JP3593860B2 - Sound pickup device - Google Patents

Description

[0001]
TECHNICAL FIELD OF THE INVENTION
The present invention relates to a sound pickup device suitable for use in, for example, a digital or 8-mm type VTR integrated video camera, or a device having a portable recording function such as a portable MD or a portable DAT.
[0002]
[Prior art]
For example, the applicant of the present application has previously proposed Japanese Patent Application No. 2-132051 (refer to Japanese Patent Application Laid-Open No. 4-27298) as a sound pickup device that performs stereo sound pickup and the like suitable for use in a VTR integrated video camera. did. That is, in this device, spatial and electrical processing is performed using two microphones provided relatively close to each other so that good stereo sound pickup can be performed.
[0003]
In the case of using such a VTR-integrated video camera or the like in the field, for example, there arises a problem of a specific wind noise signal generated when the wind hits the microphone. In order to reduce such a wind noise signal, the applicant of the present application has previously proposed Japanese Patent Application No. 3-153341 (Japanese Patent Application Laid-Open No. 5-7392).
[0004]
In other words, in this device, in the signals picked up by the two microphones provided relatively close to each other, it is considered that the low-frequency sound signal has high correlation, while the wind noise signal has low correlation. It is considered. Therefore, the difference component between the signals picked up by the two microphones is extracted, the peak value of the difference component is detected, and particularly the low-frequency component of the picked up signal is attenuated according to the peak value. It is.
[0005]
Therefore, in this apparatus, a wind signal having low correlation is mainly extracted as the difference component, and the attenuation of the low-frequency component of the collected signal is controlled according to the peak value of the difference component. Only when the wind sound signal is large, the low frequency component of the signal collected is attenuated, and the wind sound signal is reduced.
[0006]
However, in this device, when the wind sound signal is large, all the low frequency components are attenuated, so that the sound signal to be collected is attenuated. Also, the fluctuation of the peak value of the wind noise signal changes sharply, and if the attenuation is controlled according to this peak value, the attenuation is frequently controlled, which may give an unnatural sense of hearing. There is.
[0007]
On the other hand, the applicant of the present application further adjusts and adjusts the control amount in order to eliminate the possibility that such an original sound signal to be picked up may be attenuated or the unnatural sound may be heard. Japanese Patent Application No. 6-59099 (refer to Japanese Patent Application Laid-Open No. 7-274280) has proposed a sound collecting device for processing.
[0008]
That is, in this apparatus, the extracted difference component is subtracted from the original signal, and the level of the subtracted difference component is controlled by a signal obtained by adding a predetermined attack and recovery characteristic to a signal obtained by converting the difference component into an absolute value. Thus, only the wind noise signal is reduced by minimizing the effect on the collected audio signal, and the wind noise signal is further reduced by performing a predetermined adaptive process.
[0009]
[Problems to be solved by the invention]
However, in the sound collecting device of the prior application, the configuration of an adaptive filter and the like for performing adaptive processing is complicated, and the scale of the device and the circuit is increased. Therefore, it has been difficult to apply the present invention to an apparatus that requires particularly small size and light weight, such as the above-described VTR integrated video camera.
[0010]
The present application has been made in view of such a point, and the problem to be solved is that in the conventional apparatus, the configuration of the filter and the like for performing the adaptive processing is complicated and the scale of the apparatus and the circuit is large. And it is difficult to apply the apparatus to a device that requires a particularly small size and light weight, such as a video camera integrated with a VTR.
[0011]
[Means for Solving the Problems]
Therefore, in the present invention, the difference component is subtracted from the collected signal, and the signal from which the difference component is subtracted is added, and the added signal is subtracted from the signal from which the difference component is subtracted. However, according to this, a favorable wind noise signal can be reduced with an extremely simple configuration.
[0012]
BEST MODE FOR CARRYING OUT THE INVENTION
That is, in the present invention, two or more microphones, a means for extracting a difference component between signals picked up by these microphones, and a first low-frequency component of the difference component are output to a signal picked up by each microphone. And a means for adding the signal from which the difference component has been subtracted and subtracting the added signal from the signal from which the difference component has been subtracted, respectively. The level is controlled by a signal obtained by adding a predetermined attack characteristic and a recovery characteristic to a signal obtained by converting the second low-frequency component of the difference component into an absolute value.
[0013]
【Example】
Hereinafter, the present invention will be described with reference to the drawings. FIG. 1 is a block diagram showing a configuration of an example of a sound collecting device to which the present invention is applied.
[0014]
In FIG. 1, microphones 1L and 1R indicate two microphones provided in close proximity to each other, such as a stereo microphone of a VTR-integrated video camera. Left (L) / right (R) signals collected by these microphones 1L, 1R are supplied to an A / D converter (ADC) 4 through amplifiers 2L, 2R and an automatic gain control (AGC) circuit 3, respectively. Is done. The left and right signals digitized by the A / D converter 4 are supplied to arithmetic units (Σ) 6L and 6R through delay (DL) circuits 5L and 5R, respectively.
[0015]
Further, the left and right signals from the A / D converter 4 are supplied to a computing unit (Σ) 7 where, for example, a right (R) signal is subtracted from a left (L) signal. The subtraction output of the arithmetic unit 7 is supplied to a first low-pass filter (LPF) 8 having a pass band of, for example, several kHz or less. As a result, the low-pass filter 8 extracts a low-frequency difference component corresponding to a wind sound signal of the signal collected by the two microphones 1L and 1R. Therefore, a low-frequency difference component corresponding to the wind sound signal is supplied to the arithmetic units 6L and 6R through the gain control amplifier 9, and is subtracted from the original left and right signals.
[0016]
Further, the subtraction output of the arithmetic unit 7 is supplied to a second low-pass filter (LPF) 10 having a pass band of, for example, several tens Hz or less. The output signal of the low-pass filter 10 is supplied to a detection circuit (DET) 12 through an amplifier 11. Further, the detection circuit 12 converts the absolute value of the signal, and supplies the absolute value of the signal to a circuit 13 which provides an attack characteristic and a recovery characteristic. A signal having these characteristics is supplied to the gain control amplifier 9, and the gain of the gain control amplifier 9 is controlled by the signal.
[0017]
Thereby, the level of the output signal of the low-pass filter 8 passed through the gain control amplifier 9 is controlled. The output signal of the gain control amplifier 9 is supplied to the arithmetic unit 6L, and is subtracted from the left (L) signal from the A / D converter 4 through the delay circuit 5L. The output signal of the gain control amplifier 9 is supplied to the arithmetic unit 6R and added to the right (R) signal from the A / D converter 4 through the delay circuit 5R.
[0018]
Therefore, in the above-described apparatus, the low-pass filter 8 outputs a low-frequency difference component of a signal collected by the two microphones 1L and 1R provided close to each other, that is, a low-frequency component having low correlation, that is, wind noise. The signal is extracted. Then, the wind noise signal is level-controlled by the gain control amplifier 9 and is subtracted from the original left and right signals from the A / D converter 4 by the calculators 6L and 6R. Since the arithmetic unit 7 subtracts the right signal of the A / D converter 4, the arithmetic unit 6R performs the subtraction by addition.
[0019]
Thus, these arithmetic units 6L and 6R subtract only the components of the wind sound signal from the signals collected by the microphones 1L and 1R, and remove the wind sound signal without attenuating the original sound signal to be collected. It can be performed. At this time, the component of the wind sound signal from the low-pass filter 10 is converted into an absolute value, and the level of the wind sound signal subtracted by the calculators 6L and 6R is controlled in accordance with the absolute value. Only when an error occurs, the wind noise signal is removed in accordance with the level.
[0020]
Further, the left and right signals from the above-described computing units 6L and 6R are supplied to computing units (Σ) 15L and 15R through delay (DL) circuits 14L and 14R, respectively. The left and right signals from the computing units 6L and 6R are supplied to a computing unit (Σ) 16 where the left and right signals are added. Then, the added output is supplied to a third low-pass filter (LPF) 17 having a predetermined pass band corresponding to a wind noise signal. Thus, the low-pass filter 17 extracts a signal corresponding to the in-phase component of the wind sound signal.
[0021]
A signal corresponding to the in-phase component of the wind noise signal is supplied to a gain control amplifier 18, and the gain of the gain control amplifier 18 is controlled by a signal from the circuit 13. Then, the signal from the gain control amplifier 18 is supplied to the calculators 15L and 15R, and is subtracted from the original left and right signals passed through the delay circuits 14L and 14R. The signals subtracted by the computing units 15L and 15R are output to output terminals 19L and 19R.
[0022]
As a result, in the arithmetic units 15L and 15R, the in-phase component of the wind sound signal in the signals collected by the microphones 1L and 1R is subtracted. At this time, the level of the wind noise signal subtracted by the computing units 15L and 15R is controlled in accordance with the absolute value of the wind noise signal, so that the wind noise is generated only when the wind noise signal is generated. The signal is removed, and the wind noise signal can be further reduced without attenuating the original audio signal to be picked up.
[0023]
Further, by adding a predetermined attack characteristic and a recovery characteristic to the control signal for controlling the level of the wind sound signal, it is possible to control the removal of the wind sound signal according to the fluctuation of the peak value of the wind sound signal. Is performed. That is, for example, at the beginning of a wind noise signal, an attack characteristic with a quick response is provided to improve the response to a gust or the like. Further, when the level of the wind noise signal is reduced, if the response of the control is made faster, the change point can be easily recognized by the auditory sense. Therefore, a recovery characteristic that makes the response slower is provided for the purpose of solving the problem.
[0024]
Therefore, in this apparatus, by subtracting the difference component from the collected signal, adding the signal from which the difference component has been subtracted, and subtracting the added signal from the signal from which the difference component has been subtracted, it is very simple. With the configuration, it is possible to effectively reduce a wind noise signal.
[0025]
In this case, in the conventional apparatus, for example, the configuration of filters and the like for performing adaptive processing is complicated, and the scale of the apparatus and the circuit becomes large. According to the present invention, these problems can be easily solved, which has been difficult to apply to a required device.
[0026]
Further, in this device, by providing the delay circuits 5L, 5R, 14L, and 14R in the signal path of the original audio signal, the phases of the original audio signal and the wind sound signal can be accurately matched, whereby the audio signal can be adjusted. Only the component of the wind signal inside can be accurately removed.
[0027]
In this apparatus, a first low-pass filter 8 for extracting a difference component of a wind noise signal and a second low-pass filter 10 for extracting a difference component forming a control signal are provided independently. Thereby, the characteristics of each pass band can be optimally matched.
[0028]
That is, for example, the cut-off frequency of the first low-pass filter 8 is set to several kHz including the band of the audio signal in order to pass the entire band of the wind sound signal, and the cut-off frequency of the second low-pass filter 10 is set to It is possible to set a frequency of several tens of Hz which is hardly affected by the audio signal. Further, characteristics such as the order of each filter can be optimized.
[0029]
Further, the characteristics and the like of the third low-pass filter 17 for extracting the in-phase component of the wind sound signal can be set independently, and the characteristics and the like can be optimized in removing the in-phase component of the wind sound signal. .
[0030]
In the above-described device, the delay circuits 5L, 5R, 14L, and 14R can be configured by a memory in the case of a digital circuit. In this case, the required memory capacity is only several kbits (bit length: 16 bits). For example, when this device is realized by a DSP (Digital Signal Processor), it can be constituted by a built-in RAM. In addition, even if it is realized by an ASIC, it is a capacity that can be easily incorporated.
[0031]
Thus, according to the above-described sound collection device, two or more microphones, a means for extracting a difference component between signals collected by these microphones, and a first low-frequency component of the difference component are collected by each microphone. Means for subtracting from the sounded signal, and means for adding the signal from which the difference component has been subtracted and subtracting the added signal from the signal from which the difference component has been subtracted, respectively, wherein the difference component to be subtracted and / or the subtracted signal is subtracted. The level of the added signal is controlled by a signal obtained by adding a predetermined attack characteristic and a recovery characteristic to a signal obtained by converting the second low-frequency component of the difference component into an absolute value, so that a good wind noise can be obtained with an extremely simple configuration. The signal can be reduced.
[0032]
【The invention's effect】
Therefore, according to the first aspect of the present invention, the difference component is subtracted from the collected signal, the signal from which the difference component is subtracted is added, and the added signal is subtracted from the signal from which the difference component is subtracted. Accordingly, it is possible to reduce a good wind noise signal with an extremely simple configuration.
[0033]
In this case, in the conventional device, for example, the configuration of filters and the like for performing the adaptive processing is complicated, and the scale of the device and the circuit becomes large. According to the present invention, these problems can be easily solved, which has been difficult to apply to a required device.
[0034]
Further, in the present invention, by providing a delay circuit in the signal path of the original sound signal, the phase of the original sound signal and the wind sound signal can be accurately matched, thereby making it possible to adjust the phase of the wind sound signal in the sound signal. Only the components can be accurately removed.
[0035]
Further, in the present invention, the first low-pass filter for extracting the difference component of the wind sound signal and the second low-pass filter for extracting the difference component forming the control signal are independently provided, The characteristics of each pass band can be optimally matched.
[Brief description of the drawings]
FIG. 1 is a configuration diagram of an example of a sound collection device to which the present invention is applied.
[Explanation of symbols]
1L, 1R microphone, 2L, 2R, 11 amplifier, 3 automatic gain control (AGC) circuit, 4 A / D converter (ADC), 5L, 5R, 14L, 14R delay (DL) circuit, 6L , 6R, 7, 15L, 15R, 16 ... arithmetic unit (演算), 8, 10, 17 ... low-pass filter (LPF), 9, 18 ... gain control amplifier, 12 ... detection circuit (DET), 13 ... attack characteristics and Circuits for providing recovery characteristics, 19L, 19R ... output terminals

Claims (3)

Two or more microphones,
Means for extracting the difference components of the signals picked up by these microphones,
Means for subtracting the first low frequency component of the difference component from the signals picked up by the respective microphones;
Means for adding the signal from which the difference component has been subtracted and subtracting the added signal from the signal from which the difference component has been subtracted,
Controlling the level of the subtracted difference component and / or the added signal to be subtracted by a signal obtained by adding a predetermined attack characteristic and a recovery characteristic to a signal obtained by converting a second low-frequency component of the difference component into an absolute value; A sound pickup device characterized by the above-mentioned. A first low-frequency component forming the difference component subtracted from the signal picked up by the microphone;
2. The sound pickup device according to claim 1, wherein the second low-frequency component forming the difference component to be detected by the envelope detection is extracted with appropriate characteristics. 2. A method according to claim 1, wherein a predetermined delay is added to the signal collected by the microphone for subtracting the difference component and / or the signal from which the difference component is subtracted for subtracting the added signal. The sound pickup device as described.

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