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JP2006267534

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DESCRIPTION JP2006267534
An acoustic system capable of performing acoustic characteristic control adapted to different
environments is provided. An audio system includes an audio signal source unit 1 for outputting
an audio signal, and a signal processing unit 2 for processing the audio signal output from the
audio signal source unit 1 and outputting a plurality of processed audio signals. , Amplification
means 4 to 7 for amplifying a plurality of processed audio signals, a plurality of speakers 8 to 11
for outputting sounds based on each of the amplified plurality of processed audio signals, and a
plurality of speakers 8 to 11 And sound collection means 12 for collecting the sound to be
output, and the signal processing means 2 processes the processing content of the sound signal
so as to optimize the acoustic characteristics at the listening position based on the collection
sound by the sound collection means 12 decide. [Selected figure] Figure 1
Sound system
[0001]
The present invention relates to, for example, an acoustic system configured by connecting a
plurality of acoustic devices to an in-vehicle network, and more particularly to an acoustic system
that can be adjusted to optimize acoustic characteristics at a listening position.
[0002]
As an on-vehicle audio system, the audio signal reproduced by an audio device (for example, a CD
player) is subjected to signal processing in consideration of the sound field in the vehicle
compartment, and each speaker outputs audio based on the audio signal subjected to signal
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1
processing There is something to do.
However, in general, signal processing applied to audio signals does not correspond in detail to
the cabin environment for each user, but is processing selected based on data selected from
among prepared data patterns, Alternatively, the listener manually adjusts the acoustic
characteristics.
[0003]
In addition, according to the user's request, the acoustic characteristics of the vehicle interior can
be transmitted to an external data distribution center, and the data distributed from the data
distribution center can be used flexibly to respond to the vehicle room environment for each
user. There is also a proposal of a system (see, for example, Patent Document 1).
[0004]
Furthermore, in a system built based on an in-vehicle network (optical network) conforming to
the Media Oriented Systems Transport (MOST) standard which is a European standard of invehicle networks, a document proposed on the operation method of connected A / V devices (
Although there is also a patent document 2), this document does not disclose control of acoustic
characteristics based on an in-vehicle network.
[0005]
JP 2003-91290 A JP JP 2001-223718 A
[0006]
However, the conventional audio system processes the contents based on the data pattern
prepared in advance when performing signal processing on the audio signal, or a method in
which the listener manually adjusts the audio characteristics. It was not used to perform signal
processing that was adapted in detail to the cabin environment for each user.
Also, the method using data distributed from the data distribution center takes time because
communication with the outside is required, and there is also a problem that the device for
communication with the outside becomes large and expensive. .
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2
[0007]
Therefore, the present invention has been made to solve the problems of the prior art as
described above, and it is an object of the present invention to provide an acoustic system
capable of performing acoustic characteristic control adapted to different environments. is there.
[0008]
The audio system according to the present invention comprises audio signal source means for
outputting an audio signal, signal processing means for processing the audio signal and
outputting a plurality of processed audio signals, and each of the plurality of processed audio
signals. A plurality of speaker means for outputting a sound based on the sound; and an audio
collection means for collecting the sound output from the plurality of speaker means, wherein
the signal processing means is based on the collected sound by the audio collection means. It is
characterized in that the processing content of the audio signal is determined.
[0009]
According to the sound system of the present invention, the processing content of the sound
signal by the signal processing means is determined based on the sound collected by the sound
collection means, so that the sound characteristic control adapted to the difference in
environment can be performed. It has the effect of
[0010]
Embodiment 1
FIG. 1 is a block diagram showing a configuration of an acoustic system according to
Embodiment 1 of the present invention.
As shown in FIG. 1, the acoustic system according to the first embodiment includes an audio
signal source unit 1, a signal processing unit 2, a transmission medium 3, signal amplification
units 4 to 7, and signal amplification units 4 to 7. , The voice collecting means 12 and the
microphone 13 connected to the voice collecting means 12.
[0011]
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The audio signal source unit 1 is, for example, a reproduction device such as a CD player, and
outputs an audio signal in a digital signal format.
The signal processing means 2 applies signal processing to the audio signal output from the
audio signal source means 1 and outputs a plurality of processed audio signals to a plurality of
streams respectively.
The signal amplification means 4 to 7 receive and amplify the signal assigned to each of the
plurality of processed audio signals output from the signal processing means 2 and amplify the
respective amplified signals to the speakers 8 to 11. Output.
[0012]
The voice collecting means 12 outputs a voice signal output from the microphone 13 based on
the sound acquired by the microphone 13 as a voice signal of digital signal format.
[0013]
The transmission medium 3 is, for example, an optical network conforming to the MOST
standard.
To the transmission medium 3, an audio signal source unit 1, a signal processing unit 2, signal
amplification units 4 to 7, and an audio collecting unit 12 are connected.
Various signals are transmitted from the audio signal source unit 1 to the signal processing unit
2, from the audio collection unit 12 to the signal processing unit 2, and from the signal
processing unit 2 to the signal amplification units 4 to 7 via the transmission medium 3.
[0014]
FIG. 2 is a view showing the positions of the speakers 8 to 11 and the microphone 13 in the case
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4
where the sound system according to the first embodiment is applied to a car 30 (that is, a car
sound system). In the example shown in FIG. 2, the speakers 8 and 9 are disposed in the front
(front) 31 of the vehicle 30, the speakers 10 and 11 are disposed in the rear (rear) 32, and the
left rear seat (left of the rear seat 33) The microphone 13 is arranged in the column). The
position of the microphone 13 indicates the measurement point of the acoustic characteristic, in
an attempt to obtain an optimal listening environment in the rear left seat. As a method of
correcting the acoustic characteristics, a time alignment method for correcting an adverse effect
due to a difference in arrival time of sound due to a difference in distance between the speakers
8 to 11 and a listening position, and a standing wave generated in a vehicle interior An equalizer
method for correcting non-flatness of frequency characteristics is generally used. Therefore, as
the operation of the signal processing means 2 in the first embodiment, the case of using the
time alignment method and the equalizer method will be described. However, the present
invention is not limited to these methods. Further, the positions and the number of the speakers
8 to 11 and the positions and the number of the microphones 13 are not limited to the illustrated
example.
[0015]
In the case of FIG. 2, since the distance between each of the speakers 8 to 11 in the vehicle room
and the microphone 13 corresponding to the listening position is different, the time taken for the
sound output from each of the speakers 8 to 11 to reach the microphone 13 Arrival time) is
different. In the time alignment method, an audio signal of a channel having an early arrival time
of sound at a listening position is delayed, and the arrival times of sounds from the speakers 8 to
11 are corrected to be equal at the listening position. For example, speaking in the distance
relationship of FIG. 2, since the sound from the speaker 11, the sound from the speaker 10, the
sound from the speaker 9, and the sound from the speaker 8 reach the microphone 13 in this
order, arrival of the sound from the speaker 8 The audio signal input to each of the speaker 11,
the speaker 10, and the speaker 9 is delayed so as to match the time.
[0016]
FIG. 3 is a block diagram showing the configuration of the signal processing means 2 of the
acoustic system according to the first embodiment. As shown in FIG. 3, the signal processing
means 2 is referred to as an interface circuit (hereinafter referred to as “I / F circuit”)
responsible for signal exchange with the transmission medium 3. 2a, a signal generation circuit
2b for generating a reference audio signal Sr for measurement, and a selection circuit 2c for
switching between the reference audio signal Sr and the original audio signal S1 from the audio
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signal source 1 And a processing circuit 2d for processing the original audio signal S1 based on
the microphone audio signal Sm. The I / F circuit 2a takes out the original audio signal S1
transmitted from the audio signal source means 1 from the transmission medium 3 and outputs
it to the select circuit 2c, and processes the microphone audio signal Sm transmitted from the
audio collecting means 12 Output to 2d. Further, the I / F circuit 2a outputs the four systems of
processed audio signals S4, S5, S6, S7 output from the processing circuit 2d to the transmission
medium 3 in correspondence with the signal amplification means 4-7.
[0017]
FIG. 4 is a block diagram showing the configuration of the audio signal source means 1, the
signal amplification means 4, and the audio collecting means 12. As shown in FIG. As shown in
FIG. 4, the audio signal source means 1 comprises an audio signal source 1a and an I / F circuit
1b. Further, the sound collecting means 12 is composed of an A / D circuit 12a for converting an
input signal from the microphone 13 into a digital signal format, and an I / F circuit 12b. Further,
the signal amplification means 4 is composed of an I / F circuit 4a and an amplification circuit
4b. Each of the signal amplification means 5 to 7 has the same configuration as the signal
amplification means 4. The I / F circuits 1 b, 4 b, and 12 b in each means transfer signals to the
transmission medium 3 in the same manner as the I / F circuit 2 a of the signal processing means
2.
[0018]
5 (a) to 5 (j) are waveform diagrams for explaining the operation in the case where the signal
processing means 2 of the acoustic system according to the first embodiment performs
processing based on a time alignment method. 5 (a) and 5 (f) show a measurement impulse
signal (reference sound signal Sr), and FIGS. 5 (b) to 5 (e) show microphone collected sound
signals by the speakers 8 to 11 before time alignment processing. 5 (g) to 5 (j) show microphone
collected sound signals by the speakers 8 to 11 after time alignment processing.
[0019]
The procedure of the time alignment process determines the time (arrival time) and arrival time
difference until the sound output from each of the speakers 8 to 11 reaches the microphone 13
first, and the path with the slowest arrival time (each speaker 8 to 11 To match the route from
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the microphone 13 to the microphone 13), the audio signal corresponding to each of the
speakers 8 to 11 is delayed. At the time of the operation | movement which calculates | requires
an arrival time difference, the signal generation circuit 2b generate | occur | produces the
impulse signal (FIG. 5 (a)) for time alignment methods based on the instruction | indication of the
processing circuit 2d. At this time, the select circuit 2c selects the measurement impulse signal
(reference voice signal) Sr and outputs it to the processing circuit 2d. The processing circuit 2d
sequentially outputs the measurement impulse signal Sr corresponding to each of the speakers 8
to 11 to the path of the processed audio signal, and measures the arrival time for each of the
speakers 8 to 11 from the microphone sound collection signal. . FIG.5 (b)-(e) has shown the
difference of the arrival time measured before the time alignment process. Although the
waveforms shown in FIGS. 5 (a) to 5 (j) are drawn as similar impulse waveforms for the sake of
simplicity, in fact, due to the influence of a reflected wave in the vehicle interior, etc., FIG. b) The
waveforms of (e) and (g) to (j) are different. However, in general, since the direct wave arrives the
fastest, if the arrival time is determined at the leading edge of the measurement signal waveform,
it is not affected by the reflected wave or the like. Therefore, the times Tb, Tc, Td, and Te in FIGS.
5A to 5J indicate the arrival times of the outputs of the speakers 8 to 11, respectively. The
difference in arrival time of the outputs of the speakers 8 to 11, that is, the arrival time
difference corresponds to the shortest path (distance) from each of the speakers 8 to 11 to the
microphone 13 as shown in FIG.
[0020]
Thus, based on the arrival time difference determined for each of the speakers 8 to 11, the
processing circuit 2d gives a time delay to the audio signal corresponding to each of the speakers
8 to 11, and the time alignment of FIGS. 5 (g) to 5 (j). The arrival times of the outputs of the
speakers 8 to 11 are corrected as in the microphone collected sound signal by the outputs of the
speakers 8 to 11 after processing. When the processing content is determined, the original audio
signal from the audio signal source means 1 is input to the processing circuit 2d by the selection
of the selection circuit 2c, and the processing shifts to the normal operation subjected to the time
alignment processing.
[0021]
FIGS. 6 (a) to 6 (c) are graphs for explaining the operation in the case where the signal
processing means 2 of the acoustic system according to Embodiment 1 performs processing
based on the equalizer method. The configuration of the signal processing means 2 that performs
processing based on the equalizer method is FIG. 3 like the signal processing means 2 that
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performs processing based on the time alignment method, and the operation content by the
signal generation circuit 2b and the processing circuit 2d is the time alignment method Only
replace the equalizer method with.
[0022]
In the procedure of processing based on the equalizer method, first, frequency characteristics are
individually determined for each of the sounds from the speakers 8 to 11 by octave band pass
filters, and the frequency or gain of the peak or dip of each frequency characteristic is detected.
Correct to flatten the characteristics. At the time of the operation | movement which calculates |
requires the frequency characteristic by each speaker 8-11, the signal generation circuit 2a
generate | occur | produces the pink noise signal for equalizer methods based on the instruction
| indication of the processing circuit 2d. The pink noise has a slope of -3 dB / oct (that is, the
level decreases by 3 dB each time the frequency is doubled) as shown in FIG. Has a characteristic
of falling. When a pink noise signal is measured by an octave band pass filter, as shown in FIG. 6
(b), the energy is uniform in any octave, so the pink noise signal is generally used as a
measurement signal. There is. FIG. 6C shows the frequency characteristics of the microphone
audio signal collected by the microphone 13 when one of the speakers 8 to 11 emits pink noise,
which is measured by the octave band pass filter in the processing circuit 2d. . The non-flatness
of the frequency characteristic resulting from the standing wave which arises in a vehicle interior
arises, for example, it becomes a frequency characteristic as shown with a broken line. The
frequency and gain of this peak or dip are detected, and for example, an infinite impulse
response (IIR) digital filter coefficient for flattening the peak or dip is obtained based on a
correction table prepared in advance, and this filtering process is performed on the audio signal.
And correct the frequency characteristics. The solid line in FIG. 6C is the corrected characteristic,
and the frequency characteristic is corrected so as to be flattened.
[0023]
FIG. 7 is a flowchart showing the processing procedure of the audio system according to the first
embodiment. At the start of the operation, the user first operates to enter the measurement mode
of the correction coefficient (step ST1). When entering the measurement mode, the user operates
to measure the correction factor of the time alignment process or to measure the correction
factor of the equalizer process (step ST2).
[0024]
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8
When the correction coefficient of the time alignment process is measured, the arrival time of the
speakers 8 to 11 output sound to the microphone 13 is sequentially measured (steps ST3 to
ST6), and the respective delay times are determined (step ST7). In order to reduce the influence
of erroneous detection, the measurement and determination in step ST3-ST7 are repeated plural
times (N times) and averaged (step ST8), and a time alignment correction coefficient is
determined (step ST9).
[0025]
When measuring the correction coefficient of the equalizer processing, measure the frequency
and gain of the peak or dip of each frequency characteristic for the speakers 8 to 11 output
sound (steps ST10 to ST13) and analyze the frequency characteristics respectively (step ST14) .
In order to reduce the influence of erroneous detection, the measurement and determination in
steps ST10 to ST14 are repeated a plurality of times (N times) and averaged (step ST15), and an
equalizer correction coefficient is determined (step ST16).
[0026]
Thus, after the measurement mode ends, the process proceeds to time alignment processing and
/ or equalizer processing (step ST1). The signal processing means 2 performs time alignment
processing and / or equalizer processing based on the measurement result (time alignment
correction coefficient and equalizer correction coefficient) on the original audio signal S1 from
the audio signal source means 1 as a normal operation (step ST17 and ST18) The processed
audio signals S4, S5, S6 and S7 are outputted to the respective signal amplifying means 4 to 7.
[0027]
In the above example, the signal generation circuit 2b is provided in the signal processing means
2 and the signal generation circuit 2b generates the reference audio signal Sr for measurement.
However, the signal generation circuit 2b performs signal processing. It may be provided outside
the means 2 (for example, in the audio signal source means 1 and the signal amplification means
4 to 7), and the signal processing means 2 may control the signal generation operation.
[0028]
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9
As described above, according to the acoustic system of the first embodiment, acoustic
characteristic control adapted to the difference in environment can be performed.
In addition, when the equalizer processing is performed, it is possible to achieve equalization of
the acoustic characteristics realized in each vehicle compartment. Furthermore, by setting the
reference voice signal generation operation and the voice collection operation for voice collection
to be performed in cooperation with each other, it is possible to automate the adjustment of
acoustic characteristics.
[0029]
Further, the signal processing means 2 may be included in the audio signal source means 1
without making the processing function of the signal processing means 2 independent.
[0030]
FIG. 8 is a block diagram showing a configuration example using a class D amplifier as the
amplification circuit 4 b in the signal amplification means 4 of the acoustic system according to
the first embodiment.
The signal amplification means 5 to 7 have the same configuration as the signal amplification
means 4. As shown in FIG. 8, the amplification circuit 4b performs noise shaping on the audio
signal from the I / F circuit 4a to convert it into a PWM signal (pulse width modulation signal),
and based on the PWM signal It has a power driver 4b2 that performs switching operation, and
an LC filter 4b3 that removes the PWM carrier and drives the speaker 8 with an analog audio
signal. Class D amplifiers are becoming more popular because they are compact and highly
efficient and can produce high-quality sound output without crossover distortion, but strong high
frequency noise is superimposed on the speaker drive line because the output stage is switching
operation. There's a problem. Therefore, by arranging the signal amplifying means 4 to 7 on the
speaker side via the in-vehicle network as in the present configuration, the speaker drive line can
be minimized, and while utilizing the features of the class D amplifier, unnecessary radiation
Occurrence can be reduced.
[0031]
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10
Second Embodiment FIG. 9 is a block diagram showing a configuration of an acoustic system
according to Embodiment 2 of the present invention. As shown in FIG. 9, the audio system
according to the second embodiment includes the audio signal source means 1, the signal
processing means 2, the transmission medium 3, the signal amplification / audio collection
processing means 14 to 17, and the signal amplification. The speakers 8 to 11 are connected to
the voice collection processing means 14 to 17, and the microphones 18 to 21 are connected to
the signal amplification / voice collection processing means 14 to 17. The signal amplification /
voice collecting means 14 is composed of an I / F circuit 14a, an amplification circuit 14b, and an
A / D circuit 14c. The other signal amplification / voice collecting means 15 to 17 have the same
configuration as the signal amplification / voice collecting means 14.
[0032]
FIG. 10 is a view showing the positions of the speakers 8 to 11 and the microphones 18 to 21
when the sound system according to the second embodiment is applied to a car 40 (that is, a car
sound system). In the example shown in FIG. 10, the speakers 8 and 9 and the microphones 18
and 19 are disposed in the front (front) 41 of the vehicle 40, and the speakers 10 and 11 and the
microphones 20 and 21 are disposed in the rear (rear) 42. , The rear left seat (the left row of the
rear seats 42) is the listening position 22. FIG. 10 shows an example using two systems of
speakers and microphones for the front and rear. Although the microphones 18 to 21 are
described on the inside of the compartment with respect to the speakers 8 to 11, respectively, on
the display of the figure, the present invention is not limited to this positional relationship, and
usually, the front and rear of the compartment respectively Are located at the four corners of the
The positions of the microphones 18 to 21 indicate the measurement points of the acoustic
characteristics in the second embodiment, and are different from the first embodiment in that the
listening position 22 of the rear left seat and the measurement points are different from each
other. . In other words, the second embodiment is to obtain an optimal listening environment at
the listening position 22 which is a position different from the measurement point (microphone
position). The correction method of the acoustic characteristic in the second embodiment is
basically the same as that in the first embodiment.
[0033]
11 to 14 are explanatory views of a case where correction is performed by the time alignment
method at the listening position 22 in the acoustic system according to the second embodiment.
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11
In FIGS. 11-14, it is assumed that the speakers 8-11 exist in the same position as the
microphones 18-21 for simplification of description, and only the microphones 18-21 are
described. The listening position 22 is represented by coordinates (X, Y) with the microphone 18
depicted at the upper left in FIGS. 11 to 14 as the origin.
[0034]
FIG. 11 shows the case where the speakers 8 to 11 sequentially output the sound based on the
measurement impulse signal, and the microphone 21 measures the arrival time. In FIG. 11, A 21
is the shortest path (also referred to as distance or path length) from the speaker 8 to the
microphone 21, B 21 is the shortest path from the speaker 9 to the microphone 21, and C 21 is
the shortest path from the speaker 10 to the microphone 21. . The shortest path D11 from the
speaker 11 to the microphone 21 is not shown because the speaker 11 and the microphone 21
overlap. Further, in FIG. 11, Aa is the shortest path (also referred to as distance or path length)
from the speaker 8 to the listening position 22, Ba is the shortest path from the speaker 9 to the
listening position 22, Ca is from the speaker 10 to the listening position 22 Is the shortest path
of The arrival times until the microphones 21 reach the microphones 21 through the paths A21,
B21, C21, and D21 are TA21, TB21, TC21, and TD21, respectively. Here, in the TD 21, the time
taken for the impulse signal Sr for measurement substantially output from the signal processing
means 2 to be outputted as a sound from each of the speakers 8 to 11 and the sound taken in by
the microphone 21 are signal processing means Since it can be regarded as the sum of the time
to reach 2, the substantial sound arrival time becomes as in the following equations (1) to (3).
Arrival time of path A21 = TA21-TD21 (1) Arrival time of path B21 = TB21-TD21 (2) Arrival time
of path C21 = TC21-TD21 (3) Further, from equations (1) to (3) and the speed of sound Vs Each
path length A21, B21, C21 can be calculated.
[0035]
Furthermore, the path lengths Aa, Ba, Ca from the speakers 8-11 to the listening position become
as in the following equations (4)-(6). Aa=√(X<2>+Y<2>) (4)
Ba=√(X<2>+(C21−Y)<2>) (5) Ca=√((B21−X)<2>+Y<2>)
(6)
[0036]
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12
Therefore, the arrival time of each path Aa, Ba, Ca at the listening position is as shown in the
following formulas (7) to (9). Arrival time of path Aa = (Aa / A21) * (TA21-TD21) (7) Arrival time
of path Ba = (Ba / B21) * (TB21-TD21) (8) Arrival time of path Ca = (Ca / C21 ) * (TC21-TD21) (9)
[0037]
FIG. 12 shows the case where sounds based on the measurement impulse signal are sequentially
output from the speakers 8 to 11 and the arrival time is measured by the microphone 20. In FIG.
12, A20 is the shortest path from the speaker 8 to the microphone 20 (also referred to as
distance or path length), B20 is the shortest path from the speaker 9 to the microphone 20, and
D20 is the shortest path from the speaker 11 to the microphone 20. . The shortest path C20
from the speaker 10 to the microphone 20 is not shown because the speaker 10 and the
microphone 20 overlap. Further, in FIG. 12, Aa is the shortest path (also referred to as distance
or path length) from the speaker 8 to the listening position 22, Ba is the shortest path from the
speaker 9 to the listening position 22, and Da is from the speaker 11 to the listening position 22
Is the shortest path of The arrival times until the microphones 20 reach through the paths A20,
B20, and D20 and reach the microphone 20 are TA20, TB20, TC20, and TD20, respectively.
Here, in the TD 20, the time until the measurement impulse signal substantially output from the
signal processing means 2 is output as a sound from each of the speakers 8 to 11 and the sound
taken in by the microphone 21 are signal processing means 2 Since it can be regarded as the
sum of the time until it reaches to, the substantial arrival time of sound becomes like following
Formula (10)-(12). Arrival time of path A20 = TA20-TC20 (10) Arrival time of path B20 = TB20TC20 (11) Arrival time of path D20 = TD20-TC20 (12) Further, from equations (1) to (3) and the
speed of sound Vs Each path length A20, B20, D20 can be calculated.
[0038]
Further, the path lengths Aa, Ba, Da from the speakers 8 to 11 to the listening position become as
in the following equations (13) to (15). Aa=√(X<2>+Y<2>) (13)
Ba=√(X<2>+(D20−Y)<2>) (14)
Da=√((A20−X)<2>+(D20−Y)<2>) (15)
[0039]
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13
Therefore, the arrival time of each path Aa, Ba, Da at the listening position is as shown in the
following equations (16) to (18). Arrival time of route Aa = (Aa / A20) * (TA20-TC20) (16) Arrival
time of route Ba = (Ba / B20) * (TB20-TC20) (17) Arrival time of route Da = (Da / D20 ) * (TD20TC20) (18)
[0040]
FIG. 13 shows the case where sounds based on the measurement impulse signal are sequentially
output from the speakers 8 to 11 and the arrival time is measured by the microphone 19. In FIG.
13, A 19 is the shortest path from the speaker 8 to the microphone 19 (also referred to as
distance or path length), C 19 is the shortest path from the speaker 10 to the microphone 19,
and D 19 is the shortest path from the speaker 11 to the microphone 19. . The shortest path B
19 from the speaker 9 to the microphone 19 is not shown because the speaker 9 and the
microphone 19 overlap. Further, in FIG. 13, Aa is the shortest path from the speaker 8 to the
listening position (also referred to as distance or path length), Ba is the shortest path from the
speaker 9 to the listening position, Da is the shortest path from the speaker 11 to the listening
position It is. It is assumed that TA19, TB19, TC19, and TD19 are arrival times until signals are
output from the speakers 8 to 11 and reach the microphone 20 through the paths A19, B19,
C19, and D19, respectively. Here, in the TD 19, the time until the measurement impulse signal
substantially output from the signal processing means 2 is output as a sound from each of the
speakers 8 to 11 and the sound taken in by the microphone 21 are signal processing means 2.
Since it can be regarded as the sum of the time until it reaches to, the substantial arrival time of
sound becomes like following Formula (19)-(21). Arrival time of path A19 = TA19-TB19 (19)
Arrival time of path C19 = TC19-TB19 (20) arrival time of path D19 = TD19-TB19 (21) Further,
from equations (19) to (21) and the speed of sound Vs Each path length A19, C19, D19 can be
calculated.
[0041]
Therefore, the arrival time of each path Aa, Ca, Da at the listening position is as expressed by the
following equations (22) to (24). Aa=√(X<2>+Y<2>) (22)
Ca=√((D19−X)<2>+Y<2>) (23)
Da=√((D19−X)<2>+(A19−Y)<2>) (24)
[0042]
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14
Therefore, the arrival time of each path Aa, Ca, Da at the listening position is as shown in the
following formulas (25) to (27). Arrival time of route Aa = (Aa / A19) * (TA19-TB19) (25) Arrival
time of route Ca = (Ca / C19) * (TC19-TB19) (26) Arrival time of route Da = (Da / D19 ) * (TD19TB19) (27)
[0043]
FIG. 14 shows the case where sounds based on the measurement impulse signal are sequentially
output from the speakers 8 to 11 and the arrival time is measured by the microphone 18. In FIG.
14, B18 is the shortest path from the speaker 9 to the microphone 18 (also referred to as
distance or path length), C18 is the shortest path from the speaker 10 to the microphone 18,
D18 is the shortest path from the speaker 11 to the microphone 18 . The shortest path B 18
from the speaker 8 to the microphone 18 is not shown because the speaker 8 and the
microphone 18 overlap. Further, in FIG. 14, Ba is the shortest path (also referred to as distance or
path length) from the speaker 9 to the listening position 22, Ca is the shortest path from the
speaker 10 to the listening position 22, and Da is from the speaker 11 to the listening position
22 Is the shortest path of It is assumed that TA 18, TB 18, TC 18, and TD 18 are output times
from the speakers 8 to 11 to reach the microphone 18 through the paths A 18, B 18, C 18, and D
18, respectively. Here, in the TA 18, the time until the measurement impulse signal substantially
output from the signal processing means 2 is output as a sound from each of the speakers 8 to
11 and the sound taken in by the microphone 21 are signal processing means 2. Since it can be
regarded as the sum of the time until it reaches to, the arrival time of the substantial sound
becomes like following Formula (28)-(30). Arrival time of path B18 = TB18-TA18 (28) Arrival
time of path C18 = TC18-TA18 (29) arrival time of path D18 = TD18-TA18 (30) Further, from
equations (28) to (30) and the speed of sound Vs Each path length B18, C18, D18 can be
calculated.
[0044]
Therefore, the arrival time of each path Aa, Ca, Da at the listening position is as expressed by the
following equations (31) to (33). Ba=√(X<2>+(B18−Y)<2>) (31)
Ca=√((C18−X)<2>+Y<2>) (32)
Da=√((C18−X)<2>+(B18−Y)<2>) (33)
[0045]
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Therefore, the arrival time of each path Ba, Ca, Da at the listening position is expressed by the
following equations (34) to (36). Arrival time of the route Ba = (Ba / B18) * (TB18-TA18) (34)
Arrival time of the route Ca = (Ca / C18) * (TC18-TA18) (35) Arrival time of the route Da = (Da /
D18 ) * (TD18-TA18) (36)
[0046]
As described above, according to the measurements shown in FIGS. 11 to 14, for example, the
arrival time of the route Aa from the speaker 8 to the listening position 22 can be expressed by
the equations (7), (16), and (25). It is possible to determine the arrival time from the speaker 8 at
the listening position 22 by calculating the obtained value and averaging the obtained values.
The same can be applied to other matters. The time alignment correction based on these is the
same as that of the first embodiment.
[0047]
Similarly, with regard to the equalizer processing, voices are collected individually in the
microphones 18 to 21 based on the pink noise generated from the speakers 8 to 11 in FIG. 10,
and frequency characteristics measured by the octave band pass filter are obtained. As a result,
the frequency characteristic at the measurement point of each microphone can be determined,
and the filter coefficient for correcting the non-flatness of the frequency characteristic at each
point can be determined. For the listening position 22, for example, there is a method in which
the correction contents of the near measurement points are highly weighted (for example,
weighted in inverse proportion to the path length), and the correction contents of the respective
measurement points are combined. It is applicable.
[0048]
It is a block diagram which shows the structure of the acoustic system which concerns on
Embodiment 1 of this invention. It is a figure which shows arrangement | positioning of the
structure at the time of applying the acoustic system which concerns on Embodiment 1 to a
motor vehicle. FIG. 2 is a block diagram showing a configuration of signal processing means of
the acoustic system according to Embodiment 1. FIG. 2 is a block diagram showing
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16
configurations of audio signal source means, signal amplification means, and audio collection
means of the acoustic system according to Embodiment 1. (A)-(j) is a signal waveform diagram
for demonstrating the operation | movement in the case of performing a time alignment process
in the signal processing means of the acoustic system which concerns on Embodiment 1. FIG. (A)(c) is a graph for demonstrating the operation | movement in the case of performing an equalizer
process in the signal processing means of the acoustic system which concerns on Embodiment 1.
FIG. 5 is a flowchart showing an operation of the sound system according to the first
embodiment. FIG. 6 is a block diagram showing a configuration using a class D amplifier as an
amplifier circuit in the signal amplification means of the acoustic system according to the first
embodiment. It is a block diagram which shows the structure of the sound system which
concerns on Embodiment 2 of this invention. It is a figure which shows arrangement |
positioning of the structure at the time of applying the acoustic system which concerns on
Embodiment 2 to a motor vehicle. FIG. 16 is an operation explanatory view (No. 1) when
performing time alignment processing in the signal processing means of the acoustic system
according to the second embodiment; FIG. 17 is an operation explanatory view (No. 2) in the case
where time alignment processing is performed in the signal processing means of the acoustic
system according to the second embodiment. FIG. 17 is an operation explanatory view (No. 3) in
the case of performing time alignment processing in the signal processing means of the acoustic
system according to the second embodiment. FIG. 16 is an operation explanatory view (No. 4) in
the case of performing time alignment processing in the signal processing means of the acoustic
system according to the second embodiment.
Explanation of sign
[0049]
DESCRIPTION OF SYMBOLS 1 audio | voice signal source means, 2 signal processing means, 3
transmission media, 4-7 signal amplification means, 8-11 speakers, 12 audio collection means,
13 microphones, 14-17 signal amplification / audio collection means, 18-21 microphones, 22
listening position.
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