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JPH05181489

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DESCRIPTION JPH05181489
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a
sound field correction apparatus.
[0002]
2. Description of the Related Art In an audio apparatus or the like, the sound emitted from a
speaker is influenced by some kind of sound field until it reaches a listening point, ie, the
influence of reflection or diffraction, unless the room is a free sound field like an anechoic room
etc. It is known that, as a result, the disturbance of the frequency characteristic due to the
difference of the wavelength of the frequency is generated. In order to correct such disturbance
of the frequency characteristic due to the influence of the sound field, a fixed equalizer, a graphic
equalizer, a parametric equalizer or the like is used.
[0003]
However, in the case of the above-described fixed equalizer, there is a drawback that it can not
cope with the change of the sound field at all. Although readjustment is possible in the case of
graphic equalizers and parametric equalizers, there is a drawback that it is necessary to rely on
the sense of hearing or to set the measuring instrument, etc., and manual adjustment is necessary
and automatic adjustment can not be performed. . The present invention aims to solve the
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drawbacks of these conventional sound field correction devices.
[0004]
SUMMARY OF THE INVENTION In order to achieve the above object, the sound field correction
apparatus of the present invention listens to an acoustic signal at a predetermined listening
point, converts it into an electrical signal, and outputs it. Sound field correction is performed on
the basis of the difference between the frequency characteristic of the source side signal output
as an acoustic signal and the frequency characteristic of the listened signal for each
predetermined frequency band, and the difference of the frequency characteristic And sound
field correction means.
[0005]
The difference between the frequency characteristic of the signal obtained by the listening means
and the frequency characteristic of the signal on the source side is determined, the frequency
characteristic is corrected based on the difference, and the sound field correction is performed.
[0006]
Embodiments of the present invention will be described below with reference to the drawings.
In FIG. 1, the output of the preamplifier 1 is sent to the power amplifier 3 through the electronic
n-element graphic equalizer 2 and is output as sound from the speaker 4.
In the initial state, the graphic equalizer 2 is in a flat state, and an uncorrected audio signal is
output from the speaker 4 via the power amplifier 3. A microphone 5 is installed at a
predetermined listening point of a sound field formed by the speaker 4, and a signal from the
microphone 5 is configured to be supplied to a plurality of band pass filters 7 via the microphone
preamplifier 6. The band pass filter 7 divides the audible band into n predetermined bands and at
least several bands, but the more accurate the correction, the more the band can be corrected.
The signal divided into n bands by the band pass filter 7 is sampled by the sample and hold
circuit 8, converted into a digital signal by the A / D converter 9, and sent to the microcomputer
10. The timing of sampling by the sample and hold circuit 8 is controlled by the microcomputer
10.
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[0007]
On the other hand, the source signal from the preamplifier 1 is similarly sent to the band pass
filter 21 through the delay circuit 20, divided here into the above-mentioned n frequency bands,
sampled by the sample and hold circuit 22, and A / D converted. The digital signal is converted
into a digital signal by the unit 23 and input to the microcomputer 10. The delay circuit 20 is for
delaying by the time to reach from the preamplifier 1 to the microphone 5, but a method of
adjusting the control signal of the sample and hold circuit 22 is also possible.
[0008]
The microcomputer 10 is configured to process the source signal a input from the preamplifier 1
and the feedback signal b input from the microphone 5 as follows. As shown in FIG. 2, when the
operator issues a correction start operation command, a sample signal is sent to the sample and
hold circuit 8 and the sample and hold circuit 22 to execute a sample (step 30). When a
predetermined time has passed, the microcomputer 10 sends a hold signal, digitizes the sampled
analog signal by the A / D converter 9 and the A / D converter 23, and reads the data (step 31).
After data reading, level matching of the source signal a and the feedback signal b is performed
(step 32). This is to correct the difference between the level of the source signal a and the level of
the feedback signal b due to the gain of the power amplifier, the efficiency of the speaker, the
sensitivity of the microphone 5, the gain of the microphone preamplifier 6, and the like. Then,
the correction is performed (step 33), and the data is transferred (step 34).
[0009]
The level alignment described above will be described in detail. As shown in FIG. 3, k is set to 0,
and the difference between the source signal a and the feedback signal b is determined from 1 to
n for each frequency band (steps 40, 41, 42, 43). This is illustrated in FIG. Assuming that the
difference between the source signal a and the feedback signal b is ck, an average value of the ck
is determined to obtain a difference dk between ck and the average value in each frequency band
(steps 44, 45, 46, 47). FIG. 6 shows the dk and the source signal a.
[0010]
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This dk represents the difference between the source signal a and the feedback signal b after
level adjustment, and the sound field correction is performed based on this dk. That is, since dk
shown in FIG. 6 is due to the influence of the sound field, it is possible to correct the dk by
equalizing the characteristic reverse to the frequency characteristic. For that purpose, as shown
in FIG. 4, first, dk is subtracted from the source signal a which is a flat characteristic for each
frequency band, and ek is determined as shown in FIG. 7 (step 50).
[0011]
However, it is not necessary to simply convert the electrical signal into an acoustic signal, and
additional auditory correction (step 51) is added because the auditory sensitivity needs to be
considered. With regard to hearing correction, methods based on Fletcher / Munson curves or
Robinson curves are known, and any such conventionally known methods can be employed. In
this embodiment, a simplified method is used, and as shown in FIG. 8, the auditory characteristic
fk is subtracted from the characteristic ek before auditory correction for each frequency band to
obtain the characteristic gk after auditory correction.
[0012]
Also, the sensitivity of the microphone 5 is not necessarily a flat characteristic with respect to the
audible band of 20 Hz to 20 KHz, and the low end or high end sensitivity is often poor depending
on the aperture of the microphone or the like. Therefore, the microphone correction (step 52) is
further added. This correction is performed based on the sensitivity data of the microphone 5 to
be used. FIG. 9 shows an example of this. The microphone characteristics pk are obtained by
subtracting the microphone characteristics pk from the characteristics gk before the correction
for each frequency band.
[0013]
This data is used as the final correction characteristic zk as shown in FIG. 10 as the gain of each
element of the graphic equalizer 2 (step 53). Then, this data is transferred from the
microcomputer 10 to the graphic equalizer 2 and equalization is performed at the front stage of
the power amplifier 3 to perform sound field correction.
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[0014]
As described above, in the above embodiment, since the signal representing the disturbance of
the frequency characteristic due to the sound field is fed back by the microphone 5 and
equalization is performed in the graphic equalizer 2 so as to cancel the disturbance, it is
automatically performed. Sound field correction is performed accurately.
[0015]
As described above, the sound field correction apparatus of the present invention listens to an
acoustic signal at a predetermined listening point, converts it into an electrical signal, and
outputs it as an acoustic signal. Means for obtaining for each predetermined frequency band the
difference between the frequency characteristic of the source side signal and the frequency
characteristic of the signal heard; and sound field correction means for performing sound field
correction based on the difference between the frequency characteristics. Since it is provided, the
difference between the frequency characteristic of the signal obtained by the listening means and
the frequency characteristic of the signal on the source side is determined, the frequency
characteristic is corrected based on the difference, and the sound field correction is performed. is
there.
[0016]
Brief description of the drawings
[0017]
1 is a block diagram showing an embodiment of the present invention.
[0018]
2 is a flow chart for explaining the operation of an embodiment of the present invention.
[0019]
3 is a flowchart illustrating the operation of an embodiment of the present invention.
[0020]
4 is a flow chart for explaining the operation of an embodiment of the present invention.
[0021]
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5 is a waveform diagram for explaining the operation of an embodiment of the present invention.
[0022]
6 is a waveform diagram for explaining the operation of an embodiment of the present invention.
[0023]
7 is a waveform diagram for explaining the operation of an embodiment of the present invention.
[0024]
8 is a waveform diagram for explaining the operation of an embodiment of the present invention.
[0025]
9 is a waveform diagram for explaining the operation of an embodiment of the present invention.
[0026]
10 is a waveform diagram for explaining the operation of an embodiment of the present
invention.
[0027]
Explanation of sign
[0028]
1: Preamplifier, 2: Graphic equalizer, 3: Power amplifier, 4: Speaker, 5: Microphone, 6:
Microphone preamplifier, 7: Band pass filter, 8: Sample hold circuit, 9: A / D converter, 10:
Microcomputer, 20: delay circuit, 21: band pass filter, 22: sample and hold circuit, 23: A / D
converter.
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