And the process of correcting the phase to absorb individual
differences such as the sensitivity difference of the second sound receiving unit 14b. First, the
influence of the sensitivity difference between the first sound receiving unit 14a and the second
sound receiving unit 14b on the phase will be described.
[0027]
FIG. 4 is a graph showing the change in the sound waveform due to the difference in sensitivity
of the microphones. FIG. 4 is a graph showing the time change of the waveform of the sound
received by the microphone used as the sound receiving unit 14 of the sound processing
apparatus 1 of the present invention, the sample value is taken on the horizontal axis, and the
sound is output on the vertical axis. The amplitude value of the signal is taken to indicate the
relationship. The sample value is a value indicating the order of samples of the sound signal
sampled at a period of 96 kHz or the like. FIG. 4 shows a recorded sound (impulse response)
when an impulse sound is received using the same kind of microphones having different
sensitivities. In FIG. 4, a solid line indicates a change relating to a high sensitivity microphone,
and a broken line indicates a change relating to a low sensitivity microphone. As apparent from
the comparison between the peaks of the solid line and the broken line in FIG. 4, the sound signal
by the microphone with high sensitivity shown by the solid line has a large waveform swing up
and down compared with the sound signal by the microphone with low sensitivity shown by the
broken line . Furthermore, the sound signal from the microphone with lower sensitivity changes
at an earlier timing than the sound signal from the microphone with high sensitivity. That is, the
sound signal from the low sensitivity microphone leads the phase of the sound signal from the
high sensitivity microphone.
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[0028]
The relationship between the sensitivity difference and the advance of the phase will be
described using the relationship between the electrical system and the mechanical equivalent
circuit. FIG. 5 is a circuit diagram showing an equivalent circuit of the microphone. FIG. 5 shows
an equivalent circuit of a microphone such as a condenser microphone used as the sound
receiving unit 14 of the sound processing apparatus 1 of the present invention, and a capacitor
having a capacitance of C with respect to the output terminal and a resistance of R are shown.
The resistors are connected in parallel. The behavior of the output voltage value after the
condenser microphone is pressed by the change in sound pressure from the outside is equivalent
to the damped oscillation of the spring constant K (= 1 / C) in which the resistance value R works.
Here, in the equivalent circuit shown in FIG. 5, it is assumed that the equation of motion of the
spring vibration shown in the following equation (1) holds.
[0029]
[0030]
The solution of equation (1) above for x is equation (2) below.
[0031]
[0032]
The above equation (2) can be transformed into the following equation (3).
[0033]
[0034]
FIG. 6 is a graph showing the change of the output voltage value based on the equation of
motion.
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FIG. 6 is a graph showing the time change of the output voltage x based on the equation (3).
The solid line shows the time change of the theoretical value of the output voltage x when the
resistance value is small with R = 0.04, ω <2> = 0.026, and the broken line shows R = 0.05, ω <2
The time change of the theoretical value of the output voltage x in the case of a large resistance
value is shown as 2> = 0.026.
From the equation (3) and the graph of FIG. 6, the change in the output voltage x when the
resistance value R shown by the broken line is large is e compared to the change in the output
voltage x when the resistance value R shown by the solid line is small. The amplitude shown as <Rt>, that is, the maximum value of the output voltage x is small, and the entire waveform is
shifted earlier in time.
That is, when the resistance value R is large, the output voltage x has a large amplitude and the
phase advances.
Assuming that the amplitude of the output voltage x corresponds to the sensitivity of the
microphone, when a plurality of microphones with different sensitivity differences are used, the
sound signal from the microphone with low sensitivity leads in phase to the sound signal from
the microphone with high sensitivity. This corresponds to the impulse response experiment
shown using FIG.
[0035]
As described above, since the sensitivity difference of the microphone can be confirmed by the
amplitude related to the sound signal, and the sensitivity difference affects the phase, the sound
processing apparatus 1 of the present invention can set the value of the power spectrum
corresponding to the amplitude. By correcting the phase based on the above, the influence of the
sensitivity difference between the sound receiving units 14 and 14 is suppressed.
[0036]
Next, processing of the sound processor 1 according to the first embodiment of the present
invention will be described.
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FIG. 7 is a flowchart showing an example of processing of the sound processing device 1
according to Embodiment 1 of the present invention. The sound processing device 1 generates a
sound signal, which is an analog signal, based on the respective sounds received by the plurality
of sound receiving units 14 and 14 under the control of the control unit 11 that executes the
computer program 100 (S101) And anti-aliasing filter 160, and A / D conversion means 161
converts it into a digital signal.
[0037]
The sound processing device 1 generates frames of a predetermined time length as a processing
unit from the respective sound signals converted into digital signals by the processing of the
frame generation unit 120 based on the control of the control unit 11 (S102). In step S102, the
acoustic signal is framed in units of a predetermined time length of, for example, about 20 ms to
40 ms. Each frame is shifted by about 10 ms to about 20 ms to advance the processing.
[0038]
The sound processing device 1 performs processing of the sound signal of a frame unit by FFT
(Fast Fourier Transformation) processing by the processing of the FFT conversion means 121
based on the control of the control unit 11 into spectrums which are signals on the frequency
axis. It converts (S103). In step S103, conversion into a phase spectrum and an amplitude
spectrum is performed. In the following processing, a power spectrum that is the square of the
amplitude spectrum is used. Although an example using a power spectrum is shown here, the
following processing may be performed using an amplitude spectrum.
[0039]
The sound processing device 1 receives the sound received by the second sound receiving unit
14 b for the power spectrum of the sound signal based on the sound received by the first sound
receiving unit 14 a by the processing of the calculation unit 122 based on the control of the
control unit 11. The ratio of the power spectrum based on is calculated (S104). In step S104, the
value of the ratio is calculated for each frequency value of each power spectrum using the
following equation (4).
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[0040]
S2 (ω) / S1 (ω) equation (4) where ω: angular frequency S1 (ω): power spectrum based on the
sound signal of the first sound receiving portion 14a S2 (ω): of the second sound receiving
portion 14b Power spectrum based on sound signal
[0041]
The sound processing device 1 performs processing of the calculation unit 123 based on the
control of the control unit 11, based on the power spectrum ratio shown in Expression (4), based
on the sound signal on the frequency axis related to the first sound receiving unit 14a. The
correction value of the phase of the sound signal on the frequency axis related to the second
sound receiving unit 14b is calculated (S105).
In step S105, the correction value is calculated using the following equation (5).
[0042]
Pcomp (ω) = [α · F {S1 (ω) / S2 (ω)}] · ω + β (5) where Pcomp (ω): correction value of phase
α, β: constant F (): function
[0043]
The method of determining the constants α and β in the equation (5) will be described.
First, among the microphones of the type (form) used as the sound receiving unit 14, two
combinations of microphone sets of the combination of the microphone with the highest
sensitivity and the microphone with the lowest sensitivity and the combination of microphones
with the same sensitivity. Prepare the configured adjustment device. And white noise is
reproduced from the position which becomes equal distance to each microphone set, phase
difference spectrum (φ2 (ω)-φ1 (ω) of each microphone is calculated, phase difference
spectrum of the microphone set where sensitivity differs However, the constants α and β are
determined so as to fit in the phase difference spectrum of the microphone set of the same
combination of sensitivity. Then, the calculated constants α and β are recorded in the recording
unit 12 of the sound processing device 1 and the sound receiving units 14 and 14 are configured
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by using the same type of microphone as the microphone used for the adjustment. Processing
becomes possible. Further, as the function F () in the equation (5), a properly selected function
such as a common logarithm, a logarithmic function such as a natural logarithm, or a sigmoid
function is used.
[0044]
The sound processing device 1 adds the correction value of the phase calculated in step S105 to
the phase of the sound signal on the frequency axis related to the second sound receiving unit
14b by the processing of the correction unit 124 based on the control of the control unit 11.
Then, the sound signal related to the second sound receiving unit 14b is corrected (S106). In step
S106, the sound signal is corrected using the following equation (6).
[0045]
φ2 ′ (ω) = φ2 (ω) + Pcomp (ω) Equation (6) where φ2 (ω): phase spectrum based on the
sound received by the second receiver 14 b φ2 ′ (ω): after correction Phase spectrum
[0046]
Then, the sound processing device 1 performs processing of the sound processing unit 125
based on the control of the control unit 11 based on the sound signal related to the first sound
receiving unit 14 a and the sound signal corrected in phase related to the second sound receiving
unit 14 b. , And various sound processing such as suppression of the sound received by the first
sound receiving unit 14a (S107).
[0047]
Formula (5) used in step S105 can be suitably changed according to the shape of the sound
processing apparatus 1, and the content of acoustic processing.
For example, the following equation (7) can be used in place of the equation (5).
[0048]
Pcomp (ω) = α · F {S2 (ω) / S1 (ω)} + β formula (7)
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[0049]
Formula (5) is suitable for correction of the phase spectrum in the sound processing apparatus 1
in which the first sound receiving unit 14a and the second sound receiving unit 14b are arranged
in the vertical direction as shown in FIG. 1 in the normal operation state. Equation (7) is suitable
for correcting the phase spectrum in the sound processing apparatus 1 in which the first sound
receiving unit 14a and the second sound receiving unit 14b are disposed in the left-right
direction.
However, it is desirable to appropriately consider the equation to be applied depending on the
arrangement.
[0050]
When the phase of the sound signal related to the first sound receiving unit 14a is corrected
instead of correcting the phase of the sound signal related to the second sound receiving unit
14b, the function F in the equation (5) or (7) The denominator and the numerator of equation (1)
may be exchanged, but the phase of the sound signal relating to the first sound receiving unit
14a may be corrected using equation (8) below instead of equation (6). good.
[0051]
φ1 ′ (ω) = φ1 (ω) −Pcomp (ω) equation (8) where φ1 (ω): phase spectrum based on the
sound received by the first sound receiving unit 14a φ1 ′ (ω): after correction Phase spectrum
of
[0052]
Next, the correction result of the sensitivity difference by the sound processing device 1
according to the first embodiment of the present invention will be described.
FIG. 8 is a radar chart showing an example of the correction result of the sensitivity difference by
the sound processing apparatus 1 according to the first embodiment of the present invention.
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In FIG. 8, as the sound processing of the sound processing unit 125 included in the sound
processing device 1, the arrival direction of the sound is specified based on the phase difference
between the sounds received by the first sound receiving unit 14 a and the second sound
receiving unit 14 b. The directivity characteristic formed by performing processing such as
suppression of the sound received by the first sound receiving unit 14a according to the arrival
direction is shown.
The directional characteristics shown in the radar chart of FIG. 8 indicate the signal strength (dB)
after acoustic processing on the sound received by the first sound receiving unit 14 a for each
direction of arrival of the sound. The situation where the sound comes from the front direction
where the first sound receiving unit 14a of the case 10 of the sound processing device 1 is
disposed is 0 degrees, the situation that the sound comes from the direction of the right side is
90 degrees, the situation from the back direction The situation is 180 degrees, and the situation
coming from the direction of the left side is 270 degrees. FIG. 8A shows directivity characteristics
when the sensitivity difference between the first sound receiving unit 14a and the second sound
receiving unit 14b is not corrected, and the solid lines indicate the first sound receiving unit 14a
and the second sound receiving unit 14a. The state 1 in which the sensitivity of the sound
receiving unit 14b is the same is indicated, and the broken line indicates a state 2 in which the
sensitivity of the first sound receiving unit 14a is higher than that of the second sound receiving
unit 14b. The part 14b shows a state in which the sensitivity is higher than that of the first sound
receiving part 14a. FIG. 8 (b) shows the directivity characteristics when the sensitivity difference
is corrected by the sound processing apparatus 1 of the present invention, and the solid lines
indicate the sensitivity of the first sound receiving unit 14a and the second sound receiving unit
14b. The same state 1 is shown, and the broken line indicates a state 2 in which the first sound
receiving unit 14a has a higher sensitivity than the second sound receiving unit 14b. The
alternate long and short dash line indicates that the second sound receiving unit 14b is first. It
shows a state in which the sensitivity is higher than that of the sound receiving unit 14a.
[0053]
In FIG. 8A, the sensitivity of the first sound receiving unit 14a and the second sound receiving
unit 14b is different compared to the state 1 in which the sensitivities of the first sound receiving
unit 14a and the second sound receiving unit 14b are the same. 2 and 3 have variations in the
directional characteristics in the side and rear. On the other hand, in FIG. 8B, the influence of the
sensitivity difference between the states 2 and 3 is eliminated, and the directivity characteristics
of the states 2 and 3 approximate the state 1 in all directions.
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[0054]
In the first embodiment, the form according to the sound processing apparatus including the two
sound receiving units is shown, but the present invention is not limited thereto, and may be
applied to a sound processing apparatus including three or more sound receiving units. It is. In
the case of a sound processing apparatus provided with three or more sound receiving units,
calculation of power spectral ratio with respect to each sound signal related to other plural sound
receiving units with reference to a sound signal related to one sound receiving unit, By
performing the calculation of the phase correction value and the phase correction process, it is
possible to suppress the sensitivity difference.
[0055]
Second Embodiment The second embodiment is an improvement of the sound processing
apparatus according to the first embodiment from the viewpoint of reduction of processing load,
prevention of abrupt change in sound quality, and the like. Since the external shape of the sound
processing apparatus according to the second embodiment and the configuration example of the
hardware are the same as those of the first embodiment, the first embodiment shall be referred
to, and the description thereof will be omitted. In the following description, the same components
as in the first embodiment will be described with the same reference numerals as in the first
embodiment.
[0056]
FIG. 9 is a functional block diagram showing an example of functions of the sound processor 1
according to Embodiment 2 of the present invention. The sound processing apparatus 1 of the
present invention includes a first sound receiving unit 14a and a second sound receiving unit
14b, an anti-aliasing filter 160, and an A / D conversion unit 161 that performs A / D conversion.
The first sound receiving unit 14a and the second sound receiving unit 14b include an amplifier
(not shown) that amplifies a sound signal that is an analog signal.
[0057]
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The sound processing apparatus 1 according to the present invention includes a frame
generation unit 120, an FFT conversion unit 121, a calculation unit 122 that calculates a power
spectrum ratio, a calculation unit 123 that calculates a phase correction value, a correction unit
124, A sound processing means 125, frequency selection means 126 for selecting a frequency to
be used for calculation of the power spectrum ratio by the calculation means 122, and smoothing
means 127 for smoothing the time change of the correction value calculated by the calculation
means 123; Is equipped. The frame generation unit 120, the FFT conversion unit 121, the
calculation unit 122, the calculation unit 123, the correction unit 124, the sound processing unit
125, the frequency selection unit 126, and the smoothing unit 127 execute various computer
programs in the recording unit 12. Although the function as software realized by the above is
shown, it may be realized using dedicated hardware such as various processing chips.
[0058]
Next, processing of the sound processor 1 according to the second embodiment of the present
invention will be described. FIG. 10 is a flowchart showing a process example of the sound
processing device 1 according to the second embodiment of the present invention. The sound
processing device 1 generates a sound signal, which is an analog signal, based on the respective
sounds received by the plurality of sound receiving units 14 and 14 under the control of the
control unit 11 that executes the computer program 100 (S201 And anti-aliasing filter 160, and
A / D conversion means 161 converts it into a digital signal.
[0059]
The sound processing apparatus 1 generates a frame of a predetermined time length as a
processing unit from each sound signal converted into a digital signal by the processing of the
frame generation unit 120 based on the control of the control unit 11 (S202). By the processing
of the FFT conversion means 121 based on the control of the unit 11, the sound signal in units of
frames is converted into a spectrum which is a signal on the frequency axis by FFT processing
(S203).
[0060]
The sound processing apparatus 1 performs SNR (signal-to-noise ratio) for each frequency within
a frequency band such as 1000 to 3000 Hz not affected by the anti-aliasing filter 160 by the
processing of the frequency selection unit 126 based on the control of the control unit 11. A
frequency at which the ratio: Signal to Noise Ratio) is equal to or higher than a preset setting
value is selected (S204).
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[0061]
The sound processing device 1 calculates the power spectrum ratio for each frequency selected
in step S204 by the processing of the calculation means 122 based on the control of the control
unit 11 (S205), and calculates the average value of the calculated power spectrum ratios. Is
calculated based on the control of the control unit 11, based on the average value of the power
spectrum ratio, based on the sound signal on the frequency axis related to the first sound
receiving unit 14a as a second The correction value of the phase of the sound signal on the
frequency axis related to the sound receiving unit 14b is calculated (S207).
The process of steps S205 to S207 is shown as the following equation (9) or equation (10).
[0062]
[0063]
[0064]
The correction value of the phase shown in the equation (9) and the equation (10) is a
representative value calculated based on the average value of the power spectral ratio for each
selected frequency, so there is no change with respect to the frequency.
In the second embodiment, the processing load can be reduced because the correction value is
calculated based on the spectrum of the selected N frequencies.
In the subsequent processing, in order to process the temporal change of the correction value,
the phase correction value Pcomp is treated as a correction value Pcomp (t) which is a function
of time (frame) t.
[0065]
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The sound processing device 1 smoothes the temporal change of the correction value by the
processing of the smoothing unit 127 based on the control of the control unit 11 (S208).
In step S208, smoothing processing is performed using the following equation (11).
[0066]
Pcomp (t) = γ · Pcomp (t−1) + (1−γ) · Pcomp (t) (11) However, a constant of γ: 0 or more and
1 or less
[0067]
As shown in equation (11), in step S208, by smoothing the temporal change using the correction
value Pcomp (t-1) one frame before, abrupt change of the correction value is prevented and there
is no sense of incongruity. It becomes possible to provide a sound.
As the constant γ, a numerical value such as 0.9 is used. Also, when the number N of the
selected frequency is less than a predetermined value such as 5 set in advance, calculation is
performed when SNR is low by temporarily setting constant γ to 1 and stopping update of the
correction value. It is possible to avoid the use of a correction value which is less accurate and to
improve the reliability. Furthermore, in order to prevent sudden overcorrection due to noise or
the like, it is desirable to set upper and lower limits to the correction value. It is also possible to
smooth the temporal change of the correction value using a sigmoid function instead of using the
equation (11).
[0068]
The sound processing apparatus 1 adds the correction value of the phase calculated in step S208
to the phase of the sound signal on the frequency axis related to the second sound receiving unit
14b by the processing of the correction unit 124 based on the control of the control unit 11.
Then, the sound signal related to the second sound receiving unit 14b is corrected (S209). In step
S209, correction with a constant correction value is performed over the entire frequency band.
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[0069]
Then, the sound processing device 1 performs processing of the sound processing unit 125
based on the control of the control unit 11 based on the sound signal related to the first sound
receiving unit 14 a and the sound signal corrected in phase related to the second sound receiving
unit 14 b. , And various sound processing such as suppression of the sound received by the first
sound receiving unit 14a (S210).
[0070]
The first and second embodiments merely exemplify a part of the infinite embodiment of the
present invention, and various hardware and software configurations can be set as appropriate,
and the examples It is possible to combine various processes other than the basic process
described above.
[0071]
It is a perspective view which shows an example of the external shape of the sound processing
apparatus which concerns on Embodiment 1 of this invention.
It is a block diagram which shows the structural example of the hardware of the sound
processing apparatus which concerns on Embodiment 1 of this invention.
It is a functional block diagram which shows the function example of the sound processing
apparatus which concerns on Embodiment 1 of this invention. It is a graph which shows the
change of the waveform of sound by the difference in the sensitivity difference of a microphone.
It is a circuit diagram showing an equivalent circuit of a microphone. It is a graph which shows
the change of the output voltage value based on an equation of motion. It is a flowchart which
shows the process example of the sound processing apparatus which concerns on Embodiment 1
of this invention. It is a radar chart which shows an example of the correction result of the
sensitivity difference by the sound processing apparatus concerning Embodiment 1 of this
invention. It is a functional block diagram which shows the function example of the sound
processing apparatus which concerns on Embodiment 2 of this invention. It is a flowchart which
shows the process example of the sound processing apparatus which concerns on Embodiment 2
of this invention. It is a perspective view which shows the external shape of a sound processing
apparatus. It is a radar chart which shows the measurement result of the directional
characteristic of a sound processing apparatus.
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Explanation of sign
[0072]
DESCRIPTION OF SYMBOLS 1 sound processing apparatus 11 control part 12 recording part
120 frame generation means 121 FFT conversion means 122 calculation means 123 calculation
means 124 correction means 125 sound processing means 126 frequency selection means 127
smoothing means 14 sound receiving part 14a first sound receiving part 14b Second receiver 16
Sound converter 160 Anti-aliasing filter 161 A / D converter 100 Computer program
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