KR102174850B1 - Environment adaptation type beam forming apparatus for audio - Google Patents

Abstract

The present invention discloses an environmentally adaptive audio beam forming apparatus. The environmentally adaptive audio beam forming apparatus according to the present invention includes a microphone array receiving at least two sound sources from an installed area, and a main sound source generation point of the installed area by analyzing an angle between each of the received sound sources received by the microphone array. And a deflection control unit for synchronizing time attributes of each sound source through correlation information including the main sound source generation point, and a mixer configured to receive the synchronized sound sources and output multiplexed deflection sound sources.

Description

Environment adaptation type beam forming apparatus for audio

The present invention relates to an environmentally adaptive audio beam forming apparatus, and more particularly, an environmentally adaptive for extracting an optimal audio beam from the surrounding environment and improving the listening effect based on the attributes of the extracted optimal audio beam. It relates to an audio beam forming apparatus.

Acquisition of sound pickup of the surveillance camera is possible by converting the sound signal of the omni-directional capacitor into a voltage. These acoustic signals, without directionality, depend on the directivity of the front or of the microphone, and have a significant effect on the detection distance and sound quality.

In particular, it is difficult to determine the sound object for the location where the surveillance camera is to be installed, which sound is mainly present or which specific sound is present at what level.

Due to this limitation, the sound related technology in the surveillance camera is limited to extremely limited functions, and a considerable extra cost is required to implement more special sound related functions in the surveillance camera, as well as a considerable amount to support the special sound related functions. This is possible only when the database of the database is built.

In addition, even if a considerable amount of database is built to support the special sound-related functions and algorithms to be executed in the surveillance camera are secured, various noises appearing during sound learning are effectively removed to secure sound detection capabilities to a practical level. There is a limit to this.

And, the directivity and sensitivity of the microphone show a significant performance difference depending on the indoor and outdoor environment.

In the case of indoors, there is a limitation in processing signals by differentially distinguishing sounds transmitted after reverberation according to not only the shape and size of the space around the microphone is located, but also the material of the wall.

In the case of outdoors, unlike indoors, the reverberation of sound may be relatively small, but there is a disadvantage that the distance according to sensitivity decreases compared to indoors when a microphone under the same conditions is used.

In addition, when a surveillance camera is installed adjacent to an air conditioning facility such as an underground parking lot, it is impossible to obtain smooth sound absorption due to a considerable amount of noise generated from an adjacent air conditioner.

Therefore, there is a need for a method for more easily realizing the acquisition of sound pickup of a surveillance camera.

Republic of Korea Patent Publication No. 10-2012-0068875 (2012.06.27)

Accordingly, the present invention was created to solve the above problem, and the object of the present invention is to extract the optimal audio beam from the surrounding environment, and to synchronize the electrical deflection delay based on the attributes of the extracted optimal audio beam. It is to provide an environmentally adaptive audio beam forming device that converts a received signal into a signal that is advantageous for listening through a method.

The subject of the present invention is not limited to the above-mentioned object, and other objects that are not mentioned will be clearly understood by those skilled in the art from the following description.

The environment-adaptive audio beam forming apparatus according to the aspect of the present invention for achieving the above object is a microphone array receiving at least two sound sources from an installed area, and analyzing the angle between the received sound sources received by the microphone array and the A deflection control unit that determines the main sound source generation point of the installed area, synchronizes the time attributes of each sound source through correlation information including the main sound source generation point, and outputs multiplexed deflection sound sources by receiving the synchronized sound sources. Includes a mixer.

The deflection control unit includes a sound source region detection module that detects an effective sound source region of each of the received sound sources, a correlation estimation module that calculates the correlation information through a phase difference and signal strength between the detected effective sound source regions, and the correlation A deflection delay coefficient calculation module that calculates a deflection delay coefficient through information and a predetermined synchronization table, and a deflection delay module that synchronizes time attributes of each received sound source based on the deflection delay coefficient.

The deflection control unit may further include a search signal generation module for controlling generation of a search signal serving as a source for the at least two sound sources.

The deflection control unit has an installation mode for setting synchronization between each of the received sound sources, and an operation mode for synchronizing each of the received sound sources based on a preset synchronization setting in the installation mode, and can be separately executed for each mode. .

The deflection control unit may set synchronization between the received sound sources whenever there is each of the received sound sources, and may perform synchronization between the received sound sources based on a synchronization setting changed in real time.

When setting synchronization between the received sound sources, the deflection control unit may specify sound sources to be analyzed among received sound sources based on a predetermined analysis pool selection criterion.

Therefore, in the present invention, the received signal can be converted into a signal advantageous for hearing through a synchronization method in which an optimal audio beam is extracted from the surrounding environment and electrically deflected based on the properties of the extracted optimal audio beam. There is an advantage.

The effects of the present invention are not limited to the effects mentioned above, and other effects that are not mentioned will be clearly understood by those skilled in the art from the description of the claims.

1 is a block diagram showing an environment-adaptive audio beam forming apparatus according to the present invention.
FIG. 2 is a circuit diagram illustrating the audio beam forming apparatus of FIG. 1 as an example.
3 is a block diagram showing a deflection control unit of FIG. 1 as an example.
4 is a block diagram illustrating a deflection control unit of FIG. 1 according to another embodiment.
5 is a conceptual diagram illustrating an example of a pickup method of the device shown in FIG. 1.
6 is a conceptual diagram illustrating another example of a pickup method of the device shown in FIG. 1.
7 is a conceptual diagram showing another example of a pickup method of the apparatus shown in FIG. 1.
8 is a conceptual diagram illustrating a deflection control method of the device shown in FIG. 1 as an example.
9 is a conceptual diagram illustrating a deflection control method of the apparatus shown in FIG. 1 as another example.
10 is a flowchart illustrating an example of an operation process of the audio beam forming apparatus of FIG. 1.
11 is a flowchart illustrating another example of an operation process of the audio beam forming apparatus of FIG. 1.
12 is a flowchart illustrating another example of an operation process of the audio beam forming apparatus of FIG. 1.
13 is an exemplary diagram illustrating a first sound-receiving characteristic through audio beamforming according to the present invention.
14 is an exemplary diagram showing a second sound-receiving characteristic through audio beamforming according to the present invention.
And, FIG. 15 is an exemplary diagram showing a third sound-receiving characteristic through audio beamforming according to the present invention.

Advantages and features of the present invention, and a method of achieving them will become apparent with reference to the embodiments described below in detail together with the accompanying drawings. However, the present invention is not limited to the embodiments disclosed below, but may be implemented in a variety of different forms, and only these embodiments make the disclosure of the present invention complete, and are common knowledge in the technical field to which the present invention pertains. It is provided to completely inform the scope of the invention to those who have, and the invention is only defined by the scope of the claims. The same reference numerals refer to the same components throughout the specification.

Further, the embodiments described in the present specification will be described with reference to sectional views and/or schematic diagrams, which are ideal exemplary diagrams of the present invention. Therefore, the shape of the exemplary diagram may be modified by manufacturing technology and/or tolerance. In addition, in each of the drawings shown in the present invention, each component may be somewhat enlarged or reduced in consideration of convenience of description.

Hereinafter, a preferred embodiment of the present invention will be described in detail with reference to the accompanying drawings.

1 is a block diagram showing an environment-adaptive audio beam forming apparatus according to the present invention.

Referring to FIG. 1, the environment-adaptive audio beam forming apparatus 100 extracts an optimal audio beam from the surrounding environment, and is electrically deflected based on the attributes of the extracted optimal audio beam. It is equipped with a configuration that converts the received signal into a signal favorable to

In addition, the environment-adaptive audio beam forming apparatus 100 may be provided in equipment that receives sound sources from the surrounding environment. For example, the environment-adaptive audio beam forming apparatus 100 may be provided in a surveillance camera. Hereinafter, for a more detailed description, an environmentally adaptive audio beam forming apparatus 100 provided in a surveillance camera will be mainly described.

In the audio beam forming apparatus 100, a microphone array 110 is provided to receive an audio beam, that is, a specific sound source. Here, the microphone array 110 refers to a structure in which a plurality of microphones are arranged.

The above-described microphone pickup pattern can be classified into omni-directional, bi-directional, and unidirectional. That is, in the case of a microphone having a specific sound-receiving pattern among omni-directional, bi-directional, and uni-directional, it is impossible to receive all sound sources in the surrounding environment, and it is possible to receive only sound sources corresponding to the specific sound-receiving pattern. This means that there is a limitation in that the field of view (FOV) of the surveillance camera and the pickup pattern of the microphone cannot be matched.

In the case of an omni-directional condenser microphone, the relative distance can be calculated according to the signal to noise ratio (SNR), but this is when a surveillance camera and a microphone are installed in a place where noise is irregularly generated, the intensity of noise is equal to or greater than the signal. In this case, it is difficult to analyze the sound signal, and in the case of outdoors, the sound attenuation dB SPL (Sound Pressure Level) by distance of the sound of the atmospheric model according to the distance increases as the distance increases.

For example, if the current measurement noise through a microphone is 50 dB SPL, and 70 dB is measured when a sound source is generated at 1 m (SNR 20 dB), if a sound of the same size occurs at 10 m, -20 dB is attenuated and is monitored when it reaches 50 dB. There is no difference from the noise adjacent to the camera.

Since the noise and the sound source are the same size, it is difficult to detect the sound source, and it is also difficult to determine whether the input to the microphone is noise or the sound source to be detected. Therefore, there is a need for improvement in the performance of estimating the acoustic direction by the SNR input to the microphone rather than the distance.

In addition, in the case of an omni-directional condenser microphone used in a surveillance camera, assuming that the sound detection level is 10dB and the sensitivity is a microphone with a performance of approximately -44dB, the sound detection distance must be 10dB or more to enable sound detection. -(10dB) = -33dB', it can be estimated as 4~5m distance. In other words, there is a limitation in implementing a surveillance camera capable of receiving sound with a characteristic sensitive to noise changes by applying it to surveillance cameras installed inside and outside the room as a low-cost omnidirectional condenser microphone.

In the case of omni-directional in terms of directivity, it is possible to detect sound in a range of 90 degrees to the left and right from the center of the front part, and the sound in the 180-degree direction is less sensitive than the front part, but the echoed sounds and the FOV of the box-type surveillance camera You can also pick up the sound of the part that is off. On the other hand, in the case of receiving only the area to be monitored, it is necessary to adjust the installation direction of the microphone.

In addition, in the case of narrowing or widening the directional area, it is necessary to replace it with a directional microphone, and due to the nature of the surveillance camera, the space for mechanically mounting related parts is limited. It is limited.

Therefore, there is an advantage in realizing the beam forming adjustment in an electrical method rather than a mechanical method after acoustic tracking according to the installation surrounding environment of the surveillance camera.

To this end, the environment-adaptive audio beam forming apparatus 100 of the present invention analyzes the angle between the microphone array 110 receiving at least two sound sources from the installed area and each sound source received by the microphone array 110 to be installed area A deflection control unit 120 that determines the main sound source generation point of and synchronizes the time attributes of each sound source through correlation information including the determined main sound source generation point, and outputs multiplexed deflection sound sources by receiving the synchronized sound sources. Includes a mixer 130.

FIG. 2 is a circuit diagram illustrating the audio beam forming apparatus of FIG. 1 as an example.

Referring to FIG. 2, the beam forming apparatus 100 may further include an amplifier 140, a stereo analog-digital converter (ADC) 150, and a digital-analog converter (DAC) 160.

The amplifier 140 may amplify the sound source provided from the microphone array 110 to facilitate signal analysis, or may amplify the multiplexed deflected sound source output from the mixer 130 to a predetermined output value.

The stereo ADC 150 quantizes the amplified sound source, converts it into a state in which signal analysis is possible, and outputs the converted signal.

The above-described deflection control unit 120 may be provided in the form of a DSP that exists separately from the main processor of the surveillance camera.

As the converted signal output from the stereo ADC 150 is transmitted to the DSP-type deflection control unit 120, signal processing corresponding to the surrounding environment of the surveillance camera is performed, thereby generating a signal with improved hearing effect. .

The DAC 160 converts the signal output from the DSP into an analog signal and then transmits the converted signal to the mixer 130.

3 is a block diagram illustrating a deflection control unit of FIG. 1 as an example, and FIG. 4 is a block diagram illustrating a deflection control unit of FIG. 1 as another exemplary embodiment.

Referring to FIG. 3, the deflection control unit 120 includes a sound source region detection module 120-1-1 for detecting an effective sound source region of each sound source, which is converted signals output from the stereo ADC 150, and the detected effective sound source region. A correlation estimation module 120-1-2 that calculates correlation information through the phase difference and signal strength between them, a deflection delay coefficient calculation module that calculates a deflection delay coefficient through the calculated correlation information and a predetermined synchronization table ( 120-1-3) and a deflection delay module 120-1-4 for synchronizing time attributes of each sound source received based on the deflection delay coefficient.

4, the deflection control unit 120 includes a sound source region detection module 120-2-1, a correlation estimation module 120-2-2, a deflection delay coefficient calculation module 120-2-3, and In addition to including the deflection delay module 120-2-4, it may further include a search signal generation module 120-2-5.

The search signal generation module (120-2-5) is a module that controls the generation of search signals serving as sources for at least two sound sources, and does not directly receive sound sources from the surrounding environment by installing a surveillance camera. It can be used for the purpose of artificially generating sound sources by predicting the surrounding environment of the installation. Therefore, it is preferable that the search signal generation module generates not only the search signal of a specific sound source, but also various types of search signals.

FIG. 5 is a conceptual diagram illustrating a pickup method of the device illustrated in FIG. 1 as an example, FIG. 6 is a conceptual diagram illustrating a pickup method of the device illustrated in FIG. 1 as another example, and FIG. 7 is a pickup method of the device illustrated in FIG. It is a conceptual diagram showing another example.

Referring to FIG. 5, when two sound sources deflected in the left 45 degree direction are received by the microphone array 110, a time difference occurs when each sound source reaches the microphone array 110. Accordingly, even when outputting each sound source due to a time difference that has occurred, the microphone array 110 outputs a state in which a time difference that has already occurred exists between the sound sources.

At this time, when the synchronization is not performed through the deflection control unit 120 of the present invention through the electric deflection delay method, the signal output from the microphone array 110 while maintaining the time difference between the sound sources is maintained as is, the mixer 130 Is delivered to. As a result, the mixer 130 weights each sound source in a state in which the time difference existing between the sound sources is reflected, thereby causing loss of a signal due to a decrease in the correlation between the two signals as shown in FIG. Accordingly, signal distortion may also occur.

Referring to FIG. 6, when two sound sources that are not deflected by the microphone array 110 are received, there is no time difference between each sound source reaching the microphone array 110.

Accordingly, the microphone array 110 simultaneously receives each sound source and outputs each sound source without a time difference as it is.

In this case, when synchronization is not performed through the deflection control unit 120 of the present invention in an electrical deflection delay method, a signal output from the microphone array 110 is transmitted to the mixer 130. As a result, the mixer 130 weights each sound source without a time difference, thereby increasing the correlation between the two signals, as shown in FIG. 6, so that signal loss does not occur and the maximum signal strength can be formed. have. In this case, even when the deflection control unit 120 is not executed, an optimal audio beam for improving sound quality received from the surroundings of the surveillance camera may be extracted.

Referring to FIG. 7, when two sound sources deflected in the right 45 degree direction are received by the microphone array 110, a time difference occurs when each sound source reaches the microphone array 110. Accordingly, even when outputting each sound source due to a time difference that has occurred, the microphone array 110 outputs a state in which a time difference that has already occurred exists between the sound sources.

At this time, when the synchronization is not performed through the deflection control unit 120 of the present invention through the electric deflection delay method, the signal output from the microphone array 110 while maintaining the time difference between the sound sources is maintained as is, the mixer 130 Is delivered to. As a result, the mixer 130 weights each sound source in a state in which the time difference existing between the sound sources is reflected, thereby causing loss of a signal due to a decrease in the correlation between the two signals as shown in FIG. 7, Accordingly, signal distortion may also occur.

Therefore, when the sound is received in the state as shown in FIGS. 5 and 7, it is necessary to perform the deflection control unit 120 to synchronize the time attribute of the sound source received.

8 is a conceptual diagram illustrating a deflection control method of the device shown in FIG. 1 as an example.

Referring to FIG. 8, when two sound sources deflected in the left 45 degree direction are received by the microphone array 110, the deflection control unit 120 calculates the phase difference between the two sound sources, including the time difference between the two sound sources. Correlation information is calculated, and time properties between the received sound sources are synchronized using the calculated correlation information.

The deflection control unit 120 uses a first sound source among the two sound sources received as a synchronization reference, and calculates a time difference between the first sound source and the second sound source, which are two sound sources. Further, the deflection control unit 120 synchronizes the time attributes of the two sound sources by delaying the time attributes of the second sound source by the calculated time difference.

For example, the deflection control unit 120 sets a sound source that has arrived late to the microphone array 110 among the two sound sources as the first sound source, which is the synchronization criterion, and first among the two sound sources, the microphone array 110 The reached sound source is set as the second sound source and is subjected to deflection control.

9 is a conceptual diagram illustrating a deflection control method of the apparatus shown in FIG. 1 as another example.

Referring to FIG. 9, it may be assumed that two sound sources that are not deflected by the microphone array 110 are received and the deflection control setting set in FIG. 8 is maintained. In this case, unwanted distortion may occur as one of the two sound sources that are not deflected is deflected.

That is, it is preferable that the deflection control unit 120 determines a degree of deflection between at least two sound sources received by the microphone array 110 and performs deflection control based on the determination result.

In addition, when there are three or more sound sources to be received, the deflection control unit 120 may select an optimal audio beam by extracting two sound sources to be synchronized among the three or more sound sources. That is, when setting synchronization between the received sound sources, the deflection control unit 120 may specify sound sources to be analyzed among the received sound sources based on a predetermined analysis pool selection criterion.

For example, when two sound sources deflected in the left 45 degree direction are received and one sound source is received in the right 45 degree direction, the deflection control unit 120 selects two sound sources in the left 45 degree direction with the same main deflection direction. Can be set as a synchronization target.

On the other hand, the deflection control unit 120 has an installation mode for setting synchronization between each received sound source and an operation mode for performing synchronization between each received sound source based on a preset synchronization setting in the installation mode. I can.

Alternatively, the deflection control unit 120 may set synchronization between each received sound source whenever there are received sound sources, and may perform synchronization between each received sound source based on a synchronization setting varied in real time.

10 is a flowchart illustrating an example of an operation process of the audio beam forming apparatus of FIG. 1.

Referring to FIG. 10, the beam forming apparatus 100 may perform beamforming by receiving a sound source reproduced from its own speaker in a silent room before the surveillance camera is shipped with its own microphone array 110, and for this purpose, first A search signal generation control for generating a sound source reproduced from its own speaker is executed (S100).

Thereafter, the beam forming apparatus 100 receives the sound source output through the own speaker through the microphone array 110 by generating a search signal, and then changes a deflection coefficient for deflecting the received sound source (S102).

Thereafter, the signal strength of the sound source received from the microphone array 110 is obtained while the beam width is fixed (S104).

Thereafter, the difference between the sound pressure coefficient stored before shipment and the sound pressure received at the installation site is calculated (S106), and the sound pressure coefficient in the installation environment is stored or the previous sound pressure coefficient is updated (S108).

The steps S102 to S108 are repeated until a predetermined threshold number of executions (S110).

Based on the result of going through step S110, the sound pressure coefficient for all directions is obtained. The larger the obtained sound pressure coefficient, the more information that an obstacle is adjacent can be obtained. This information can be used as an indicator to be used to adjust the beam width and beam direction in normal operation.

When the threshold number of executions arrives, the generation control of the search signal is stopped (S112).

11 is a flowchart illustrating another example of an operation process of the audio beam forming apparatus of FIG. 1.

Referring to FIG. 11, the beam forming apparatus 100 directly receives a sound source in the installation environment from the installation environment of the surveillance camera (S200).

Thereafter, the beam forming apparatus 100 receives the sound source in the installation environment with the microphone array 110, and then varies a deflection coefficient for deflecting the received sound source, and the microphone array 110 in a state where the beam width is fixed. The signal strength of the sound source received from is obtained (S202).

Thereafter, the difference between the sound pressure coefficient stored before shipment and the sound pressure received at the installation site is calculated, and the sound pressure coefficient in the installation environment is stored or the previous sound pressure coefficient is updated (S204).

The steps S202 to S204 are repeated until a predetermined threshold number of executions (S206).

Based on the result of going through step S206, the sound pressure coefficient for all directions is obtained. The larger the obtained sound pressure coefficient, the more information that an obstacle is adjacent can be obtained. This information can be used as an indicator to be used to adjust the beam width and beam direction in normal operation.

When the threshold number of executions arrives, the search signal generation control is stopped (S208).

12 is a flowchart illustrating another example of an operation process of the audio beam forming apparatus of FIG. 1.

Referring to FIG. 12, the beam forming apparatus 100 may operate in a manual mode or an automatic mode according to a user's selection, and manual operation may be determined on a GUI screen within the FOV of the surveillance camera (S300).

When the automatic mode is set instead of the manual mode in step S300, the beam forming apparatus 100 detects an effective sound source region after receiving sound from the surroundings of the surveillance camera (S302).

If there is a valid sound source detected in step S302, an angle between sound sources constituting the effective sound source is analyzed to determine a main sound source generation point among the installed regions (S306).

Thereafter, in a state in which the optimal audio beam is extracted by synchronizing the time attributes of each sound source through correlation information including the determined main sound source generation point, the direction and beam of the optimal audio beam at the listening level based on the initially set user input. The width is adjusted to match the favorable condition for listening (S308).

Thereafter, the beam forming adjustment in which the deflection coefficient is converted and reflected may be continued or the beam forming adjustment may be terminated (S310 and S312).

In addition, when the manual mode is set in step S300, the beam forming apparatus 100 further performs a process of receiving a manual input from the user (S300-1).

For reference, FIG. 13 is an exemplary diagram showing a first sound-receiving characteristic through audio beamforming according to the present invention, FIG. 14 is an exemplary diagram showing a second sound-receiving characteristic through audio beam forming according to the present invention, and FIG. 15 is It is an exemplary diagram showing a third sound-receiving characteristic through audio beamforming of.

The first sound-receiving characteristic shown in FIG. 13 is a characteristic in which the sound is received by being deflected 45 degrees to the left, the second sound-receiving characteristic shown in FIG. 14 is a characteristic in which the sound is received without being deflected, and the third sound-receiving characteristic shown in FIG. This is an illustration of the characteristic of being deflected by 45 degrees to the right.

Although the embodiments of the present invention have been described with reference to the above and the accompanying drawings, those of ordinary skill in the art to which the present invention pertains can be implemented in other specific forms without changing the technical spirit or essential features. You can understand that there is. Therefore, it should be understood that the embodiments described above are illustrative and non-limiting in all respects.

In addition, the present invention extracts the optimal audio beam from the surrounding environment, and converts the received signal into a signal favorable to hearing through a synchronization method that electrically deflects delay based on the attributes of the extracted optimal audio beam. As it is intended to provide a type audio beam forming device, it is an invention that has industrial applicability because it is not only sufficiently commercially available or commercially possible, but also can be implemented clearly in reality.

100: beam forming device 110: microphone array
120: deflection control unit 120-1-1, 120-2-1: sound source region detection module
120-1-2, 120-2-2: correlation estimation module
120-1-3, 120-2-3: deflection delay coefficient calculation module
120-1-4, 120-2-4: deflection delay module
120-2-5: Search signal generation module
130: mixer 140: amplifier
150: stereo ADC 160: DAC

Claims (6)

A microphone array for receiving at least two sound sources from the installed area;
Deflection for determining the main sound source generation point of the installed area by analyzing the angle between the received sound sources received by the microphone array, and synchronizing the temporal properties of each sound source through correlation information including the main sound source generation point Control unit; And
Including a mixer for receiving each of the synchronized sound source and outputting the multiplexed deflected sound source,
The deflection control unit includes a sound source region detection module for detecting an effective sound source region of each of the received sound sources;
A correlation estimation module for calculating the correlation information through the phase difference and signal strength between the detected effective sound source regions;
A deflection delay coefficient calculation module for calculating a deflection delay coefficient through the correlation information; And
Environment-adaptive audio beam forming apparatus comprising a deflection delay module for synchronizing time attributes of each of the received sound sources based on the deflection delay coefficient. delete The method of claim 1,
The deflection control unit further comprises a search signal generation module for controlling generation of a search signal serving as a source for the at least two sound sources. The method of claim 1,
The deflection control unit has an installation mode for setting synchronization between each of the received sound sources and an operation mode for synchronizing each of the received sound sources based on a synchronization setting predetermined in the installation mode, and an environment in which each mode is separately executed Adaptive audio beam forming device. The method of claim 1,
The deflection control unit sets the synchronization between the received sound sources whenever each of the received sound sources is present, and an environment-adaptive audio beamforming device that performs synchronization between the received sound sources based on a synchronization setting varied in real time. . The method according to claim 4 or 5,
When the deflection control unit sets the synchronization between the received sound sources, the environment adaptive audio beamforming apparatus specifies sound sources to be analyzed among received sound sources based on a predetermined analysis pool selection criterion.

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