KR101600195B1 - Beamforming System and Method Using Highly Directive Beamformer - Google Patents

Abstract

The present invention relates to a beam forming system. According to an embodiment of the present invention, the beam forming system comprises: a control part; multiple microphones which are disposed at a regular space in a row to acquire multiple sound signals; a filter part which filters the multiple sound signals acquired by the multiple microphones using a filter formula; and an outputting part which merges and outputs the filtered multiple sound signals. The control part controls the magnitude of the sound signal in a preset direction among the multiple sound signals using a first element in the filter formula of the filter part. The sound signal in the preset direction is at least one of the multiple sound signals.

Description

BACKGROUND OF THE INVENTION 1. Field of the Invention [0001] The present invention relates to a beam forming system and method using a supergain beamformer,

BACKGROUND OF THE INVENTION 1. Field of the Invention [0001] The present invention relates to a beam forming system, and more particularly, to a system for controlling a size of a sound signal in a specific direction in a sound signal obtained without changing a hardware configuration of a microphone array by using a supergain beam- .

When sounds from a plurality of sound sources are simultaneously recorded on one recording channel, it is very difficult to separate each sound signal from a plurality of sound sources.

Particularly, when acquiring voice from various digital devices such as a cellular phone and a digital camcorder, or a voice recognition device in a vehicle, there is a problem in that not only a human voice but also noise are inputted into a microphone and thus a clear sound can not be obtained.

In order to solve this problem, a directional microphone more sensitive to a sound generated in a specific direction may be used. However, if the point of sound generation is changed, the directional microphone can not respond sensitively to the sound of the changed direction.

Another solution is to develop a beamformer technique that uses a microphone array to form a beam in a particular direction.

The beamformer detects the direction of the sound by using the time difference of the signal arriving at each microphone, and enhances only the voice signal located in a certain direction, or conversely, eliminates unnecessary interference noise. At this time, And the placement and distance of each microphone must be known in advance

By using this beamformer technology, it is possible to improve the performance such as sound separation or speaker localization that excludes or separates other noise sources around the speaker, and it is possible to improve the performance by post-filtering. It may also reduce non-existent noise or reverberation.

That is, by acquiring speech signals existing at a long distance using a microphone array, it is possible to emphasize or suppress the speech signal inputted from a specific direction, and to eliminate sounds other than the specific direction.

However, the general beamformer determines the response to the space (directivity) by the beamforming algorithm and the hardware configuration of the microphone array (arrangement of the microphone). Therefore, it is difficult to change the directivity in the actual situation because the hardware configuration must be changed in order to change the directivity characteristic.

Therefore, it is required to develop a system that can control the relative size of sounds generated from different sound sources without changing the hardware configuration by applying a beam former that can control the response to the space with respect to sounds collected from the multi-channel microphones It is true.

It is an object of the present invention to provide a user with a beam forming system capable of adjusting the relative sizes of different sound sources recorded at the same time.

More specifically, it is an object of the present invention to provide a beamforming system capable of adjusting the size of a sound signal in a specific direction without changing the hardware configuration of the microphone array.

Another object of the present invention is to provide a user with a beam forming system capable of processing a sound signal obtained from a microphone in real time as well as post-processing a stored sound signal.

It is to be understood that both the foregoing general description and the following detailed description of the present invention are exemplary and explanatory and are not intended to limit the invention to the precise form disclosed. It can be understood.

According to an aspect of the present invention, there is provided a beam forming system including: a control unit; A plurality of microphones arranged in a line at even intervals to acquire a plurality of sound signals; A filter unit for filtering a plurality of sound signals obtained by the plurality of microphones under the control of the control unit using a filter formula; And an output unit for outputting a combined plurality of the filtered sound signals, wherein the control unit controls the first filter unit of the filter unit such that a sound signal size of a predetermined direction among the plurality of sound signals, And the sound signal in the predetermined direction may be at least one of the plurality of sound signals.

Further, the filter equation is expressed by the following equation,

Equation

는

이고, 상기

는 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 지향성, 상기

는 상기 소리신호의 주파수이고, 상기

은 상기 지향성을 나타내는 요소 중 방위각을 나타내며, 상기

는 상기 지향성을 나타내는 요소 중 천정각을 나타내고, 상기

는 소리신호의 속도이며, 상기

은 복수의 마이크로폰을 연결한 직선인 제 1 선의 중심을 원점으로 하여 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 위치를 나타내고, 상기

는 상기 제 1 선과 상기 제 1 선의 중심점으로부터 상기 복수의 소리신호의 음원을 연결한 제 2 선이 이루는 각도이며,(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,

는 상기

의 에르미트 행렬(Hermitian matrix)이며,

는

이고, 상기

는 안정화 계수이며, 상기

는 단위행렬이다.)remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)

일 수 있다.The first element may be any of the

Lt; / RTI >

In addition, the plurality of microphones may be a directional microphone.

In addition, the plurality of microphones may be a non-directional microphone.

The filter unit may perform filtering by performing an inverse Fourier transform of the equation and a plurality of sound signals obtained by the plurality of microphones to perform a convolution operation.

The filter unit Fourier-transforms a plurality of sound signals obtained by the plurality of microphones, multiplies the Fourier-transformed sound signal by the equation, and performs inverse Fourier transform on the sound signal multiplied by the equation .

The Fourier transform and the inverse Fourier transform may be Discrete Fourier Transform and Inverse Discrete Fourier Transform.

According to an aspect of the present invention, there is provided a beamforming system including: a first step of acquiring a plurality of sound signals using a plurality of microphones arrayed at equal intervals; A plurality of sound signals obtained by the plurality of microphones are filtered using a filter expression, and filtering is performed so that a size of a sound signal in a predetermined direction of the plurality of sound signals is adjusted using the first element of the filter expression Step 2; And a third step of merging and outputting the filtered plurality of sound signals, wherein the sound signal in the predetermined direction may be at least one of the plurality of sound signals.

Further, the filter equation is expressed by the following equation,

Equation

는

이고, 상기

는 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 지향성 , 상기

는 상기 소리신호의 주파수이고, 상기

은 상기 지향성을 나타내는 요소 중 방위각을 나타내며, 상기

는 상기 지향성을 나타내는 요소 중 천정각을 나타내고, 상기

는 소리신호의 속도이며, 상기

은 복수의 마이크로폰을 연결한 직선인 제 1 선의 중심을 원점으로 하여 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 위치를 나타내고, 상기

는 상기 제 1 선과 상기 제 1 선의 중심점으로부터 상기 복수의 소리신호의 음원을 연결한 제 2 선이 이루는 각도이며, 상기

는 상기

의 에르미트 행렬(Hermitian matrix)이며,

는

이고, 상기

는 안정화 계수이며, 상기 는 단위행렬이다.)(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient, Is a unitary matrix.)

일 수 있다.The first element may be any of the

Lt; / RTI >

In addition, the plurality of microphones may be a directional microphone.

In addition, the plurality of microphones may be a non-directional microphone.

In the second step, the inverse Fourier transformed expression of the above equation and a plurality of sound signals obtained by the plurality of microphones may be filtered by convolution operation.

In the second step, the plurality of sound signals obtained by the plurality of microphones are Fourier-transformed, the Fourier-transformed sound signal is multiplied by the mathematical expression, and the sound signal multiplied by the mathematical expression is subjected to an inverse Fourier transform .

The Fourier transform and the inverse Fourier transform may be Discrete Fourier Transform and Inverse Discrete Fourier Transform.

Meanwhile, a post-processing beam forming system related to an example of the present invention for realizing the above-mentioned problems includes a control unit; A file loading unit for loading a plurality of files storing a plurality of sound signals obtained by a plurality of microphones arrayed at equal intervals; A filter unit for filtering a plurality of sound signals stored in the plurality of files under the control of the controller using a filter formula; And an output unit for outputting a combined plurality of the filtered sound signals, wherein the control unit controls the first filter unit of the filter unit such that a sound signal size of a predetermined direction among the plurality of sound signals, And the sound signal in the predetermined direction may be at least one of the plurality of sound signals.

Further, the filter equation is expressed by the following equation,

Equation

는

이고, 상기

는 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 지향성 , 상기

는 상기 소리신호의 주파수이고, 상기

은 상기 지향성을 나타내는 요소 중 방위각을 나타내며, 상기

는 상기 지향성을 나타내는 요소 중 천정각을 나타내고, 상기

는 소리신호의 속도이며, 상기

은 복수의 마이크로폰을 연결한 직선인 제 1 선의 중심을 원점으로 하여 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 위치를 나타내고, 상기

는 상기 제 1 선과 상기 제 1 선의 중심점으로부터 상기 복수의 소리신호의 음원을 연결한 제 2 선이 이루는 각도이며, 상기 는 상기

의 에르미트 행렬(Hermitian matrix)이며,

는

이고, 상기

는 안정화 계수이며, 상기

는 단위행렬이다.)(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line, Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)

일 수 있다.The first element may be any of the

Lt; / RTI >

A beamforming method according to an embodiment of the present invention for realizing the above-mentioned problems includes a first step of loading a plurality of files in which a plurality of sound signals obtained by each of a plurality of microphones arrayed at equal intervals are stored; A second step of filtering a plurality of sound signals stored in the plurality of files by using a filter formula so that a size of a sound signal in a predetermined direction among the plurality of sound signals is adjusted using a first element of the filter expression, ; And a third step of merging and outputting the filtered plurality of sound signals, wherein the sound signal in the predetermined direction may be at least one of the plurality of sound signals.

Further, the filter equation is expressed by the following equation,

Equation

는

이고, 상기

는 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 지향성 , 상기

는 상기 소리신호의 주파수이고, 상기

은 상기 지향성을 나타내는 요소 중 방위각을 나타내며, 상기

는 상기 지향성을 나타내는 요소 중 천정각을 나타내고, 상기

는 소리신호의 속도이며, 상기

은 복수의 마이크로폰을 연결한 직선인 제 1 선의 중심을 원점으로 하여 상기 복수의 마이크로폰 중 N번째 위치하는 마이크로폰의 위치를 나타내고, 상기

는 상기 제 1 선과 상기 제 1 선의 중심점으로부터 상기 복수의 소리신호의 음원을 연결한 제 2 선이 이루는 각도이며,(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,

는 상기

의 에르미트 행렬(Hermitian matrix)이며,

는

이고, 상기

는 안정화 계수이며, 상기

는 단위행렬이다.)remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)

일 수 있다.The first element may be any of the

Lt; / RTI >

A recording medium related to an example of the present invention for realizing the above-mentioned problem is a program for tangibly embodying a program of instructions executable by a digital processing apparatus to perform a beam forming method, The method comprising: a first step of acquiring a plurality of sound signals using a plurality of microphones arrayed at equal intervals in a line; A plurality of sound signals obtained by the plurality of microphones are filtered using a filter expression, and filtering is performed so that a size of a sound signal in a predetermined direction of the plurality of sound signals is adjusted using the first element of the filter expression Step 2; And a third step of merging and outputting the filtered plurality of sound signals, wherein the sound signal in the predetermined direction may be at least one of the plurality of sound signals.

Also disclosed is a recording medium on which a program of instructions executable by a digital processing apparatus to perform a post processing beamforming method is tangibly embodied and which can be read by the digital processing apparatus, A first step of loading a plurality of files storing a plurality of sound signals obtained by each of a plurality of microphones arrayed at regular intervals; A second step of filtering a plurality of sound signals stored in the plurality of files by using a filter formula so that a size of a sound signal in a predetermined direction among the plurality of sound signals is adjusted using a first element of the filter expression, ; And a third step of merging and outputting the filtered plurality of sound signals, wherein the sound signal in the predetermined direction may be at least one of the plurality of sound signals.

The present invention can provide a user with a beam forming system capable of adjusting the relative sizes of different sound sources simultaneously recorded.

Specifically, it is possible to provide a user with a beam forming system capable of adjusting the size of a sound signal in a specific direction without changing the hardware configuration of the microphone array.

In addition, it is possible to provide a user with a beam forming system capable of processing a sound signal obtained from a microphone in real time as well as post-processing a stored sound signal.

It should be understood, however, that the effects obtained by the present invention are not limited to the above-mentioned effects, and other effects not mentioned may be clearly understood by those skilled in the art to which the present invention belongs It will be possible.

BRIEF DESCRIPTION OF THE DRAWINGS The accompanying drawings, which are incorporated in and constitute a part of the specification, illustrate a preferred embodiment of the invention and, together with the description, serve to provide a further understanding of the technical idea of the invention, It should not be construed as limited.
1 is a block diagram of a beamforming system in accordance with one embodiment of the present invention.
Figure 2 shows an array of microphones according to one embodiment of the present invention.
FIG. 3 is a graph illustrating a gain change of a filter equation according to a stabilization coefficient according to an embodiment of the present invention. Referring to FIG.
4 is a flowchart illustrating a method of controlling a magnitude of a sound signal according to an exemplary embodiment of the present invention.
FIG. 5 is a diagram showing a location where microphone arrays and sound signals are generated according to the first embodiment of the present invention. FIG.
FIG. 6 is a graph illustrating frequency of output of a merged sound signal according to the first embodiment of the present invention. Referring to FIG.
FIG. 7 is a view showing a location where microphone arrays and sound signals are generated according to the second embodiment of the present invention.
8 is a graph showing the output of the merged sound signal with respect to frequency according to the second embodiment of the present invention

A beamforming technique has been developed that finds the direction of the sound, enhances only the speech signal located in a specific direction, or conversely uses the time difference of the signal reaching each microphone to eliminate unnecessary interference noise.

However, in the conventional beamforming technique, it is very difficult to change the directivity in the actual situation because the hardware configuration such as the microphone must be changed to control the directivity characteristic. Therefore, it is required to develop a beam forming technique that can change the directivity without changing the hardware configuration.

In this specification, a volume control system for improving the disadvantages of the conventional beam forming technique using a high directive beamformer is proposed.

Hereinafter, a preferred embodiment of the present invention will be described with reference to the drawings. In addition, the embodiment described below does not unduly limit the contents of the present invention described in the claims, and the entire configuration described in this embodiment is not necessarily essential as the solution means of the present invention.

1 is a block diagram of a beamforming system in accordance with one embodiment of the present invention.

However, the components shown in FIG. 1 are not essential, and a beamforming system having components with fewer or fewer components may be implemented.

1, the beam forming system may include a plurality of microphones 100, a filter unit 200, a controller 300, and an output unit 400.

First, a plurality of microphones 100 are devices for acquiring sound signals.

The microphone 100 may have a directivity that means a degree of better sensitivity to the sound generated in a specific direction. The width of the directivity pattern in which the directivity appears is referred to as a directivity width. On the contrary, the width of the portion where the directivity is suppressed and the sensitivity is extremely low on the directivity pattern is referred to as a null width.

로 나타낼 수 있는데 상기

는 마이크로폰(100)이 획득하는 소리신호의 주파수이고, 상기

은 구좌표계에서 마이크로폰(100)을 기준으로 한 방위각을 나타내는 인자이며, 상기

는 천정각을 나타내는 인자이다.On the other hand, an element indicating the directivity of the microphone 100

.

Is the frequency of the sound signal acquired by the microphone 100,

Is a factor indicating an azimuth angle with respect to the microphone 100 in the spherical coordinate system,

Is a factor indicating the zenith angle.

값이 상수가 된다.In the present invention, a microphone 100 having directivity may be used, or a non-directional microphone 100 may be used, which acquires the same sound regardless of the direction of sound generation. In the case of the omnidirectional microphone 100

The value is a constant.

Meanwhile, in the present invention, the plurality of microphones 100 should be arranged in a line at regular intervals.

Figure 2 shows an array of microphones according to one embodiment of the present invention.

는 좌표계를 구좌표계로 나타냈을 때 소리신호 방향의 방위각이고

는 소리신호의 방향의 천정각이다.Referring to FIG. 2, the microphones 100 may be arranged in a line at even intervals on the z-axis. 2, the origin corresponds to a center point in the arrangement of a plurality of microphones 100,

Is the azimuth of the sound signal direction when the coordinate system is represented by the spherical coordinate system

Is the zenith angle of the direction of the sound signal.

는 마이크로폰(100) 어레이에 영향을 주지 않는다.
However, in the present invention, the microphones 100 are arranged in a line at regular intervals

Does not affect the microphone 100 array.

Next, the filter unit 200 is configured to filter the sound signals obtained from the plurality of microphones 100 according to the control of the control unit using the supergain beamformer equation.

Hereinafter, the super directional beam former equation will be described.

Spherical Beamformer Type

(In the supergiant beam former equation

는 각 마이크로폰(100)의 상대적 위치 차이에 의해서 발생하는 전달 지연벡터로서

이고, 상기

는 복수의 마이크로폰(100) 중 N번째 위치하는 마이크로폰(100)의 지향성을 나타내는 요소이며, 상기

는 소리신호의 주파수이고, 상기

은 상기 지향성을 나타내는 요소 중 방위각을 나타내는 인자이며, 상기

는 상기 지향성을 나타내는 요소 중 천정각을 나타내는 인자이고, 상기

는 소리신호의 속도이며, 상기

은 상기 복수의 마이크로폰을 연결한 직선인 제 1 선의 중심을 원점으로 하여 상기 복수의 마이크로폰(100) 중 N번째 위치하는 마이크로폰(100)의 위치를 나타내고, 상기

는 상기 제 1 선과 상기 제 1 선의 중심점으로부터 상기 복수의 소리신호의 음원을 연결한 제 2 선이 이루는 각도이며, 상기

는 상기

의 에르미트 행렬(Hermitian matrix)이며,

Is a propagation delay vector caused by a relative positional difference between the microphones 100

, And

Is an element representing the directivity of the microphone 100 positioned at the N-th position among the plurality of microphones 100,

Is the frequency of the sound signal,

Is a factor indicating an azimuth angle among elements indicating the directivity,

Is a factor indicating a zenith angle of the element indicating the directivity,

Is the speed of the sound signal,

Represents a position of a microphone (100) positioned at an Nth position among the plurality of microphones (100) with the center of a first line, which is a straight line connecting the plurality of microphones, as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,

Quot;

Hermitian matrix of < RTI ID = 0.0 >

는 각 마이크로폰(100) 위치 좌표에 대한 상관 행렬로서

이고, 상기

는 안정화 계수이며, 상기

는 단위행렬이다.)

As a correlation matrix for the position coordinates of each microphone 100

, And

Is a stabilization coefficient,

Is a unitary matrix.)

는

로 나타낼 수 있으며(T 연산자는 행렬을 전치하는 연산자),

의 각 요소인

는 N번째 마이크로폰(100)가 획득한 소리신호를 필터링하는 요소이다.Spherical Beamformer Type

The

(The T operator is an operator that transposes a matrix), and

Each element of

Is an element for filtering the sound signal acquired by the Nth microphone 100. [

와 곱한 후, 이를 역푸리에변환함으로써 필터링할 수 있다. 또한, 푸리에변환 및 역푸리에변환을 할 때에는 보다 빠른 푸리에변환을 위해 이산푸리에변환법을 사용할 수 있다.On the other hand, when filtering a plurality of sound signals obtained by the microphone 100, a sound signal is subjected to Fourier transform (Fourier Transform)

And then it can be filtered by inverse Fourier transform. Further, when conducting Fourier transform and inverse Fourier transform, it is possible to use a discrete Fourier transform method for faster Fourier transform.

를 역 푸리에 변환하여 마이크로폰(100)이 획득한 소리신호와 컨볼루션 연산함으로써 이득을 적용할 수도 있다.
In addition, the super-directional beam-

And the gain can be applied by performing a convolution operation with the sound signal acquired by the microphone 100. [

값을 이용하여 복수의 소리 신호 중 기 설정된 방향(원하는 방향)의 소리신호 크기가 조절되도록 필터부를 제어하는 구성이다. 즉,

값을 조절하여 특정방향의 소리신호의 크기를 제어하는 것이다. 크기가 조절되는 소리신호는 복수개가 될 수 있으며 획득한 소리신호 중 적어도 하나이다.Next, the control unit 300 determines whether or not the < RTI ID = 0.0 >

Values are used to control the filter unit so that the size of a sound signal in a predetermined direction (a desired direction) among a plurality of sound signals is adjusted. In other words,

To control the magnitude of the sound signal in a particular direction. The size-adjusted sound signal may be a plurality of sound signals and is at least one of the acquired sound signals.

Next, the output unit 400 combines a plurality of sound signals filtered by the filter unit 200 and outputs the combined signals.

값에 따라 변하는 필터링 특성에 대해 설명한다.Hereinafter, with reference to FIG. 3

The filtering characteristic that varies depending on the value will be described.

FIG. 3 is a graph illustrating a gain change of a filter equation according to a stabilization coefficient according to an embodiment of the present invention. Referring to FIG.

값에 따라 도 3과 같은 특성을 갖게 되어 소리신호에 이득을 적용한다.Referring to FIG. 3, the gain applied to the sound signal at the time of filtering changes according to the direction of the sound signal. The filter unit 200 is connected to the control unit 300,

And the gain is applied to the sound signal.

이 0.1 또는 0.001인 경우 40° 방향에서 발생한 소리신호는 제거되지 않는다. 그러나

이 0.00001인 경우 이득이 -25dB가 되어 제거될 수 있다.E.g,

Is 0.1 or 0.001, the sound signal generated in the direction of 40 DEG is not removed. But

Is 0.00001, the gain becomes -25dB and can be removed.

값이 작을수록

(소리신호의 방향)변화에 따른 이득의 변화는 커진다. 즉,

값이 작을수록 빔포밍 시스템의 지향성이 커지는 것이다.On the other hand,

The smaller the value,

(Direction of the sound signal) changes. In other words,

The smaller the value, the greater the directivity of the beamforming system.

값을 조절하여 소리신호의 방향에 따른 이득(gain)의 크기를 변경하여 특정 방향의 소리 신호의 크기를 증가 또는 감소시킬 수 있는 것이다.
like this

The magnitude of the sound signal in the specific direction can be increased or decreased by changing the magnitude of the gain according to the direction of the sound signal.

Hereinafter, with reference to FIG. 4, a method for controlling the size of a sound signal along a direction will be described in detail with reference to the above-described configurations.

4 is a flowchart illustrating a method of controlling a magnitude of a sound signal according to an exemplary embodiment of the present invention.

First, a plurality of sound signals are acquired using a plurality of microphones 100 arrayed at regular intervals (S100).

The plurality of microphones 100 may simultaneously acquire sound signals generated from a plurality of different sound sources, and the frequencies of the sound signals may be different.

의

값을 결정한다(S200).Next, the supergain beamformer equation

of

(S200).

값을 조절하여 방향

에 따른 이득(gain)을 제어할 수 있다.As described above

Adjust the value to

Can be controlled.

값이 적용된 초지향성 빔포머식을 이용하여 필터링한다(S300).Next, the obtained plurality of sound signals are determined

(S300) using the supergain beamformer equation.

값이 적용된 초지향성 빔포머식을 이용하여 복수의 소리신호 중 기 설정된 방향의 소리신호의 크기가 조절되도록 필터링하는 것이다.In other words,

And the size of the sound signal of the predetermined direction among the plurality of sound signals is adjusted by using the supergain beamformer method to which the value is applied.

와 곱한 후 역푸리에변환함으로써 필터링할 수 있고, 획득한 소리신호를

의 역푸리에 변환과 컨볼루션 연산함으로써 필터링할 수 있다.The sound signal obtained as described above is subjected to Fourier transform

And then performing inverse Fourier transform to filter the obtained sound signal,

By performing an inverse Fourier transform and a convolution operation.

Next, the plurality of filtered sound signals are merged and output (S400).

값에 따라 상이하다.
The sound signal output is

It depends on the value.

≪ Embodiment 1 >

FIG. 5 is a diagram showing a location where microphone arrays and sound signals are generated according to the first embodiment of the present invention. FIG.

가 0°인 방향에 680Hz의 소리를 발생시킨다. 또한, 마이크로폰(100) 배열의 중심을 기준으로 복수의 마이크로폰(100)이 이루는 선으로부터 45°방향, 즉

가 45°인 방향에 880Hz의 소리를 발생시킨다.Referring to FIG. 5, a plurality of microphones 100 are arranged at equal intervals in a line, and a plurality of microphones 100 are arranged in a linear direction

The sound of 680 Hz is generated in the direction of 0 °. Further, in the direction of 45 degrees from the line of the plurality of microphones 100 with respect to the center of the arrangement of the microphones 100, that is,

The sound of 880 Hz is generated in the direction of 45 °.

First, a plurality of microphones 100 acquire sound signals generated at two points.

값을 조절하여 기 설정된 방향의 소리신호 크기가 조절되도록 제어한다.Next, the filter unit 200 filters the converted plurality of sound signals using a sidelobe beamformer, and the controller 300 controls the sidelobe beamformer

And controls the size of the sound signal in the predetermined direction to be adjusted.

Next, the output unit 400 merges and outputs the plurality of filtered sound signals.

FIG. 6 is a graph illustrating frequency of output of a merged sound signal according to the first embodiment of the present invention. Referring to FIG.

값을 0.1, 0.001 및 0.00001 으로 한 초지향성 빔포머식을 적용하여 필터링하면 안정화 계수가 작아질수록 680Hz의 출력은 변화가 없으나 880Hz의 출력은 줄어든다.Referring to FIG. 6, the unfiltered sound signal shows the maximum output at 680 Hz and 880 Hz, respectively, of the generated sound frequency. But

When the value is set to 0.1, 0.001, and 0.00001, the output is not changed at 680 Hz, but the output at 880 Hz is reduced as the stabilization coefficient is decreased.

값을 조절함으로써 복수의 방향에서 발생한 소리 중 특정 방향에서 발생한 소리신호의 크기가 조절되도록 제어할 수 있는 것이다.
That is, a plurality of sounds generated in different directions are simultaneously acquired

It is possible to control the magnitude of the sound signal generated in a specific direction among the sounds generated in a plurality of directions.

≪ Embodiment 2 >

FIG. 7 is a view showing a microphone array according to a second embodiment of the present invention and a sound generation position. FIG.

가 0°인 방향에서 1070Hz의 소리를 발생시키고

가 45°인 방향에서 1270Hz의 소리를 발생시킨 것이다.Referring to FIG. 7, in the second embodiment,

A sound of 1070 Hz is generated in the direction of 0 DEG

The sound of 1270 Hz is generated in the direction of 45 °.

FIG. 8 is a graph illustrating frequency of output of a merged sound signal according to a second embodiment of the present invention. Referring to FIG.

값을 0.1, 0.001 및 0.00001 으로 한 초지향성 빔포머식을 적용하여 필터링하면 안정화 계수가 작아질수록 1070Hz의 출력은 변화가 없으나 1270Hz의 출력은 줄어든다.
Referring to FIG. 8, unfiltered sound signals have maximum outputs at frequencies of 1070 Hz and 1270 Hz, respectively, as in the first embodiment. But

When the value is set to 0.1, 0.001 and 0.00001, the output of 1070Hz is not changed but the output of 1270Hz is reduced as the stabilization coefficient is decreased.

≪ Third Embodiment >

The present invention can be implemented in a state in which a plurality of microphones 100, a filter unit 200, a control unit 300 and an output unit 400 are provided, Lt; RTI ID = 0.0 > stored < / RTI > from the recording medium.

In this case, it is not necessary to separately provide the microphone 100 in the present invention, and a file loading unit 110 for loading a file storing sound signals obtained by a plurality of microphones arranged at equal intervals in a row, a filter unit 200, a control unit 300, and an output unit 400 to adjust the size of a sound signal in a specific direction.

However, in the case where sounds are acquired by the array of n microphones (100), n stored files are required. Also in this case, it is assumed that sound is acquired in a state in which n microphones (100) .

The present invention can also be embodied as computer-readable codes on a computer-readable recording medium. A computer-readable recording medium includes all kinds of recording apparatuses in which data that can be read by a computer system is stored. Examples of the computer-readable recording medium include a ROM, a RAM, a CD-ROM, a magnetic tape, a floppy disk, an optical data storage device, and the like, and may be implemented in the form of a carrier wave (for example, transmission over the Internet) . In addition, the computer-readable recording medium may be distributed over network-connected computer systems so that computer readable codes can be stored and executed in a distributed manner. In addition, functional programs, codes, and code segments for implementing the present invention can be easily inferred by programmers of the technical field to which the present invention belongs.

The above-described volume control system is not limited to the configuration and method of the above-described embodiments, but the embodiments may be modified so that all or some of the embodiments may be selectively combined .

According to the present invention applying the above-described configuration, the directivity can be changed without changing the hardware configuration of the microphone array by performing the beam forming using the supergain beamformer, and when the sound signals input to the microphone array are individually stored, The size of the sound signal in a specific direction can be adjusted.

Claims (20)

A control unit;
A plurality of microphones arranged in a line at even intervals to acquire a plurality of sound signals;
A filter unit for filtering a plurality of sound signals obtained by the plurality of microphones under the control of the control unit using a filter formula; And
And an output unit for merging and outputting the plurality of filtered sound signals,
Wherein the control unit adjusts a value of a first element of the filter type of the filter unit to control a size of a sound signal in a predetermined direction among the plurality of sound signals,
Wherein the sound signal in the predetermined direction is at least one of the plurality of sound signals,
The filter equation is expressed by the following equation,
Equation

(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,
remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)
The first element may be any of the

The beam forming system comprising: delete The method according to claim 1,
Wherein the plurality of microphones are directional microphones. The method according to claim 1,
Wherein the plurality of microphones are non-directional microphones. The method according to claim 1,
Wherein the filter unit performs filtering by performing an inverse Fourier transform of the equation and a plurality of sound signals obtained by the plurality of microphones by performing a convolution operation. The method according to claim 1,
The filter unit Fourier-transforms a plurality of sound signals obtained by the plurality of microphones, multiplies the Fourier-transformed sound signal by the equation, and performs filtering by inverse Fourier transforming the sound signal multiplied by the equation Beam forming system. The method according to claim 6,
Wherein the Fourier transform and the inverse Fourier transform are a Discrete Fourier Transform and an Inverse Discrete Fourier Transform. A first step of acquiring a plurality of sound signals using a plurality of microphones arrayed at regular intervals;
Wherein the plurality of sound signals obtained by the plurality of microphones are filtered using a filter expression, and the value of the first element of the filter expression is adjusted to filter the sound signals of the predetermined direction among the plurality of sound signals, ; And
And a third step of merging and outputting the plurality of filtered sound signals,
Wherein the sound signal in the predetermined direction is at least one of the plurality of sound signals,
The filter equation is expressed by the following equation,
Equation

(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,
remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)
The first element may be any of the

The beam forming method comprising the steps of: delete 9. The method of claim 8,
Wherein the plurality of microphones are directional microphones. 9. The method of claim 8,
Wherein the plurality of microphones are non-directional microphones. 9. The method of claim 8,
The second step comprises:
Wherein the inverse Fourier transformed expression of the equation and the plurality of sound signals obtained by the plurality of microphones are convoluted by filtering. 9. The method of claim 8,
The second step comprises:
Characterized in that a plurality of sound signals obtained by the plurality of microphones are Fourier-transformed, the Fourier-transformed sound signal is multiplied by the equation, and the sound signal multiplied by the equation is filtered by inverse Fourier transform Foaming method. 14. The method of claim 13,
Wherein the Fourier transform and the inverse Fourier transform are a Discrete Fourier Transform and an Inverse Discrete Fourier Transform. A control unit;
A file loading unit for loading a plurality of files storing a plurality of sound signals obtained by a plurality of microphones arrayed at equal intervals;
A filter unit for filtering a plurality of sound signals stored in the plurality of files under the control of the controller using a filter formula; And
And an output unit for merging and outputting the plurality of filtered sound signals,
Wherein the control unit adjusts a value of a first element of the filter type of the filter unit to control a size of a sound signal in a predetermined direction among the plurality of sound signals,
Wherein the sound signal in the predetermined direction is at least one of the plurality of sound signals,
The filter equation is expressed by the following equation,
Equation

(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,
remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)
The first element may be any of the

Wherein the post-processing beam forming system comprises: delete A first step of loading a plurality of files storing a plurality of sound signals obtained by each of a plurality of microphones arrayed at regular intervals;
A plurality of sound signals stored in the plurality of files are filtered using a filter expression, and a value of a first element of the filter expression is adjusted to filter a sound signal of a predetermined direction among the plurality of sound signals to be adjusted, Step 2; And
And a third step of merging and outputting the plurality of filtered sound signals,
Wherein the sound signal in the predetermined direction is at least one of the plurality of sound signals,
The filter equation is expressed by the following equation,
Equation

(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,
remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)
The first element may be any of the

Wherein the post-processing beam forming method comprises: delete A recording medium on which a program of instructions executable by a digital processing apparatus to perform a beamforming method is tangibly embodied and which can be read by the digital processing apparatus,
The beam forming method includes:
A first step of acquiring a plurality of sound signals using a plurality of microphones arrayed at regular intervals;
Wherein the plurality of sound signals obtained by the plurality of microphones are filtered using a filter expression, and the value of the first element of the filter expression is adjusted to filter the sound signals of the predetermined direction among the plurality of sound signals, ; And
And a third step of merging and outputting the plurality of filtered sound signals,
Wherein the sound signal in the predetermined direction is at least one of the plurality of sound signals,
The filter equation is expressed by the following equation,
Equation

(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,
remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)
The first element may be any of the

And a recording medium. A recording medium on which a program of instructions executable by a digital processing apparatus to perform a post processing beamforming method is tangibly embodied and which can be read by the digital processing apparatus,
The post-processing beamforming method includes:
A first step of loading a plurality of files storing a plurality of sound signals obtained by each of a plurality of microphones arrayed at equal intervals;
A plurality of sound signals stored in the plurality of files are filtered using a filter expression, and a value of a first element of the filter expression is adjusted to filter a sound signal of a predetermined direction among the plurality of sound signals to be adjusted, Step 2; And
And a third step of merging and outputting the plurality of filtered sound signals,
Wherein the sound signal in the predetermined direction is at least one of the plurality of sound signals,
The filter equation is expressed by the following equation,
Equation

(remind

The

, And

The directivity of the microphone located at the N-th position among the plurality of microphones,

Is the frequency of the sound signal,

Represents an azimuth angle among elements indicating the directivity,

Represents a zenith angle of an element representing the directivity,

Is the speed of the sound signal,

Represents a position of a microphone located at the Nth position among the plurality of microphones with the center of a first line that is a straight line connecting a plurality of microphones as an origin,

Is an angle formed by the first line and the second line connecting the sound sources of the plurality of sound signals from the center point of the first line,
remind

Quot;

Hermitian matrix of < RTI ID = 0.0 >

The

, And

Is a stabilization coefficient,

Is a unitary matrix.)
The first element may be any of the

And a recording medium.

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