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DESCRIPTION JPH0965481

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DESCRIPTION JPH0965481
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an
audio processing circuit having a microphone and capable of recording and reproduction. More
specifically, the invention relates to a technique for reducing wind noise of a microphone.
[0002]
2. Description of the Related Art As an example of a recording and reproducing apparatus having
a built-in microphone, a tape recorder and a camera integrated video can be mentioned. In
particular, camera-integrated video is often used outdoors for recording, and it is susceptible to
wind when recording, and wind noise specific to a microphone affects the intended sound.
[0003]
In order to prevent this effect, it is known that attaching a windshield to the microphone is the
most effective and superior in terms of performance. However, with this method, the windshield
covering the microphone becomes quite large. In particular, in the case of a product such as a
camera integrated video in which small and light weight is a feature of the product, the
windshield as described above is an inappropriate process.
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[0004]
Therefore, a sound processing circuit as shown in FIG. 2 can be considered because of the nature
in which wind noise is widely distributed in the low tone range. In FIG. 2, voice centered on the
left side is input from the input terminal 21, amplified by the amplifier 23 and directly connected
to the changeover switch 27, and connected to the switch 27 through a high pass filter
(hereinafter HPF) 25. There is a route to be done. The output of the switch 27 is connected to the
output terminal 29 and output. Similarly, the voice centered on the right side is input from the
input terminal 22, and there is a path that is amplified by the amplifier 24 and directly connected
to the changeover switch 28, and a path connected to the switch 28 through the HPF 26. The
switch 28 output is connected to the output terminal 30 and output.
[0005]
Usually, when there is no influence of wind, the changeover switches 27 and 28 are inclined in
the off direction, and the amplified signal is output as it is. In a windy state, the switch 27 and the
switch 28 are turned on to pass the HPF to attenuate the low frequency component including a
lot of wind noise components, thereby reducing the level of the wind noise. . The wind noise
reduction effect can be changed by the setting of the HPF.
[0006]
However, in the conventional method as described above, when setting the HPF is effective for
reducing wind noise, the bass range of the target voice is also cut off due to the effect. The sound
quality was unnatural. Moreover, if it is going to reduce this influence, the reduction effect of
wind noise will also decrease, and contradiction had arisen in making it compatible.
[0007]
Therefore, an object of the present invention is to provide a voice processing circuit that
attenuates only the extra wind noise component without removing the low frequency component
of the target voice.
[0008]
SUMMARY OF THE INVENTION In order to achieve the above object, the invention of claim 1 is
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characterized in that a first input terminal (1) to which an audio signal centered on the left side is
input, and a center on the right side A second input terminal (2) to which an audio signal to be
input is input, a first delay means (5) connected to the first input terminal (1), and the second
input terminal (2) A second delay means (6) connected; a first subtraction means (7) connected
to the first input terminal (1) and the second delay means (6); A second subtraction means (8)
connected to the terminal (2) and the first delay means (5), and a first low pass means (9)
connected to the first subtraction means (7) And a first high pass means (11) and a second
reduction pass means (10) connected to the second subtraction means (8) and a second high
pass means (12). A first addition means (15) connected to the first low-pass means (9) and the
second low-pass means (10), and the first high-pass means (11) A second addition means
connected to the first addition means (15), (13), a third high pass means (12) and a third addition
means connected to the first addition means (15) And an adding means (14).
[0009]
The invention of claim 2 further comprises the cutoff frequencies of the first low pass means (9)
and the second low pass means (10), the first high pass means (11) and the first high pass means
(11) 2. A speech processing circuit according to claim 1, characterized in that the cutoff
frequency of the two high-pass means (12) is identical.
[0010]
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS First, an outline of an embodiment
according to the present invention will be described with reference to FIG. 1 in which an audio
signal low frequency component LL + voice high frequency component LH + wind noise
component WL + dRL + dRH is inputted to a first input terminal 1; The voice signal low-pass
component RL + the voice signal high-pass component RH + the wind noise component WR + dLL
+ dLH is input to the second input terminal 2.
[0011]
The output signal of the LPF 5 which is the delay means is the high-frequency component I + WL
that is LL + delayed and attenuated.
The output signal of the delay circuit LPF 6 is RL + delayed and attenuated high-frequency
component r + WR.
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[0012]
Therefore, the input signal of the subtraction circuit 7 is output as LL + LH + WL- (RL + r + WR) =
(LL-RL) + (LH-r) + (WL-WR).
Since the first term and the second term are speech signals, they have a correlation even if there
is a phase shift, and have the property as a synthesized signal.
However, the wind noise component is not correlated and does not have a property as a
composite signal because the generation method is a structural element and the vortex flow is a
main component.
Accordingly, (LL-RL) = combined signal LL '(LH-r) = combined signal LH'. Then, it is input to the
LPF 9 which is the first low-pass means, and is output as LL '+ (WL-WR).
[0013]
Similarly, the output from the LPF 10 which is the second low-pass means is RL '+ (WR-WL).
[0014]
The both outputs are opposite in positive and negative of the signal, are added by the adding
circuit 15 which is the first adding means, and the wind noise components WL and WR are
canceled and become LL ′ + RL ′, and then the first high pass The output LH 'from the HPF 11
as the means and the addition circuit 13 as the second addition means are added and finally
output as LH' + LL '+ RL'.
[0015]
Similarly, the output RH 'from the HPF 12 as the second high-pass means and the addition circuit
14 as the third adding means are added and finally output as RH' + LL '+ RL'.
[0016]
As described above, the output LH 'from the HPF 11 which is the first high-pass means and the
output RH' from the HPF 12 which is the second high-pass means are opposite in polarity of the
signal. Since both have the nature as a synthetic signal, cancellation is not performed.
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[0017]
In these final outputs, although the bass range is a mono signal (LL '+ RL') in which L and R are
combined, it is high without a specific bass range being attenuated as in the prior art. The sound
range is a stereo signal divided into right and left, and the component of wind noise is attenuated.
[0018]
Next, this embodiment will be described in more detail based on FIG.
[0019]
A microphone of channel 1 is connected to input terminal 1 and a microphone of channel 2 is
connected to input terminal 2.
Here, an audio signal L centered on the left is input to the input terminal 1, and an audio signal R
centered on the right is input to the input terminal 2.
The audio signal L is connected to the amplifier 3 and the delay circuit 5, and the audio signal R
is connected to the amplifier 4 and the delay circuit 6.
The output of the amplifier 3 and the output of the delay circuit 6 are connected to a subtraction
circuit 7 and subjected to subtraction processing.
The output of the amplifier 4 and the output of the delay circuit 5 are connected to a subtraction
circuit 8 and subjected to subtraction processing.
[0020]
At this time, it is ideal that only the signal on the left side is input to the input terminal 1 and the
signal on the right side is input to the input terminal 2; The signal is mixed and picked up.
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In particular, if a microphone with nondirectional characteristics is used, the difference between
the left and right signals will be reduced, and the stereo feeling will be lost.
In order to improve this, the phase difference between the signals picked up by the two
microphones is used, and the signals picked up each other are delayed and subtracted from the
signal of the other to attenuate the mixed signals and obtain separation. .
[0021]
Now, it is assumed that the wind noise component WL is included in the audio signal L at the
input terminal 1 and the wind noise component WR is included in the audio signal R at the input
terminal 2. As viewed from the input terminal 1, the input signal is amplified by the amplifier 3 at
L + WL, but since it does not change as a component, the output of the amplifier 3 is the same L
+ WL. This signal is input to the subtraction circuit 7 and the delay circuit 5 as it is. In the same
manner, the output of the amplifier 6 also becomes R + WR at the input terminal 2.
[0022]
Consider the case where LPFs are used for the delay circuits 5 and 6. When the output of the
delay circuit 5 expresses the input signal L + WL in more detail, L can be considered as being
divided into the low band component LL and the high band component LH. That is, LL + LH +
WL. More specifically, since the same signal of the opposite channel delayed in time always
arrives at each other's channel, it can be expressed as LL + LH + WL + dRL + dRH in terms of that
term. However, dRL and dRH mean time delay.
[0023]
Similarly, the input signal of the delay circuit 6 can be expressed as RL + RH + WR + dLL + dLH.
Then, since the delay circuit is an LPF, the output of the delay circuit 5 passes the very low
frequency component without delaying or attenuating, but the high frequency low frequency
component is slightly delayed. Let it be t, and express and think in front of the new composition.
The high frequency components are delayed and attenuated for passage. As a result, assuming
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that the delayed and attenuated high frequency component of LH is I, and the delayed and
attenuated high frequency component of RH is r, since the wind noise component WL is a low
frequency component, the output signal is t (LL + I + WL + dRL + dr) Become.
[0024]
Similarly, the output of the delay circuit 6 is t (RL + r + WR + dLL + dI).
[0025]
Therefore, the input signal of the subtraction circuit 7 is LL + LH + WL + dRL + dRH-t (RL + rWR
+ dL + dI) and the output signal a is (LL-tdLL) + (-tRL + dRL) + (WL-tWR) + (LH-tdI) + It becomes
(dRH-tr).
Since the first term and the second term, and the fourth term and the fifth term, which are voice
signals, are originally the same voice signal, they can be treated as a composite signal of signals
having a phase difference. However, the component of wind noise occurs in a structural element
in the way of generation, and since the vortex flow is the main component, it has no correlation
and can not be treated as a composite signal. Therefore, if (LL-tdLL) = LL (1-td) (LH-tdI) = LH '(tRL + dRL) = RL (d-t) (dRH-tr) = RH', then the output signal a is It becomes LL (1-td) + LH '+ RL
(dt) + RH' + (WL-tWR).
[0026]
Now, let's consider the case where there is no Rch signal to simplify the explanation here. That is,
when RL + RH = 0, the output signal a becomes LL (1-td) + LH '+ (WL-tWR).
[0027]
Similarly, the output signal b of the subtraction circuit 8 is RL + RL + WR + dLL + dLH-t (LL + I +
WL + dRL + dr) = (RL−tdRL) + (− tLL + dLL) + (WR−tWL) + (RH−tdr) + (dLH−tI) Become.
[0028]
Assuming that (RL-tdRL) = RL (1-td) (RH-tdr) = RH "(-tLL + dLL) = LL (dt) (tdLH-I) = LH", then RL
(1-td) + RH "+ LL Although (dt) + LH ′ ′ + (− tWL + WR), as described above, LH ′ ′
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component is set to be attenuated by the method for improving separation, so LH ′ ′ = 0 can
be set.
[0029]
Further, since RL + RH = 0, the output signal b is LL (d−t) + (tWL−WR).
[0030]
The output signal a is input to the LPF 9 and the HPF 11, and is similarly decomposed into the
high frequency component and the low frequency component. The output signal c of the LPF 9 is
the output signal e of LL + (1-td) + (WL-tWR) HPF11, It becomes LH '.
[0031]
Similarly, the output signal b is input to the LPF 10 and the HPF 12, and the output signal d of
the LPF 10 is zero, the output signal f of the LL (d−t) + (− tWL + WR) HPF 12.
Therefore, the output signal g of the adder circuit 15 is LL (1-td) + (WL-tWR) + LL (dt) + (-tWL +
WR) = LL ((1-td) + (dt)) + WL ( It becomes 1−t) + WR (−t + 1).
[0032]
Here, WL (1-t) indicates a subtraction of a signal and a signal which is the same as that of the
signal but is out of phase by t, and t is very small because it is a phase shift at the low frequency
of the LPF described above.
Therefore, the wind component WL on the left side is almost canceled and disappears.
Similarly, the right side component WR of the wind is almost canceled and disappears.
[0033]
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The meaning of the speech signal LL ((1-td) + (dt)) is the same as that of a certain signal and the
same signal but with the phase difference t and the phase difference due to the time difference d
and the time difference d It means the addition signal of the signal obtained by subtracting the
difference between the phase difference and the difference of t and the subtracted signal.
Therefore, the signal represented by 1-td has a phase difference larger than t, and the subtracted
signal is less likely to be canceled. Also, the signal represented by d−t is canceled and becomes
smaller because the phase difference becomes smaller. Therefore, LL ((1-td) + (d-t)), which means
the addition signal of the signal represented by 1-td and the signal represented by d-t, is not
canceled. It can be seen that the remaining component is the low frequency component of the
input audio signal.
[0034]
For example, since t = 0 in the ultra low band where the phase difference of the LPF disappears,
considering that case, WL (1-t) + WR (-t + 1) is completely subtracted because the same signals
are subtracted from each other .
[0035]
However, LL ((1-td) + (d-t)) disappears in the direction of 1-td, but in the case of d-t, the shift of
the phase difference due to the original time difference remains, so it is not completely canceled.
That is, as described in the ultra-low range, the sound always has less attenuation to the wind,
and as a result, the wind component is attenuated more.
[0036]
The final output signal h is LH '+ LL ((1-td) + (dt)), and the final output signal i is LL ((1-td) + (dt)).
[0037]
That is, in the low frequency range, L and R are combined into a mono signal.
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However, human beings do not have much ability to detect the direction to the bass range, the
directionality of the bass is detected with the high-pitched sound which is a harmonic component
of the bass, and there is almost no inconvenience even in the synthesized mono voice.
[0038]
The cutoff frequency of the first low pass filter LPF 9 and the second low pass filter LPF 10, and
the first high pass HPF 11 and the second high pass HPF 12 The cutoff frequency of H may be
completely the same or slightly different. If the two passbands are shifted in the overlapping
direction, it is possible to prevent further attenuation of the low frequency band of the target
voice, and if the two passbands are shifted in the direction away from each other, the wind noise
can be attenuated more completely.
[0039]
According to the present invention, in the final output, although the bass region is a mono signal
in which the left and right audio signals are synthesized, the particular bass region is attenuated
as in the prior art. The treble range is a stereo signal that is divided into left and right, and the
wind noise component is attenuated. That is, it is possible to attenuate only the offensive wind
noise component with little influence on the target low frequency range component of the voice.
[0040]
Brief description of the drawings
[0041]
1 is a block diagram of the audio processing circuit of the present invention.
[0042]
2 is a block diagram of a conventional audio processing circuit.
[0043]
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Explanation of sign
[0044]
DESCRIPTION OF SYMBOLS 1 1st input terminal 12 2nd input terminal 3 amplifier 4 amplifier 5
LPF6 LPF7 Subtractor circuit 8 Subtractor circuit 9 LPF10 LPF11 HPF12 HPF12 HPF13 addition
circuit 15 addition circuit 14 addition circuit
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