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JPWO2013183185

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DESCRIPTION JPWO2013183185
The HPF 702 for transforming the frequency characteristic of the target signal, the phase
correction unit 701 for correcting the phase characteristic of the target signal and making it
substantially the same as the phase characteristic of the HPF 702, and the gain of the signal
output from the phase correction unit 701 Of the gain coefficient of the first multiplier 705 and
the gain coefficient of the second multiplier 706, and the second multiplier 706 for adjusting the
gain of the signal output from the HPF 702; A coefficient determination unit that determines the
gain coefficients of the first and second multipliers 705 and 706 so that the sum becomes a
constant value, and two output from the first multiplier 705 and the second multiplier 706 And
an adder 713 for adding the signals.
Frequency characteristic deformation device
[0001]
The present invention relates to a signal processing technique for improving distortion and
sound breakup in acoustic signal reproduction.
[0002]
In a speaker reproduction system that reproduces an audio signal such as music or
announcement sound by a speaker, the sound quality may be degraded due to distortion or
sound break.
07-05-2019
1
The causes of distortion and sound breaking are roughly divided into two. The first case is a case
where the input signal to the speaker is distorted, and the second case is a case where distortion
or sound break occurs because the reproduction limit of the speaker is exceeded even if the input
signal is not distorted. The first case can be described as follows. In recent acoustic signal
reproduction systems, devices for correcting frequency characteristics and adjusting volume by
digital processing are increasing. In the correction of the frequency characteristics, for example,
when the high frequency component is increased by 10 dB, there is a possibility that the digital
signal may be saturated at a volume value of -10 dBFS or more. Note that 0 dBFS represents the
maximum amplitude value of the digital signal. As a result, when the volume is high, the
reproduced sound is distorted digitally, and the sound quality is degraded. This situation is
shown in FIG.
[0003]
In FIG. 2, the vertical axis represents the amplitude intensity of the digital signal, and the
horizontal axis represents the frequency. Also, the area where the signal is saturated and the
sound break occurs is shown in gray, and the boundary is shown by a thick line. Reference
numerals 201, 202, and 203 show an example of the frequency characteristic of the digital
acoustic signal whose frequency characteristic has been corrected. 201 is a characteristic when
the volume value is small, 202 is a characteristic when the volume value is medium, 203 is a
characteristic It is a characteristic when the volume value is large. At the volume values 201 and
202, since the acoustic signal does not exceed 0 dBFS, no sound break occurs and it can be
enjoyed with the original sound quality. However, if the volume is increased as in 203, the signal
strength of a part of the high frequency component exceeds 0 dBFS, resulting in digital
saturation. When the signal is saturated, distortion or sound break occurs and the sound quality
is degraded.
[0004]
As described above, when reproducing a digital signal whose frequency characteristic has been
corrected at a large volume, there are cases where a specific frequency component exceeds 0
dBFS, which is the maximum amplitude value of the digital signal, which causes the signal to be
saturated. Distortion or sound cracking will occur.
[0005]
Next, the second case, that is, the case where distortion or sound break occurs when the
07-05-2019
2
reproduction limit of the speaker is exceeded will be described.
In speaker reproduction, there is a maximum displacement width that allows the diaphragm of
the speaker to swing. If a signal exceeding this is input, the speaker diaphragm can not vibrate
well, and distortion or sound break occurs. Here, the displacement width of the speaker
diaphragm depends on the frequency of the input signal. This relationship is shown in FIG. FIG. 3
shows the displacement width of the speaker diaphragm when only the signal frequency is
changed and input to the speaker without changing the voltage (V). Although the characteristics
in the vicinity of F0 in FIG. 3 actually swell or flatten more than in FIG. 3 due to the difference in
Q value etc. indicating the degree of braking of the speaker, the rough tendency does not change.
In addition, since the present invention can be applied to a speaker whose characteristic of
displacement width is different from the characteristic shown in FIG. 3, the characteristic of
displacement width of the speaker diaphragm is regarded as FIG. The following explanation is
given.
[0006]
As shown in FIG. 3, the displacement width of the speaker diaphragm has a substantially constant
value at frequency components lower than F0 (the lowest resonance frequency of the speaker),
and at frequency components higher than F0, the displacement width is approximately -12 dB /
oct. It will decrease. This indicates that the speaker diaphragm shakes with a larger displacement
width than inputting a low frequency component lower than the vicinity of F0 into the speaker
than inputting a high frequency component. Therefore, when a signal including a large amount of
low frequency components is input to the speaker and the voltage is increased, the maximum
displacement width of the diaphragm is exceeded at a certain voltage or more. That is, it can be
said that as the signal including more low frequency components and the voltage increase, the
reproduction limit of the speaker is likely to be exceeded. This situation is shown in FIG. In FIG. 4,
the vertical axis represents the amplitude intensity of the signal, and the horizontal axis
represents the frequency. Also, the area where the sound cracking occurs beyond the
displacement limit of the speaker diaphragm is shown in gray, and the boundary is shown by a
thick line. Here, since the characteristic of FIG. 4 is a characteristic with respect to the amplitude
value of the acoustic signal, unlike the characteristic of the displacement width of the speaker
shown in FIG. 3, the displacement limit of the speaker diaphragm has a slope of +12 dB / oct.
[0007]
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Reference numerals 401, 402, and 403 indicate the frequency characteristics of the sound signal
reproduced by the speaker, and in particular, it is assumed that the case contains a large number
of low frequency components. 401 is a characteristic when the volume value is small, 402 is a
characteristic when the volume value is medium, and 403 is a frequency characteristic when the
volume value is large. Since the sound signal containing many low frequency components does
not exceed the maximum displacement width of the speaker diaphragm for playback with a small
volume value such as 401, no sound break occurs and it can be enjoyed with the original sound
quality . However, if the sound volume is increased as in 402 and 403, the maximum
displacement width of the speaker diaphragm is exceeded, and distortion and sound breakage
occur to degrade the sound quality.
[0008]
As described above, when a signal that exceeds the maximum displacement width of the
diaphragm is input, the diaphragm can not vibrate well, and distortion or sound cracking occurs.
Distortions and sound breaks are sounds that are not included in the original sound signal, and
thus are a major obstacle to enjoying music.
[0009]
In order to solve this problem, distortion and sound breakup have conventionally been alleviated
by the processing configuration shown in FIG. In FIG. 17, a signal 1303 is output to an input
signal 1301 through an HPF (high pass filter) 1302 that suppresses low frequency components.
By adopting such a configuration, it is possible to suppress low frequency components that cause
sound cracking before being input to the speaker, so it is possible to reduce the rate of
occurrence of distortion or sound cracking. However, in the prior art, since the low frequency
components are suppressed by the HPF 1302, for example, even when the low frequency
components of the signal to be reproduced are small and the sound is not generated even when
the speaker is driven with a large voltage. There is a problem that the low frequency component
is always suppressed and the original sound can not be reproduced. In addition, even in the case
where sound breaking does not occur without passing through the high-pass filter 1302 without
driving with such a large voltage, the low frequency component is always suppressed, so the
problem that the original sound can not be reproduced is also included. is there. That is, in the
prior art, there is a problem that the low frequency component is excessively suppressed to
prevent the user from enjoying the original sound quality, in order to prevent sound separation
at the time of large voltage drive (large volume).
07-05-2019
4
[0010]
As a technique for alleviating this problem, there is a technique disclosed in Patent Document 1,
for example. FIG. 18 is a processing block of the amplitude limiting device disclosed in Patent
Document 1. According to Patent Document 1, in the amplitude limitation for suppressing
excessive input, the amount of distortion due to the amplitude limitation characteristic is
detected, and the gain for each frequency band is controlled based on this value, thereby
degrading the sound quality due to the amplitude limitation. It is relaxing.
[0011]
JP, 2009-147701, A
[0012]
However, in the technique disclosed in Patent Document 1 described above, since the frequency
component to be suppressed is limited to the divided bandwidth, the sound quality is degraded
due to excessive suppression of frequency components that are not necessarily suppressed.
There was a problem called.
For example, consider a case where the bandwidth divided into subbands by BPF (band pass
filter) is 100 Hz. At this time, if a signal having a large intensity at a frequency component of 60
Hz or less is input, sound distortion will not occur if only the signal component of 60 Hz or less is
suppressed originally. However, in the same technology, since the intensity of the entire 0 to 100
Hz signal component is suppressed, components other than the frequency component to be
suppressed (component of 60 to 100 Hz) are also suppressed. Also, as shown in FIG. 3, the
displacement width of the speaker amplitude plate has frequency characteristics, but the
amplitude limiting device disclosed in Patent Document 1 has a processing configuration that
reflects the frequency characteristics of the displacement width. Absent. For this reason, it can be
said that it does not have a function to prevent sound cracking caused by exceeding the
maximum displacement of the speaker diaphragm.
[0013]
The present invention has been made to solve the above-described problems, and an object of the
present invention is to provide a frequency characteristic deformation device capable of
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5
preventing distortion and sound breakage in speaker reproduction while maintaining sound
quality.
[0014]
The frequency characteristic modifying apparatus according to the present invention comprises a
filter for modifying the frequency characteristic of a target signal, a phase correction unit for
correcting the phase characteristic of the target signal and making the phase characteristic of the
filter approximately the same, and First multiplier for adjusting the gain of the signal output from
the second unit, the second multiplier for adjusting the gain of the signal output from the filter,
the gain factor of the first multiplier and the second multiplier A coefficient determination unit
that determines the gain coefficients of the first and second multipliers such that the sum of the
gain coefficients of the first and second gain coefficients becomes a constant value, and the two
signals output from the first multiplier and the second multiplier And an adder for adding.
[0015]
According to the present invention, it is possible to provide a frequency characteristic
deformation device that prevents distortion or sound breakage in speaker reproduction while
maintaining sound quality.
[0016]
FIG. 1 is a principle explanatory view of a frequency characteristic modification device according
to a first embodiment.
It is a figure which shows the relationship between the amplitude limit of a digital signal, and the
frequency characteristic of a sound source.
It is a figure which shows the displacement characteristic of a speaker diaphragm.
It is a figure which shows the relationship between the vibration limit of a speaker, and the
frequency characteristic of a sound source. FIG. 7 is an explanatory view showing transition of
frequency characteristics by two gains of the frequency characteristics modification device
according to the first embodiment. FIG. 8 is a principle explanatory view of a frequency
characteristic modification device according to a second embodiment. FIG. 16 is an explanatory
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6
drawing showing transition of frequency characteristics by two gains of the frequency
characteristic modification device according to the second embodiment. FIG. 10 is a principle
explanatory view of a frequency characteristic modification device according to a third
embodiment. FIG. 16 is an explanatory drawing showing the transition of the frequency
characteristic by the three gains of the frequency characteristic modification device according to
the third embodiment. FIG. 18 is a principle explanatory view of a frequency characteristic
modification device according to a fourth embodiment. FIG. 18 is a diagram showing an example
of a low band extraction unit of the frequency characteristic modification device according to the
fourth embodiment. FIG. 18 is a diagram showing another example of the low band extraction
unit of the frequency characteristic modification device according to the fourth embodiment. FIG.
18 is a diagram showing an added image of harmonics according to the low-pass attenuation
effect according to the fourth embodiment. FIG. 18 is a principle explanatory view of a frequency
characteristic modification device according to a fifth embodiment. FIG. 21 is a diagram showing
an added image of harmonics according to the low-pass attenuation effect according to the fifth
embodiment. FIG. 18 is a principle explanatory view of a frequency characteristic modification
device according to a sixth embodiment. It is principle explanatory drawing of a prior art. FIG. 1
is a processing block diagram of a prior art amplitude limiting device.
[0017]
Hereinafter, in order to explain the present invention in more detail, a mode for carrying out the
present invention will be described according to the attached drawings. Embodiment 1 FIG. 1 is a
view showing an embodiment of the present invention. The operation of this embodiment will be
described below. The input signal 101 input to the frequency characteristic transformation
device according to the present invention is branched and sent to the phase correction unit 701
and the HPF 702. The phase correction unit 701 corrects only the phase characteristic so as to
be substantially the same as the phase characteristic of the HPF 702 without changing the
frequency amplitude characteristic of the input signal, and obtains the obtained signal 703 as a
first multiplier 705 and an excessive signal. The signal is output to the input estimation unit 102.
The HPF 702 filters the input signal 101 and outputs the obtained signal 704 to the second
multiplier 706 and the overinput estimation unit 102.
[0018]
Here, a method of realizing the phase correction unit 701 that corrects the phase so as to have
substantially the same phase characteristic as the HPF 702 will be described. When the HPF 702
is realized in one stage of a second-order IIR filter, its phase characteristic rotates just 90 degrees
07-05-2019
7
at the cutoff frequency and gradually rotates up to 180 degrees in the frequency components
thereafter. The phase correction unit 701 that realizes such phase characteristics can be
configured by an all-pass filter using a first-order IIR filter. When the HPF is realized by two
stages of second-order IIR filters, the phase characteristic rotates just 180 degrees at the cutoff
frequency, and gradually rotates up to 360 degrees in the frequency components thereafter. The
phase correction unit that realizes such phase characteristics can be configured by an all-pass
filter using a second-order IIR filter. When the HPF is realized by N stages of second-order IIR
filters, the same phase characteristics can be realized by appropriately connecting all-pass filters
of first-order IIR and second-order IIR in series. When the HPF is realized by an FIR filter, the
phase characteristic becomes a linear phase, so that the phase correction unit 701 can be
configured by sample delay processing. Thus, it is possible to realize the phase correction unit
701 that has the same phase characteristics as the HPF 702.
[0019]
The excessive input estimation unit 102 of the present embodiment is configured of a speaker
diaphragm displacement estimation unit 501. The speaker diaphragm displacement estimation
unit 501 estimates the displacement value of the speaker diaphragm when the signal 703 is
reproduced, using the volume value and the information 502 of the target speaker, etc., and the
first speaker diaphragm displacement value 707. Ask for Similarly, a displacement value of the
speaker diaphragm when the signal 704 is reproduced is estimated, and a second speaker
diaphragm displacement value 708 is obtained. As a specific example of displacement value
estimation, a LPF by a second-order IIR filter with F0 as a cutoff frequency is prepared, and a
value roughly proportional to the displacement width of the target speaker is obtained by
multiplying this with the volume value through the input signal. Is required. Further, since the Q
value can be changed in the LPF based on the second-order IIR filter, the Q value can be changed
according to the degree of braking of the target speaker to improve the estimation accuracy. Of
course, the diaphragm displacement characteristics of the target speaker may be simulated by
another method, for example, an FIR filter. The two speaker diaphragm displacement values 707
and 708 thus obtained are output to the control unit 105.
[0020]
In the control unit 105, when the input two speaker diaphragm displacement values 707 and
708 are multiplied by different gain coefficients and then added, the absolute value of the
amplitude value falls within a predetermined threshold value. Find one gain factor. However, the
sum of the two gain factors is 1. By changing the two gain coefficients under such conditions,
07-05-2019
8
different low-pass attenuation effects can be realized. FIG. 5 shows two gains when the HPF 702
is realized by two stages of a second-order IIR filter with a cutoff frequency of 80 Hz, and the
phase correction unit 701 is realized by one stage of an all-pass filter with a second-order IIR
with a cutoff frequency 80 Hz. It shows the transition of the frequency characteristic by the
coefficient. In FIG. 5, assuming that the gain coefficient for the speaker diaphragm displacement
value 707 is A1 and the gain coefficient for the speaker diaphragm displacement value 708 is
A2, 301 has characteristics of A1 = 1.0 and A2 = 0.0, 302 Shows a characteristic of A1 = 0.1 and
A2 = 0.9, and 303 shows a characteristic of A1 = 0.0 and A2 = 1.0. Thus, from the completely flat
characteristics (A1 = 1.0, A2 = 0.0), the same characteristics (A1 = 0.0, A2 = 1.0) as the two-stage
IIR filter having a cutoff frequency of 80 Hz are obtained. It can be seen that different low-pass
attenuation characteristics can be realized between In addition, with respect to frequency
components higher than the cutoff frequency, components having the same phase are added at a
ratio of 1 in total, so that the flat characteristic can be maintained without increasing or
decreasing the intensity.
[0021]
As a specific calculation method of such two gain coefficients, the speaker diaphragm
displacement value 707 is X1, the speaker diaphragm displacement value 708 is X2, the gain
coefficient for X1 is A1, the gain coefficient for X2 is A2, predetermined Assuming that the
threshold value is T, this can be realized by finding A1 and A2 that satisfy the following equation
(1). T> ABS (X1 × A1 + X2 × A2) (1) A1 + A2 = 1 Here, ABS (x) represents the absolute value of
x.
[0022]
Further, in order to minimize the deformation of the frequency characteristic, it is preferable to
find a combination in which the value of A1 is close to 1 among the combinations of A1 and A2
satisfying the above-mentioned equation (1). This is because A1 is a signal based on a signal in
which only the phase characteristic is corrected, and as A1 approaches 1 the deformation of the
frequency characteristic decreases. In order to obtain such a gain coefficient, first, with A1 = 1,
the value of ABS (X1 × A1 + X2 × A2) is obtained while gradually reducing the value of A1, and
A1 at the time when it becomes smaller than T, A2 should be adopted.
[0023]
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9
A1 thus obtained is output to the first multiplier 705 as a gain coefficient 709. Further, A 2 is
output to the second multiplier 706 as a gain coefficient 710. The first multiplier 705 multiplies
the input signal 703 by the gain coefficient 709 and outputs the obtained signal 711 to the
adder 713. The second multiplier 706 multiplies the input signal 704 by the gain coefficient 710
and outputs the obtained signal 712 to the adder 713. The adder 713 adds the two input signals
711 and 712 and outputs the obtained signal as an output signal 107.
[0024]
As described above, the processing configuration of the first embodiment makes it possible to
prevent an excessive input of the reproduction acoustic signal. Therefore, according to the
present invention, an effect that distortion and sound break can be suppressed can be obtained.
Further, by lowering the cut-off frequency as much as possible by the control unit, it is possible
to obtain an effect that distortion or sound break can be prevented with a minimum change in
frequency characteristics.
[0025]
Second Embodiment By replacing the HPF 702 described in the first embodiment with the LPF,
in the digital acoustic signal subjected to frequency characteristic correction such that correction
of high frequency components is more frequent, the frequency characteristic of the signal does
not exceed the maximum amplitude of the digital signal. It is also possible to deform. FIG. 6 is a
processing configuration showing an embodiment in which the HPF 702 is replaced with an LPF.
The input signal 101 input to the frequency characteristic modification device according to the
present invention is branched and sent to the phase correction unit 701 and the LPF 901. The
phase correction unit 701 corrects only the phase characteristic so as to have substantially the
same characteristic as the phase characteristic of the LPF 901 without changing the frequency
amplitude characteristic of the input signal, and the obtained signal 703 is converted to the first
multiplier 705 and excessive. The signal is output to the input estimation unit 102. The LPF 901
filters the input signal 101, and outputs the obtained signal 902 to the second multiplier 706
and the overinput estimation unit 102. Here, since the phase correction unit 701 can be realized
by the all-pass filter or the sample delay processing as in the first embodiment, the detailed
description will be omitted.
[0026]
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10
The excessive input estimating unit 102 of the present embodiment is configured of a digital
signal amplitude calculating unit 601. The digital signal amplitude calculation unit 601 multiplies
the volume value 602 by the input signal 703 to obtain a first amplitude value 707. Similarly, the
volume value 602 and the input signal 902 are multiplied to obtain a second amplitude value
708. The two amplitude values 707 and 708 thus obtained are output to the control unit 105.
[0027]
In the control unit 105, when the two input amplitude values 707 and 708 are multiplied by two
different gain coefficients and then added, the two gains such that the absolute value of the
amplitude value falls within a predetermined threshold value Find the coefficients respectively.
However, the sum of the two gain factors is 1. Moreover, although a predetermined threshold
value is usually set to 0 dBFS, the present invention is not limited to this, and a match between
the speaker input resistance and the amplifier output is not achieved, and a value smaller than
this is required to limit the amplifier output. May be set.
[0028]
By changing the two gain coefficients under such conditions, different high frequency
attenuation effects can be realized. FIG. 7 shows two gains when the LPF 702 is realized by two
second-order IIR filters with a cutoff frequency of 6000 Hz, and the phase correction unit 701 is
realized with one second-order IIR all-pass filter with a cutoff frequency of 6000 Hz. It shows the
transition of the frequency characteristic by the coefficient. Further, in FIG. 7, assuming that the
gain coefficient for the speaker diaphragm displacement value 707 is A1 and the gain coefficient
for the speaker diaphragm displacement value 708 is A2, 801 has the characteristics of A1 = 1.0
and A2 = 0.0, 802. Shows a characteristic of A1 = 0.1 and A2 = 0.9, and 803 shows a
characteristic of A1 = 0.0 and A2 = 1.0. Thus, from the completely flat characteristic (A1 = 1.0,
A2 = 0.0), the characteristic (A1 = 0.0, A2 = 1.0) of the second-order IIR filter of the cutoff
frequency 6000 Hz is provided. It can be seen that different high-pass attenuation characteristics
can be realized. In addition, with respect to frequency components below the cutoff frequency,
components having the same phase are added at a ratio of 1 in total, so that the flat
characteristics can be maintained without increasing or decreasing the intensity. The specific
calculation method of such two gain coefficients can be obtained in the same manner as in the
first embodiment, so the description will be omitted.
07-05-2019
11
[0029]
A1 thus obtained is output to the first multiplier 705 as a gain coefficient 709. Further, A 2 is
output to the second multiplier 706 as a gain coefficient 710. The first multiplier 705 multiplies
the input signal 703 by the gain coefficient 709 and outputs the obtained signal 711 to the
adder 713. The second multiplier 706 multiplies the input signal 902 by the gain coefficient 710,
and outputs the obtained signal 712 to the adder 713. The adder 713 adds the two input signals
711 and 712 and outputs the obtained signal as an output signal 107.
[0030]
As described above, according to the processing configuration of the second embodiment, it is
possible to suppress the amplitude value of the digital acoustic signal that corrects many high
frequency components, and it is possible to suppress distortion and sound breakup. Is obtained.
Further, in the control unit of the present embodiment, since the cutoff frequency of the LPF can
be made as large as possible, the effect that distortion and sound breakage can be prevented can
be obtained by the change of the minimum required frequency characteristic.
[0031]
Third Embodiment In the first and second embodiments, the frequency characteristic modifying
unit is realized by one phase correcting unit and one HPF or LPF. However, the present invention
is not limited to this, and a plurality of phase correcting units and a plurality of HPF or LPF may
be used. A characteristic deformation unit may be realized. FIG. 8 is a diagram showing an
example in which a frequency characteristic deformation unit is realized by three phase
correction units and three HPFs. The operation of this embodiment will be described below. The
input signal 101 input to the frequency characteristic transformation apparatus according to the
present invention is branched into three and sent to the first HPF 1101, the second HPF 1102,
and the third HPF 1103. The first HPF 1101 filters the input signal and outputs the obtained
signal 1104 to the first phase correction unit 1107. The second HPF 1102 filters the input signal
and outputs the obtained signal 1105 to the second phase correction unit 1108. The third HPF
1103 filters the input signal, and outputs the obtained signal 1106 to the third phase correction
unit 1109.
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[0032]
In the first phase correction unit 1107, without changing the frequency amplitude characteristic
of the signal, only the phase characteristic is obtained so as to be substantially the same as the
phase characteristic when both the second HPF 1102 and the third HPF 1103 are processed. The
corrected signal 1110 is output to the first multiplier 1113 and the excessive input estimation
unit 102 after correction. In the second phase correction unit 1108, without changing the
frequency amplitude characteristic of the signal, only the phase characteristic is obtained so as to
be substantially the same as the phase characteristic when both the first HPF 1101 and the third
HPF 1103 are processed. The corrected signal 1111 is output to the second multiplier 1114 and
the overinput estimation unit 102. In the third phase correction unit 1109, without changing the
frequency amplitude characteristic of the signal, only the phase characteristic is obtained so as to
be substantially the same as the phase characteristic when both the first HPF 1101 and the
second HPF 1102 are processed. The corrected signal 1112 is output to the third multiplier
1115 and the excessive input estimation unit 102 after correction.
[0033]
Here, since each phase correction unit can be realized by an all-pass filter or sample delay
processing as in the first embodiment, detailed description will be omitted. The excessive input
estimation unit 102 of the present embodiment is configured of a speaker diaphragm
displacement estimation unit 501. The loudspeaker diaphragm displacement estimation unit 501
estimates the displacement value of the loudspeaker diaphragm when the signal 1110 is
reproduced, using the volume value and the information 502 of the target loudspeaker, etc., and
the first loudspeaker diaphragm displacement value 1116 Ask for Similarly, a displacement value
of the speaker diaphragm when the signal 1111 is reproduced is estimated, and a second
speaker diaphragm displacement value 1117 is obtained. Similarly, a displacement value of the
speaker diaphragm when the signal 1112 is reproduced is estimated, and a third speaker
diaphragm displacement value 1118 is obtained. As a specific example of displacement value
estimation, since it is obtained by the same method as that of the first embodiment, detailed
description will be omitted.
[0034]
The three speaker diaphragm displacement values 1116, 1117, and 1118 thus obtained are
output to the control unit 105. In the control unit 105, when the input three speaker diaphragm
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13
displacement values 1116, 1117 and 1118 are multiplied by different gain coefficients and then
added, the absolute value of the amplitude value falls within the predetermined threshold value.
Find three gain coefficients. However, the sum of the three gain factors is 1.
[0035]
By changing the three gain coefficients under such conditions, different low-pass attenuation
effects can be realized. In FIG. 9, the first HPF 1101 is realized by two stages of second-order IIR
filters with a cutoff frequency of 30 Hz, the second HPF 1102 is realized with two stages of
second-order IIR filters with a cutoff frequency of 70 Hz, and the third HPF 1103 Is realized by
four second-order IIR filters with a cut-off frequency of 140 Hz, and the first phase correction
unit 1107 consists of one second-order IIR filter with a cut-off frequency of 70 Hz and two
second-order IIR filters with a cut-off frequency of 140 Hz. The second phase correction unit
1108 is realized by serially connecting one stage of a second-order IIR filter with a cutoff
frequency of 30 Hz and two stages of a second-order IIR filter with a cutoff frequency of 140 Hz.
Three phase correction units 1109 have a cutoff frequency of 30 Hz and a second order IIR filter
of one stage and a cutoff frequency of 70 Hz second order It represents the transition of the
frequency characteristics due to the three gain factors when implemented by connecting the 1stage IR filter in series.
[0036]
Further, in FIG. 9, assuming that the gain coefficient for the speaker diaphragm displacement
value 1116 is A1, the gain coefficient for the speaker diaphragm displacement value 1117 is A2,
and the gain coefficient for the speaker diaphragm displacement value 1118 is A3, 901 is A1 = 1.
.0, A2 = 0.0, A3 = 0.0, 902 is A1 = 0.1, A2 = 0.9, A3 = 0.0, and 903 is A1 = 0.0, A2 = 1.0, A3 = 0.0
characteristic, 904 is A1 = 0.0, A2 = 0.1, A3 = 0.9 characteristic, 905 is A1 = 0.0, A2 = 0.0, The
characteristic of A3 = 1.0 is shown. Thus, from the characteristics (A1 = 1.0, A2 = 0.0, A3 = 0.0)
of the two-stage IIR filter with a cutoff frequency of 30 Hz, the four-stage IIR filter with a cutoff
frequency of 140 Hz is obtained. It can be seen that different low-pass attenuation characteristics
can be realized up to the characteristics (A1 = 0.0, A2 = 0.0, A3 = 1.0). In addition, with respect to
frequency components higher than the cutoff frequency, components having the same phase are
added at a ratio of 1 in total, so that the flat characteristic can be maintained without increasing
or decreasing the intensity.
[0037]
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14
In addition, as a specific calculation method of such three gain coefficients, the speaker
diaphragm displacement value 1116 is X1, the speaker diaphragm displacement value 1117 is
X2, and the speaker diaphragm displacement value 1118 is X3, the gain coefficient for X1 is
Assuming that the gain coefficients for A1 and X2 are A2, the gain coefficient for X3 is A3, and
the predetermined threshold is T, this can be realized by obtaining A1, A2 and A3 satisfying the
following equation (2). T> ABS (X1 × A1 + X2 × A2 + X3 × A3) (2) A1 + A2 + A3 = 1 Here, ABS
(x) represents the absolute value of x.
[0038]
A1 thus obtained is output to the first multiplier 1113 as a gain coefficient 1119. Further, A2 is
output to the second multiplier 1114 as a gain coefficient 1120. Further, A3 is output to the
third multiplier 1115 as a gain coefficient 1121. The first multiplier 1113 multiplies the input
signal 1110 by the gain coefficient 1119, and outputs the obtained signal 1122 to the adder
713. The second multiplier 1114 multiplies the input signal 1111 by the gain coefficient 1120
and outputs the obtained signal 1123 to the adder 713. The third multiplier 1115 multiplies the
input signal 1112 by the gain coefficient 1121 and outputs the obtained signal 1124 to the
adder 713. The adder 713 adds the three input signals 1122, 1123 and 1124 and outputs the
obtained signal as an output signal 107.
[0039]
As described above, according to the processing configuration of the third embodiment, the
frequency characteristic deformation unit can be realized with three phase corrections and three
HPFs, and the frequency characteristic deformation unit is a typical HPF than the first
embodiment. The effect of being able to realize the characteristic close to is obtained. Of course,
by increasing the number of phase corrections and the number of HPFs, it is possible to realize
characteristics closer to the normal HPF. In addition, by replacing the HPF of this configuration
with an LPF, in the digital acoustic signal subjected to frequency characteristic correction such
that correction of high frequency components is frequently performed, the frequency
characteristic of the digital signal is modified so as not to exceed the maximum amplitude of the
digital signal. It is also possible to
[0040]
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Fourth Embodiment FIG. 10 is a view showing another embodiment of the present invention. In
the present embodiment, an example is shown in which the harmonics are generated and added
to the low band cut by the high pass filter. The operation of this embodiment will be described
below. The input signal 101 input to the signal processing apparatus according to the present
invention is branched and sent to the phase correction unit 701 and the HPF 702. The phase
correction unit 701 corrects only the phase characteristic so that the characteristic is
substantially the same as the phase characteristic of the HPF 702 without changing the
frequency amplitude characteristic of the input signal. The signal is output to the input
estimation unit 102 and the harmonic signal generation unit 2001. The HPF 702 filters the input
signal 101 and outputs the obtained signal 704 to the adder 2003. Here, since the phase
correction unit can be realized by the all-pass filter or the sample delay processing as in the first
embodiment, the detailed description will be omitted.
[0041]
The harmonic signal generation unit 2001 includes a low band extraction unit 2004, a harmonic
generation unit 2006, and a multiplier 2008. The low band extraction unit 2004 receives the
output signal 703 of the phase correction unit 701, extracts the low band cut by the HPF 702,
and outputs the obtained signal 2005 to the harmonic generation unit 2006. Here, the low band
extraction unit 2004 for extracting the low band cut by the HPF 702 is configured of the
subtractor 2011 as shown in FIG. 11 and the output signal 703 of the phase correction unit 701
is the output signal 704 of the HPF 702. There is a method to realize by subtracting by, or a
method to realize by making the output signal 703 of the phase correction unit 701 pass
through a filter similar to the filter specification of the HPF 702, as shown in FIG.
[0042]
The harmonic generation unit 2006 generates harmonics of the output signal 2005 of the lowfrequency extraction unit 2004 up to the n-th (n is an integer of 3 or more), and outputs the
obtained signal 2007 to the multiplier 2008. Here, the method of realizing the harmonic
generation unit 2006 is waveform generation such as peak hold, full wave rectification, half wave
rectification, etc., m times multiplication of the signal 2005 (m is an integer), division, etc. It is
sufficient to generate both harmonics and even harmonics. The multiplier 2008 multiplies the
output signal 2007 of the harmonic generation unit 2006 by a gain factor preferred by the user,
and outputs the obtained signal 2002 to the adder 2003. Here, the gain coefficient to be
07-05-2019
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multiplied by the multiplier 2008 is prepared in advance by preparing a plurality of fixed gain
coefficients, and is changed according to the preference of the user. The adder 2003 adds the
two input signals 704 and 2002, and outputs the obtained signal 2003 ′ to the overinput
estimation unit 102 and the second multiplier 706.
[0043]
The excessive input estimation unit 102 of the present embodiment is configured of a speaker
diaphragm displacement estimation unit 501. The speaker diaphragm displacement estimation
unit 501 estimates the displacement value of the speaker diaphragm when the signal 703 is
reproduced, using the volume value and the information 502 of the target speaker, etc., and the
first speaker diaphragm displacement value 707. Ask for Similarly, a displacement value of the
speaker diaphragm when the signal 2003 ′ is reproduced is estimated, and a second speaker
diaphragm displacement value 708 is obtained. As a specific example of displacement value
estimation, since it is obtained by the same method as that of the first embodiment, detailed
description will be omitted.
[0044]
The two loudspeaker diaphragm displacement values 707 and 708 determined by the
loudspeaker diaphragm displacement estimation unit 501 are output to the control unit 105. In
the control unit 105, when the input two speaker diaphragm displacement values 707 and 708
are multiplied by different gain coefficients and then added, the absolute value of the amplitude
value falls within a predetermined threshold value. Find one gain factor. However, the sum of the
two gain factors is 1. By changing the two gain coefficients under such conditions, different lowpass attenuation effects can be realized. A specific example of the low-pass attenuation effect has
the same characteristics as in the first embodiment, and thus the detailed description is omitted.
[0045]
In addition, it is possible to change the amount of addition of the cut low band harmonics
according to the low band attenuation effect. FIG. 13 shows low pass attenuation when the HPF
702 is realized by two stages of second-order IIR filters with a cutoff frequency of 80 Hz, and the
phase correction means 701 is realized by one stage of all-pass filters with a second-order IIR
with a cutoff frequency 80 Hz. It represents a summed image of harmonics according to the
07-05-2019
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effect. In FIG. 13, assuming that the gain coefficient for the speaker diaphragm displacement
value 707 is A1 and the gain coefficient for the speaker diaphragm displacement value 708 is
A2, 2021 is a characteristic image of A1 = 0.0 and A2 = 1.0, and 2022 is A characteristic image
of A1 = 0.1, A2 = 0.9, 2023 shows a characteristic image of A1 = 0.5, A2 = 0.5. From this, it is
understood that as the value of A2 is larger and the amount of attenuation in the lower band is
larger, more harmonics in the lower band of 80 Hz or less are added. Further, since the specific
calculation method of the gain coefficients A1 and A2 can be obtained in the same manner as in
the first embodiment, the description will be omitted.
[0046]
A1 obtained by the control unit 105 is output to the first multiplier 705 as a gain coefficient 709.
Also, A 2 is output to the second multiplier 706 as a gain coefficient 710. The first multiplier 705
multiplies the input signal 703 by the gain coefficient 709 and outputs the obtained signal 711
to the adder 713. The second multiplier 706 multiplies the input signal 2003 ′ by the gain
coefficient 710, and outputs the obtained signal 712 to the adder 713. The adder 713 adds the
two input signals 711 and 712 and outputs the obtained signal as an output signal 107.
[0047]
As described above, according to the processing configuration of the fourth embodiment, the
lower harmonics cut by the frequency characteristic deformation unit are generated up to the nth order in the harmonic signal generation unit (n is an integer of 3 or more). The effect "Missing
fundamental" has the effect of making it possible to artificially feel the low region cut. In
addition, since the control unit controls the output signal of the HPF and the gain of the
generated harmonic signal together, it is possible to change the low-frequency interpolation
effect according to the attenuation characteristic of the low region. Here, the “missing
fundamental” is an illusion that when the sound of two or more frequencies is heard, the sound
of the difference frequency is heard.
[0048]
Embodiment 5 In the fifth embodiment, the frequency characteristic deformation unit is realized
by one phase correction unit and one HPF, but the frequency characteristic deformation unit is
realized by a plurality of phase correction units and a plurality of HPFs. It is good. FIG. 14 is a
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diagram illustrating an example in which a frequency characteristic deformation unit is realized
by three phase correction units and three HPFs. The operation of this embodiment will be
described below. The input signal 101 input to the signal processing apparatus according to the
present invention is branched into six, and is applied to the first HPF 1101, the second HPF
1102, the third HPF 1103, the first LPF 2101, the second LPF 2102, and the third LPF 2103.
Sent. The first HPF 1101 filters the input signal and outputs the obtained signal 1104 to the first
phase correction unit 1107. The second HPF 1102 filters the input signal and outputs the
obtained signal 1105 to the second phase correction unit 1108. The third HPF 1103 filters the
input signal, and outputs the obtained signal 1106 to the third phase correction unit 1109.
[0049]
In the first phase correction unit 1107, without changing the frequency amplitude characteristic
of the signal, only the phase characteristic is obtained so as to be substantially the same as the
phase characteristic when both the second HPF 1102 and the third HPF 1103 are processed. The
corrected signal 1110 is output to the adder 2125. In the second phase correction unit 1108,
without changing the frequency amplitude characteristic of the signal, only the phase
characteristic is obtained so as to be substantially the same as the phase characteristic when both
the first HPF 1101 and the third HPF 1103 are processed. The corrected signal 1111 is output to
the adder 2126. In the third phase correction unit 1109, without changing the frequency
amplitude characteristic of the signal, only the phase characteristic is obtained so as to be
substantially the same as the phase characteristic when both the first HPF 1101 and the second
HPF 1102 are processed. The corrected signal 1112 is output to the adder 2127. Here, since
each phase correction unit can be realized by an all-pass filter or sample delay processing as in
the first embodiment, detailed description will be omitted.
[0050]
The first LPF 2101 processes the input signal 101 with a filter having a filter specification
similar to that of the first HPF 1101, and outputs the obtained signal 2104 to the fourth phase
correction unit 2107. The second LPF 2102 processes the input signal 101 with a filter having a
filter specification similar to that of the second HPF 1102, and outputs the obtained signal 2105
to the fifth phase correction unit 2108. The third LPF 2103 processes the input signal 101 with
a filter having the same filter specification as the third HPF 1103, and outputs the obtained
signal 2106 to the sixth phase correction unit 2109.
07-05-2019
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[0051]
In the fourth phase correction unit 2107, without changing the frequency amplitude
characteristic of the signal, only the phase characteristic is obtained so as to be substantially the
same as the phase characteristic when both the second LPF 2102 and the third LPF 2103 are
processed. The corrected signal 2110 is output to the first harmonic generation unit 2113. In the
fifth phase correction unit 2108, without changing the frequency amplitude characteristic of the
signal, only the phase characteristic is obtained so as to be substantially the same as the phase
characteristic when both the first LPF 2101 and the third LPF 2103 are processed. The corrected
signal 2111 is output to the second harmonic generation unit 2114 after correction. In the sixth
phase correction unit 2109, without changing the frequency amplitude characteristic of the
signal, only the phase characteristic is obtained so as to be substantially the same as the phase
characteristic when both the first LPF 2101 and the second LPF 2102 are processed. The
corrected signal 2112 is output to the third harmonic generation unit 2115.
[0052]
The first harmonic generation unit 2113 generates harmonics of the signal 2110 and outputs the
obtained signal 2116 to the multiplier 2119. The second harmonic generation unit 2114
generates harmonics of the signal 2111 and outputs the obtained signal 2117 to the multiplier
2120. The third harmonic generation unit 2115 generates harmonics of the signal 2112 and
outputs the obtained signal 2118 to the multiplier 2121.
[0053]
Here, the method of realizing each harmonic generation unit is waveform generation such as
peak hold, full wave rectification, half wave rectification, etc., m multiplication of the signal (m is
an integer), division, etc. What is necessary is just to produce | generate both a wave and an
even-order harmonic. The multiplier 2119 multiplies the signal 2116 by the user's favorite gain
coefficient, and outputs the obtained signal 2122 to the adder 2125. The multiplier 2120
multiplies the signal 2117 by the user's favorite gain coefficient, and outputs the obtained signal
2123 to the adder 2126. The multiplier 2121 multiplies the signal 2118 by the user's favorite
gain coefficient, and outputs the obtained signal 2124 to the adder 2127. Here, the gain
coefficients to be multiplied by the multipliers 2119, 2120, and 2121 are prepared in advance
by preparing a plurality of fixed gain coefficients, and are changed according to the user's
preference.
07-05-2019
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[0054]
The adder 2125 adds the two input signals 1110 and 2122 and outputs the obtained signal
2128 to the overinput estimation unit 102 and the first multiplier 1113. The adder 2126 adds
the two input signals 1111 and 2123, and outputs the obtained signal 2129 to the overinput
estimation unit 102 and the second multiplier 1114. The adder 2127 adds the two input signals
1112 and 2124 and outputs the obtained signal 2130 to the overinput estimation unit 102 and
the third multiplier 1115.
[0055]
The excessive input estimation unit 102 is configured of a speaker diaphragm displacement
estimation unit 501. The speaker diaphragm displacement estimation unit 501 estimates the
displacement value of the speaker diaphragm when the signal 2128 is reproduced, using
information 502 such as the volume value and the F0 of the target speaker, and the first speaker
diaphragm displacement value 1116 Ask for Similarly, a displacement value of the speaker
diaphragm when the signal 2129 is reproduced is estimated, and a second speaker diaphragm
displacement value 1117 is obtained. Similarly, a displacement value of the speaker diaphragm
when the signal 2130 is reproduced is estimated, and a third speaker diaphragm displacement
value 1118 is obtained. As a specific example of displacement value estimation, since it is
obtained by the same method as that of the first embodiment, detailed description will be
omitted.
[0056]
The three speaker diaphragm displacement values 1116, 1117, and 1118 determined by the
speaker diaphragm displacement estimation unit 501 are output to the control unit 105. In the
control unit 105, when the input three speaker diaphragm displacement values 1116, 1117 and
1118 are multiplied by different gain coefficients and then added, the absolute value of the
amplitude value falls within the predetermined threshold value. Find three gain coefficients.
However, the sum of the three gain factors is 1. By changing the three gain coefficients under
such conditions, different low-pass attenuation effects can be realized. A specific example of the
low-pass attenuation effect has the same characteristics as in the third embodiment, so the
detailed description will be omitted.
07-05-2019
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[0057]
In addition, it is possible to change the amount of addition of the cut low band harmonics
according to the low band attenuation effect. In FIG. 15, the first HPF 1101 is realized by two
second-order IIR filters having a cutoff frequency of 30 Hz, the second HPF 1102 is realized by
two second-order IIR filters having a cutoff frequency of 70 Hz, and the third HPF 1103 Is
realized by four second-order IIR filters with a cut-off frequency of 140 Hz, and the first phase
correction unit 1107 includes one second-order IIR filter with a cut-off frequency of 70 Hz and
two second-order IIR filters with a cut-off frequency of 140 Hz The second phase correction unit
1108 is realized by serially connecting one stage of a second-order IIR filter with a cutoff
frequency of 30 Hz and two stages of a second-order IIR filter with a cutoff frequency of 140 Hz.
Three phase correction units 1109: one stage of a second-order IIR filter with a cutoff frequency
of 30 Hz and two stages with a cutoff frequency of 70 Hz It illustrates a sum image of the
harmonic corresponding to the low-frequency damping effect when implemented by connecting
the IIR filter 1 stage in series.
[0058]
In FIG. 15, assuming that the gain coefficient for the speaker diaphragm displacement value
1116 is A1, the gain coefficient for the speaker diaphragm displacement value 1117 is A2, and
the gain coefficient for the speaker diaphragm displacement value 1118 is A3, 2131 is A1 = 1.0 ,
A2 = 0.0, A3 = 0.0 characteristic image, 2132: A1 = 0.9, A2 = 0.1, A3 = 0.0 characteristic image,
2133: A1 = 0.0, A2 = The characteristic image of 0.0 and A3 = 1.0 is shown. From this, it can be
seen that the low frequency harmonics cut by the HPF are added to the frequency band above
the cut band. Further, specific calculation methods of the gain coefficients A1, A2, and A3 can be
obtained in the same manner as in the third embodiment, and thus the description thereof will be
omitted.
[0059]
A1 obtained by the control unit 105 is output to the first multiplier 1113 as a gain coefficient
1119. Further, A2 is output to the second multiplier 1114 as a gain coefficient 1120. Further, A3
is output to the third multiplier 1115 as a gain coefficient 1121. The first multiplier 1113
multiplies the input signal 2128 by the gain coefficient 1119, and outputs the obtained signal
07-05-2019
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1122 to the adder 713. The second multiplier 1114 multiplies the input signal 2129 by the gain
coefficient 1120 and outputs the obtained signal 1123 to the adder 713. The third multiplier
1115 multiplies the input signal 2130 by the gain coefficient 1121 and outputs the obtained
signal 1124 to the adder 713. The adder 713 adds the three input signals 1122, 1123 and 1124
and outputs the obtained signal as an output signal 107.
[0060]
As described above, according to the processing configuration of the fifth embodiment, the lower
harmonics cut by the frequency characteristic deformation unit are generated in the harmonic
signal generation unit up to the n th order (n is an integer of 3 or more). The effect "Missing
fundamental" has the effect of making it possible to artificially feel the low region cut. In
addition, since the control unit controls the output signal of the HPF and the gain of the
generated harmonic signal together, it is possible to change the low-frequency interpolation
effect according to the attenuation characteristic of the low region.
[0061]
Sixth Embodiment In the fourth and fifth embodiments, the output signal 2002 of the harmonic
signal generation unit 2001 is added to the output signal of the HPF, but the output signal 2002
of the harmonic signal generation unit 2001 is changed to the frequency characteristic
transformation unit The configuration may be such that the addition is performed after the step
103. FIG. 16 is a diagram illustrating an example of a configuration in which the addition
position of the output signal 2002 of the harmonic signal generation unit 2001 is changed to the
subsequent stage of the frequency characteristic modification unit 103 in the fourth
embodiment. The operation of this embodiment will be described below. The input signal 101
input to the signal processing apparatus according to the present invention is branched and sent
to the phase correction unit 701 and the HPF 702. The phase correction unit 701 corrects only
the phase characteristic so that the characteristic is substantially the same as the phase
characteristic of the HPF 702 without changing the frequency amplitude characteristic of the
input signal. The signal is output to the input estimation unit 102 and the harmonic signal
generation unit 2001. The HPF 702 filters the input signal 101, and outputs the obtained signal
704 to the multiplier 706 and the overinput estimation unit 102. Here, since the phase
correction unit can be realized by the all-pass filter or the sample delay processing as in the first
embodiment, the detailed description will be omitted.
07-05-2019
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[0062]
The excessive input estimation unit 102 of the present embodiment is configured of a speaker
diaphragm displacement estimation unit 501. The speaker diaphragm displacement estimation
unit 501 estimates the displacement value of the speaker diaphragm when the signal 703 is
reproduced, using the volume value and the information 502 of the target speaker, etc., and the
first speaker diaphragm displacement value 707. Ask for Similarly, a displacement value of the
speaker diaphragm when the signal 704 is reproduced is estimated, and a second speaker
diaphragm displacement value 708 is obtained. As a specific example of displacement value
estimation, since it is obtained by the same method as that of the first embodiment, detailed
description will be omitted.
[0063]
The two speaker diaphragm displacement values 707 and 708 determined by the speaker
diaphragm displacement estimation unit 501 are output to the control unit 105. In the control
unit 105, when the input two speaker diaphragm displacement values 707 and 708 are
multiplied by different gain coefficients and then added, the absolute value of the amplitude
value falls within a predetermined threshold value. Find one gain factor. However, the sum of the
two gain factors is 1. By changing the two gain coefficients under such conditions, different lowpass attenuation effects can be realized. A specific example of the low-pass attenuation effect has
the same characteristics as in the first embodiment, and thus the detailed description is omitted.
In FIG. 16, assuming that the gain coefficient for the speaker diaphragm displacement value 707
is A1 and the gain coefficient for the speaker diaphragm displacement value 708 is A2, the
specific calculation method of the gain coefficients A1 and A2 is the same as the first
embodiment. The description is omitted because it is similarly required.
[0064]
A1 obtained by the control unit 105 is output to the first multiplier 705 as a gain coefficient 709.
Further, A2 is output to the second multiplier 706 and the multiplier 2201 as the gain coefficient
710. The first multiplier 705 multiplies the input signal 703 by the gain coefficient 709 and
outputs the obtained signal 711 to the adder 713. The second multiplier 706 multiplies the input
signal 704 by the gain coefficient 710 and outputs the obtained signal 712 to the adder 713. The
adder 713 adds the two input signals 711 and 712 and outputs the obtained signal 2203 to the
addition 2204.
07-05-2019
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[0065]
The harmonic signal generation unit 2001 includes a low band extraction unit 2004, a harmonic
generation unit 2006, and a multiplier 2008. The low band extraction unit 2004 receives the
output signal of the phase correction unit 701, extracts the low band cut by the HPF 702, and
outputs the obtained signal 2005 to the harmonic generation unit 2006. Here, a method of
realizing the low band extracting unit 2004 for extracting the low band to be cut by the HPF 702
is obtained in the same manner as the fourth embodiment, and thus the description thereof is
omitted.
[0066]
The harmonic generation unit 2006 generates harmonics of the output signal 2005 of the lowfrequency extraction unit 2004 up to the n-th (n is an integer of 3 or more), and outputs the
obtained signal 2007 to the multiplier 2008. Here, the method of realizing the harmonic
generation unit 2006 is waveform generation such as peak hold, full wave rectification, half wave
rectification, etc., m times multiplication of the signal 2005 (m is an integer), division, etc. It is
sufficient to generate both harmonics and even harmonics.
[0067]
The multiplier 2008 multiplies the output signal 2007 of the harmonic generation unit 2006 by
the gain coefficient preferred by the user, and outputs the obtained signal 2002 to the multiplier
2201. Here, the gain coefficient to be multiplied by the multiplier 2008 is prepared in advance
by preparing a plurality of fixed gain coefficients, and is changed according to the preference of
the user. The multiplier 2201 multiplies the input signal 2002 by the gain coefficient 710, and
outputs the obtained signal 2202 to the adder 2204. Here, the multiplier 2201 can change the
amount of addition of the cut low-band harmonics according to the low-pass attenuation effect.
About the addition image of the harmonic according to a low-pass attenuation effect, since it
becomes the same as that of Example 4, description is abbreviate | omitted. The adder 2204 adds
the two input signals 2202 and 2203 and outputs the obtained signal as an output signal 107.
[0068]
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As described above, according to the processing configuration of the sixth embodiment, the
harmonic signal generation unit generates low-order harmonics cut by the frequency
characteristic deformation unit up to the nth order (n is an integer of 3 or more). The effect
"Missing fundamental" has the effect of making it possible to artificially feel the low region cut. In
addition, since the control unit controls the output signal of the HPF and the gain of the
generated harmonic signal together, it is possible to change the low-frequency interpolation
effect according to the attenuation characteristic of the low region.
[0069]
In the scope of the invention, the present invention allows free combination of each embodiment,
or modification of any component of each embodiment, or omission of any component in each
embodiment. .
[0070]
As described above, the frequency characteristic deformation device according to the present
invention can improve distortion and sound breakup in sound signal reproduction, and can be
used for an audio reproduction device or the like.
[0071]
101 input signal, 102 excessive input estimation unit, 103 frequency characteristic deformation
unit, 105 control unit, 107 output signal, 501 speaker diaphragm displacement estimation unit,
502 information, 701 phase correction unit, 702 HPF, 705 first multiplier, 706 Second
multiplier, 713, 2003, 2125, 2126, 2127, 2204 Adder, 901, 2012 LPF, 1101 first HPF, 1102
second HPF, 1103 third HPF, 1107 first phase correction Part, 1108 Second phase correction
unit, 1109 Third phase correction unit, 1113 First multiplier, 1114 Second multiplier, 1115
Third multiplier, 2001 Harmonic signal generation unit, 2004 Low frequency extraction Part,
2006 harmonics generation part, 2008, 2119, 2120, 211, 221, multiplier, 2011 differencer, 2
01 first LPF, 2102 second LPF, 2103 third LPF, 2107 fourth phase correction unit, 2108 fifth
phase correction unit, 2109 sixth phase correction unit, 2113 first harmonic generation unit ,
2114 second harmonic generator, 2115 third harmonic generator.
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