KR101721621B1 - Audio amplifying apparatus united microphone and speaker - Google Patents

Abstract

The present invention relates to an audio amplifying apparatus integrated with a microphone and a speaker, capable of collecting a sound by performing beamforming toward a speaker after accurately tracking the speaker under an echo environment when a multi-party conversion is performed, and amplifying and outputting a voice of the speaker in an output direction matched with a communication direction of the speaker. The audio amplifying apparatus integrated with a microphone and a speaker is configured to be installed in the center part of a place where persons are gathered, collects the sound by performing the beamforming toward the speaker after tracking the position of the speaker by using a microphone array, suppresses an echo element to be matched with a beamforming direction according to a pre-trained method, and sets and uses the output direction to accurately communicate with all other conversion parties to be matched with the position of the speaker, if possible. The audio amplifying apparatus integrated with a microphone and a speaker includes an echo learning processor which obtains a filer coefficient, an echo remover which suppresses the echo element, a sound source tracker which tracks a sound source direction, and a switch which outputs a sound signal to suppress the echo element.

Description

TECHNICAL FIELD [0001] The present invention relates to an audio amplifying apparatus,

The present invention relates to a microphone-speaker-integrated audio amplifying apparatus for accurately tracking a speaker even in an echo environment when multiply-communicated, beam-forming and collecting the speaker toward a speaker, and amplifying and outputting the speaker's voice in an output direction corresponding to a communication direction of the speaker .

As an example of a multi-party conversation, when a large number of people gather for a meeting, an audio amplifying device is used in order to accurately transmit the contents to be uttered and communicated to all persons.

In such an audio amplifying apparatus, a microphone unit having a microphone and a speaker is generally used by each person, and when there is a microphone unit that relays each microphone unit and recognizes the voice of the speaker, To all the microphone units and outputs them. In addition, a function selection button may be provided in the microphone unit for the function of performing the amplification operation or the volume control function for reducing the occurrence of howling.

 In addition, as shown in Patent No. 10-1074188, it is possible to selectively output the voice of a speaker through an individual channel.

However, such an audio amplifying apparatus is required to have a microphone unit as many as the number of people, and a device for relaying between the microphone units must be additionally installed, so that it is difficult to carry, install and manage, It is necessary to provide an echo component suppressing means every time.

As another example, an audio amplifying device may be required even when a small number of people sit facing each other. However, in such a case, it is very inefficient to use a conventional audio amplifying device which is difficult to transport, install and manage and is expensive.

Furthermore, when two or more people gather to face each other, or to sit in a conversation, a meeting, or a consultation, it is common to amplify and output the voice of the speaker so that the direction toward the speaker It is preferable to amplify and output it, but it is difficult or impossible in a conventional audio amplifying apparatus in which each person uses a microphone unit.

KR 10-1074188 B1 Oct. 10, 2011.

Accordingly, a problem to be solved by the present invention is to provide a speaker system in which each person is constituted by a simple structure that does not use a microphone unit one by one and is more convenient for carrying, installing and using, And an output direction is set according to the output of the microphone, thereby amplifying and outputting the voice of the speaker.

According to another aspect of the present invention, there is provided an audio amplifying apparatus comprising a microphone and a microphone, wherein the speaker for amplifying and outputting acoustic signals obtained from the microphone and the microphone is integrated into one case or adjacent thereto within 1m, An echo learning processor for modeling an echo channel on a space where a microphone is located by using a filter in advance to obtain a filter coefficient and a filter to which the filter coefficient is applied to an acoustic signal acquired by the microphone to suppress an echo component due to an echo channel, And an echo canceller for outputting the echo canceller.

And a plurality of the speakers are arranged in the circumferential direction around the microphone.

The microphones are further provided with a plurality of microphones and are spaced apart from each other. The microphones further include a sound source tracker for correlating and analyzing sound signals of the microphones to track the direction of sound sources.

And a switch for selecting a speaker in the output direction based on the sound output direction of each sound source direction which is predetermined for each sound source direction in advance and outputting an acoustic signal suppressing the echo component.

Wherein the echo learning processor obtains a filter coefficient for each microphone by modeling an echo channel formed for each microphone by a filter, and the sound source tracker suppresses an echo component individually for each microphone signal by applying a filter coefficient for each microphone, And the direction of the sound source is analyzed.

Further comprising a beam former for processing an acoustic signal acquired from at least two or more microphones to acquire a beamformed acoustic signal in a sound source direction, wherein the echo canceller suppresses echo components of the beam- And acquires an acoustic signal to be output.

Wherein the echo learning processor obtains a filter coefficient for each beam forming direction by modeling an echo channel generated by the beam forming direction in advance by varying a beam forming direction in advance, and the echo canceller performs a beam forming operation for a beam- The echo component is suppressed based on the filter coefficient corresponding to the filter coefficient.

And a communication module for short-range wireless communication with the smartphone. The execution of the echo learning processor for acquiring the filter coefficient for the echo channel and the selection of the sound output direction for each sound source direction are performed by a smartphone .

Wherein the switch selects a speaker arranged at least in a direction opposite to a sound source direction and outputs a sound signal.

According to the present invention configured as described above, the components for suppressing echo and outputting the amplified output are integrated into one device so that they can be transported by a simple structure that can be used by connecting only the power source Easy to install, use and manage.

In addition, since the present invention collects signals by beamforming toward a speaker, only the echo components of the beamforming region are removed in a state where the inflow of the ambient noise and the echo components is minimized, and the echo suppression performance can be enhanced.

In addition, since the present invention selects the output direction differently for each sound source direction, it is possible to accurately convey the intention of the speaker.

Also, according to the present invention, the echo is removed for each microphone separately from the beamforming area to track the speaker, so that the speaker can accurately track the actual speaker without tracking the false speaker due to the echo.

Also, according to the present invention, echo cancellation for sound source tracking and echo cancellation for sound collection are independently performed on an acoustic signal inputted through a microphone, and accurate sound source position tracking and echo suppressed signal acquisition are simultaneously satisfied.

FIG. 1 is a perspective view of an audio amplifying apparatus integrated with a microphone according to an embodiment of the present invention. FIG.
BACKGROUND OF THE INVENTION Field of the Invention [0001] The present invention relates to an audio amplifying apparatus, and more particularly,
3 is an electric signal configuration diagram of an audio amplifying apparatus with a built-in microphone in accordance with an embodiment of the present invention.
4 is a block diagram of a sound source tracker 20 connected to the microphone array 10. FIG.
5 is a view for explaining a principle of sound source tracking;
6 is a schematic block diagram of a beamformer 30 connected in parallel with a source tracker 20 to a microphone array 10. Fig.
7 is a flowchart of an audio amplification method by a microphone-integrated audio amplification apparatus according to an embodiment of the present invention.
FIG. 8 is a use state diagram of an audio amplifying apparatus with a built-in microphone speaker according to an embodiment of the present invention; FIG.
FIG. 9 is a use state diagram of an audio amplifying apparatus integrated with a microphone according to another embodiment of the present invention. FIG.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Hereinafter, preferred embodiments of the present invention will be described with reference to the accompanying drawings.

Referring to the perspective view of FIG. 1 and the top view of FIG. 2, a microphone-integrated audio amplifying apparatus according to an embodiment of the present invention includes a microphone for acquiring a sound wave transmitted by a speaker in any direction, S1, S2, S3, and S4 for echoing and outputting the acoustic signals obtained from the speakers M1, M2, M3, and M4 and the microphone by using the built-in components as described later, For example, when a plurality of people in the vicinity communicate with each other by communicating with each other, the voice of the speaker can be amplified and heard, and the components for performing the amplification operation are integrated into a single case (A), it is convenient to carry, move, install and use.

Although a plurality of the microphones M1, M2, M3, and M4 are shown, it is also possible to provide only one microphone. Similarly, the plurality of speakers S1, S2, S3, and S4 may be provided. It is also possible to do.

However, in a specific embodiment of the present invention, a plurality of micro-array microphones 10 are used for collecting and outputting only the speaker's voice, and a plurality of speakers S1, S2, S3 , S4) were used.

As shown in the electric signal configuration diagram of FIG. 3, the audio amplifying apparatus for this purpose includes a plurality of speakers S1, S2, S3, and S4 having different arranging directions, and a speaker (sound source) Beam forming means (10, 20, 30) employing a microphone array (10) to obtain an acoustic signal beam -formed in the direction of a speaker accurately by tracking the echo component, A switch 60 for determining an output direction of a sound signal, an echo learning processor 50 for learning an echo environment in advance for suppressing echo components, and an operation mode according to a request of the user And the control unit 70 is provided in one case A.

The plurality of loudspeakers S1, S2, S3, and S4 are disposed on the outer side surface of the case A at intervals along the circumferential direction, and are preferably arranged in an isosceles square shape to enable sound output in any direction good.

Here, the case (A) may be a structure in which the speaker and the microphone array 10 can be installed so as to be suitable for the performance of the function. First, as the structure for installing the speaker, in the specific embodiment shown in the figure, And a speaker may be provided on each side of the speaker. Alternatively, a speaker may be provided on each side of the speaker in a rectangular column shape having an isosceles right angle Speakers may be installed at the position.

On the other hand, the microphone array 10 is installed on the upper surface of the case A so that each of the microphones M1, M2, M3, and M4 constituting the microphone array 10 receives sound waves propagated independently from each other, M2, M3, and M4 constituting the microphone array 10 are provided so as to be spaced apart from each other when they are installed on the upper surface of the case A so as to protrude therefrom.

The beam forming means 10, 20 and 30 are means for acquiring a beamformed acoustic signal by tracking the direction of a sound source (e.g., a speaker) that generates sound in the surroundings and includes a microphone array 10, a sound source tracker 20, and a beamformer 30. As shown in FIG.

The microphone array 10 is constituted by arranging at least two microphones M1, M2, M3, and M4 for receiving sound waves propagated by sounds generated from nearby sound sources and converting the sound waves into electric acoustic signals, Since the microphones M1, M2, M3, and M4 are spaced apart from each other, the time for receiving sound waves may be different.

It is preferable to employ a method of arranging the plurality of microphones M1, M2, M3, and M4 equidistantly on an isosceles right angle with respect to the center of the upper surface of the case A in terms of the complexity of programs for source tracking and beamforming It is good to lower. That is, a microphone is provided at each of the circles on the basis of the center of the upper surface of the case (A) and at the equally divided positions, and the intervals between the microphones are uniform along the circumferential direction. Make it simple.

Here, each of the microphones M1, M2, M3, and M4 may be an omnidirectional microphone that senses sound waves coming from any direction, but if the microphone is regarded as having a direction of less than 180 but almost no direction, Respectively.

The sound source tracker 20 performs signal processing on the acoustic signals acquired by the microphones M1, M2, M3, and M4 in advance by using the echo canceling unit 21 that suppresses the echo components individually, A correlation analysis unit 22 that obtains a cross correlation between acoustic signals for respective microphones in which echo components are suppressed and tracks a sound source direction based on a time difference of sounds arriving at the respective microphones, To more accurately track the direction of the sound source.

4, an analog sound signal obtained through the microphones M1, M2, M3, and M4 is amplified by the amplifier 11 in the microphone array 10, and then amplified by an A / D converter 12 to be converted into a digital sound signal and then transmitted to the sound source tracker 20 so that digital signal processing can be performed.

The echo canceler 21 in the sound source tracker 20 is configured to subtract an echo signal generated by filtering the acoustic signal output from the speaker with the echo cancellation filter 21a from the acoustic signal by the subtracter 21b, And a plurality of acoustic signals received from the respective microphones M1, M2, M3, and M4 are individually provided for signal processing.

The echo cancellation filter 21a is a filter modeling an echo channel formed on a space between a corresponding microphone and a plurality of speakers S1, S2, S3 and S4. Coefficients are obtained. Since the positions of the microphones M1, M2, M3, and M4 are different from each other, the filter coefficient is obtained for each echo cancellation filter 21a corresponding to one microphone.

On the other hand, since the respective speakers S1, S2, S3, and S4 are also arranged in different directions, the echo channels will also be formed differently for each speaker. The filter coefficient for each echo cancellation filter 21a is obtained by obtaining a filter coefficient for each speaker, and when the filter coefficient is used, the echo component obtained by applying a corresponding filter coefficient to the acoustic signal output for each speaker is summed , And the summed echo component may be subtracted by the subtracter 21b.

Since the false sound source due to the echo is removed from the acoustic signal in which the echo component is suppressed for each microphone, the correlation analyzer 22 can more accurately track the sound source position. Here, the false sound source is a sound source that is detected by an acoustic signal inputted through an echo channel.

Since the correlation analyzer 22 is a known technique for acquiring sound source positions by calculating a cross correlation between sound signals obtained through a plurality of microphones as disclosed in, for example, Japanese Patent No. 10-0877914 5 will be briefly described.

5A and 5B show the arrangement of the microphones M1, M2, M3 and M4 and the arrangement between the microphones M1, M2, M3 and M4 and the sound source 1 on the coordinate plane, (B) is a graph showing waveforms of sound signals obtained through the microphones M1, M2, M3, and M4.

5, when a sound source 1 in the direction of the azimuth angle? With respect to the arrangement center of the microphones M1, M2, M3 and M4 sounds, sound waves due to the sound of the sound source 1 are transmitted to the respective microphones The times (? 1,? 2,? 3,? 4) reaching the sound sources 1, M1, M2, M3 and M4 may be different depending on the distances L1, L2, L3, L4 from the sound source 1 .

Thus, correlation analysis of the acoustic signals acquired at different times (? 1,? 2,? 3,? 4) results in a difference in arrival delay time.

Also, the distance difference can be obtained by reflecting the propagation speed of the sound wave, and the direction of the sound source 1 can also be obtained by applying the distance difference between the sound source and the microphone array position. The method of correlating and analyzing the acoustic signals obtained through the plurality of microphones to obtain the sound source direction is well known in, for example, Japanese Patent Application No. 10-0877914, and thus a detailed description thereof will be omitted.

However, in order to track the direction of the sound source, it is necessary to correlate the acoustic signals acquired from at least two microphones, and if possible, correlate and analyze the sound signals obtained from many microphones to track the direction of the sound source more accurately.

The sound source direction thus tracked is notified to the beam former 30, the echo canceller 40, and the switch 60 as shown by the broken line in Fig.

The beam former 30 processes an acoustic signal obtained from at least two microphones among the acoustic signals acquired through the microphones M1, M2, M3, and M4, and generates an acoustic signal beam- .

Here, the acoustic signal to be used is not an acoustic signal suppressing the echo component for each microphone, but an acoustic signal obtained from each microphone.

Beamforming of sound is a technique for extracting only sound waves propagated from a desired direction and generating an acoustic signal from which sound waves propagated from other directions are removed. For example, Japanese Patent Application Laid-Open No. 10-0873000 It will be briefly described with reference to the block diagram shown in Fig.

The beam former 30 adds complex weights w and w to the direction of the beam forming with respect to the acoustic signals received from the microphone array 10 for each of the microphones M1, M2, M3 and M4 ) To generate a (32) beamformed acoustic signal. Since the beamforming for a sound signal has a frequency-variant, a beamformed sound signal is generated by performing a series of processes of converting to a frequency domain, adding weighting, summing, and converting to a time domain .

Here, the complex weights (?, W) are applied to acoustic signals of respective microphones in different values according to the direction of the sound source, as described in Patent No. 10-0873000, When the weight is added and then summed, only the sound waves propagated in the direction of the sound source are picked up, and the sound waves propagated in the other direction are removed.

The sound signal thus beamformed is transmitted to the echo canceling unit 41. [

The echo canceller 40 is a component for suppressing echo components in a beamformed acoustic signal. The echo canceller 40 includes a domain-specific filter coefficient storage 42 for storing filter coefficients of a filter modeling an echo channel for each beam forming direction, A filter coefficient corresponding to a direction of a sound source is loaded upon receipt of a sound source tracker 20 and a beamformer 30 receives a beamformed sound signal and then an echo component mixed with the beam- And an echo canceling unit 41 for suppressing by applying a coefficient.

Echo components that are output from the speakers S1, S2, S3, and S4 and then input to the microphones M1, M2, M3, and M4 along the beam forming direction are also changed in the beam forming direction The filter coefficient for each beam forming direction is generated and stored in advance by the echo processing processor 50 as described later.

The echo canceller 41 applies the filter coefficient loaded from the filter coefficient storage unit 42 to the echo cancellation filter 41a and then filters the acoustic signal output to the speaker with the echo cancellation filter 41a, And then subtracts the echo component from the beamformed acoustic signal transmitted from the beam former 30 using the subtracter 41b.

Thus, the acoustic signal whose echo component is suppressed by the echo channel in the beam forming direction is transmitted to the switch 60.

The switch 60 is a component for determining an output direction of an acoustic signal in which the echo component is suppressed and outputting it through a speaker disposed in the output direction. The acoustic output direction is preset (or preset) And when an acoustic signal is received from the echo canceller 20 and the sound signal is received from the echo canceller 40, an output direction matching the sound source direction is selected based on the predetermined sound source output direction according to the predetermined sound source direction Thereby outputting the acoustic signal through the speaker disposed in the rear output direction.

The sound source output direction for each sound source direction may be to output a sound signal to a speaker disposed at least in a direction opposite to the sound source direction.

Of course, the acoustic signal is analog-converted by a D / A converter (not shown) and then output to a speaker.

However, when using the present invention to amplify and transmit a multi-party conversation, it is preferable to appropriately adjust the direction of magnification depending on the position of the person. That is, depending on the use situation of the apparatus according to the present invention, a method of selecting all the speakers arranged in the remaining directions other than the sound source direction may be appropriate, or a method of selecting at least one of the speakers arranged in the remaining directions except for the sound source direction May be appropriate.

Accordingly, in the embodiment of the present invention, the sound output direction of each sound source direction can be set by the controller 70 as described later.

The echo learning processor 50 is a component for modeling the echo channel for each microphone and the echo channel for each beam forming direction as a filter and acquiring a filter coefficient. The echo processor 50 includes a training signal generator 51 And an echo channel modeling unit 52 for obtaining the coefficients of the filters modeled on the echo channels for each microphone and the echo channels for each beam forming direction by using a known adaptive algorithm employed for modeling the echo channels as filters And will be described in detail below with reference to FIG.

The control unit 70 controls to perform a setting step S10, a learning step 20 and an operation step 30, which will be described later.

In the embodiment of the present invention, a communication module 71 for short-range wireless communication with a smartphone is provided so that the execution of the setting step S10, the execution of the learning step 20 and the execution of the operation step 30 are performed by the communication module 71, and the control unit 70 receives the instruction from the smartphone through the communication module 71 and performs the instruction.

Here, the user's instruction may include a user interface for sequentially selecting the setting step, the learning step, and the operating step, and may be directly received from the user. The user interface can be, for example, a keypad.

Although not shown in the drawing, the power supply for the operation of the present invention is of course also provided, and for example, electricity may be supplied from the outside.

According to the flowchart shown in FIG. 7, the user sequentially selects the execution of the setting step S10, the execution of the learning step 20 and the operation step 30, and uses the present invention.

First, when the user instructs to perform the setting step S10, the controller 70 guides the sound output direction according to the sound source direction.

The user selects a direction in which the acoustic signal obtained in the multi-party communication is amplified and sent, and selects the sound output direction according to the sound source direction according to the position of the person and the position of the apparatus according to the present invention It is preferable to select at least the opposite direction of the speaker centering on the device according to the present invention.

At this time, the selected sound output direction for each sound source direction is used to select a speaker in the switch 60 when performing the following operation step S30.

Next, when the user instructs the performance of the learning step S20, the control unit 70 operates the echo learning processor 50 to learn the echo environment by the echo channel for each microphone (S21) The echo environment by the channel is learned (S22).

Here, the learning of the echo environment is performed by outputting a training signal to a speaker, and at the same time, adaptively changing an echo signal inputted through a microphone and an echo signal estimation signal generated by filtering the training signal with a filter, Employs a known adaptive algorithm that updates the coefficients.

The echo canceller may individually model the echo channels on the basis of the echo signals input to the microphones corresponding to the echo cancellation filters 21a provided for the respective microphones in the sound source tracker 20 and sequentially model them or simultaneously model them.

At this time, the filter coefficient of each echo cancellation filter 21a obtained by modeling is used in the following operation step S30.

When modeling the echo channel by the echo cancellation filter 41a provided in the echo canceller 40, the beam weighting direction is varied by adjusting the complex weights w and w applied to the beam former 30, And the echo channels formed for each forming direction are respectively modeled.

That is, at this time, the beamformer 30 is operated to generate a beamformed acoustic signal, and the beamforming direction is varied at predetermined time intervals, and the beamformer (for example, 30) to reduce the error between the acoustic signal (here, the echo signal of the training signal) transmitted to the echo canceller 40 and the signal obtained by filtering the training signal with the echo cancellation filter 41a, And the echo channel in the beam forming direction is modeled by the echo cancellation filter 41a. Thus, it is possible to obtain a filter coefficient for each beam forming direction obtained when modeling by beam forming direction.

At this time, the obtained filter coefficient for each beam forming direction is stored in the filter coefficient storage unit 42.

Meanwhile, any of the steps of obtaining the filter coefficient for the microphone-specific echo channel (S21) and acquiring the filter coefficient for the echo channel for each beam forming direction (S22) may be performed first. This is because the signal processing block including the beam former 30 and the echo canceller 40 and the sound source tracker 20 are directly connected to the microphone array 10 and arranged in parallel.

Next, when the user instructs to perform the operation step S30 after performing the setting step S10 and the learning step S20 as described above, the control unit 70 controls the beam forming unit 10, 20, 30, The echo canceller 40, the switch 60, and the speakers S1, S2, S3, and S4 to perform the magnification operation.

For convenience of explanation, it is assumed that the sound source is a person as shown in the use state diagram shown in Fig.

According to the use state diagram of FIG. 8, there are many people sitting on a chair and sitting in a chair around the table. The apparatus according to the present invention placed on the table performs the setting step S10 and the learning step S20 as described above After that, the speech of the speaker is amplified and output by the operation step 30 as follows.

The sound source tracker 20 processes the acoustic signals of the microphones acquired through the plurality of microphones M1, M2, M3, and M4 constituting the microphone array 10 by a microphone-specific echo canceller 21, The echo component due to the echo channel formed on the space is removed, and then the cross-correlation analysis of the acoustic signal for each microphone by removing the echo component is performed (S31). At the same time, (S32).

There is a sound output from the speaker while the speaker speaks, and there may be a situation where another speaker is additionally generated. Therefore, the echo component is removed.

Since the power of the acoustic signal acquired through the microphone can be obtained by estimating during the correlation analysis process, it is also possible that a human being is uttered when it is sensed as a predetermined power or more. That is, in an environment where the apparatus according to the present invention is actually used, there may exist noise such as wind noise and object sound in addition to sounds due to human voices, so that the preset power is appropriately selected, It does not make it sound like a voice.

When the presence of the speaker is confirmed, the position of the speaker is obtained from the result of the correlation analysis (S33).

The obtained speaker position is notified to the beam former 30, the echo canceller 40 and the switch 60 so that the beam forming operation S34 by the beam former 30 and the beam forming operation S43 by the echo canceller 40 Echo component removal (S35, S36), and speaker selection (S37) by the switch 60 are sequentially performed.

That is, the beam former 30 processes a beamforming signal corresponding to the position of the speaker with respect to the acoustic signal obtained for each microphone, and generates a sound signal for collecting the speaker's voice (S34).

Next, the echo canceller 40 applies a filter coefficient corresponding to the position of the speaker in the filter coefficient for each beam forming direction stored in advance (S35), and removes the echo component mixed in the acoustic signal (S36).

Next, the switch 60 selects the output direction corresponding to the speaker's position in the sound source output direction for each sound source direction set in advance (S37), and outputs the sound signal through the speaker in the selected output direction (S38) .

By performing the operation step S30 as described above, when the speaker exists, the position of the speaker is directed as a directional microphone, suppressing the echo component according to the direction of the sound signal collected, and then outputting the selected sound signal to the speaker .

On the other hand, a plurality of people may speak simultaneously.

For example, there may be situations in which one person is speaking while another person is speaking. In this case, while one person keeps talking, the person is continuously detected by correlation analysis of the sound source tracker 20, While the voice continues to be amplified and output, and the other person begins to speak, the result of correlation analysis in which two persons are detected from the moment appears.

Accordingly, the beam former 30 generates an acoustic signal that is beam-shaped toward the two talkers, respectively, and the echo canceller 40 suppresses the echo components for the generated acoustic signals in accordance with the beam-forming direction. Then, the switch 60 outputs the corresponding acoustic signal to the speaker in the output direction corresponding to each speaker. At this time, there may be a situation where two persons are simultaneously outputting a voice to one speaker.

As a result, when the voice of a plurality of persons is amplified, the signal processing speed is delayed when the signal is processed by one beam former 30 and one echo canceller 40, so that the difference between the vocalization time and the speaker output time is largely auditory .

If possible, a plurality of signal processing blocks constituted by the beam former 30 and the echo canceller 40 are provided, and when a plurality of speakers are detected by the sound source tracker 20, a plurality of signal processing means are allocated for each speaker position So that the signal processing is performed. Then, it is preferable that the acoustic signals processed by the respective signal processing blocks are individually transmitted to the switch 60, and the switch 60 selects the speakers in accordance with the output direction of each acoustic signal.

The audio amplifying apparatus according to the present invention may be configured such that the speakers S1, S2, S3, and S4 are not provided in the case A but are separated and connected by a cable as shown in FIG. 9 .

In this case, the user can determine the position and the output direction of each speaker when the speaker is placed, and the sound output direction can be determined individually. For example, as shown in FIG. 9, When seated, each speaker output direction can be adjusted in a direction in which people can hear sound.

The cable used to connect the speakers S1, S2, S3, and S4 to the case A is directed mainly toward a position to transmit the output sound. When the speaker is excessively long, it is inconvenient to use. It is preferable to set it to be within 1 m.

It is also possible to constitute a detachable structure in which each of the speakers S1, S2, S3 and S4 is hung on the case A such that the output direction thereof is radially directed, A) in the vicinity of each other.

As another embodiment, it is also possible that the respective speakers are mounted on the case A so as to be rotatable so that the output direction can be adjusted.

While the present invention has been particularly shown and described with reference to exemplary embodiments thereof, it is to be understood that the invention is not limited to the disclosed exemplary embodiments, . ≪ / RTI > Accordingly, such modifications are deemed to be within the scope of the present invention, and the scope of the present invention should be determined by the following claims.

A: Case
S1, S2, S3, S4: Speaker
10: Microphone array M1, M2, M3, M4: Microphone
20: Sound tracker
21: Reverberation rejection by microphone 21a: Reverberation filter
22: correlation analysis section
30: Beamformer
40: echo canceller
41: Echo canceler 41a: Echo canceler
42: Filter coefficient storage unit for each region
50: echo processing processor
51: training signal generating unit 52: echo channel modeling unit
60: Switch
70: control unit 71: communication module

Claims (9)

delete delete delete A plurality of microphone units and a plurality of microphone units may be arranged so that sound signals obtained from the microphones are amplified and output, and a plurality of speakers are arranged in a circumferential direction around the microphone, In addition,
An echo learning processor for modeling the echo channel on the space where the speaker and the microphone are located by using a filter to obtain a filter coefficient; And
An echo canceller for suppressing an echo component due to an echo channel and outputting the echo signal to the speaker by applying a filter coefficient to an acoustic signal obtained from the microphone;
Respectively,
A sound source tracker for tracking the direction of a sound source by correlating and analyzing acoustic signals per microphone; And
A switch for selecting a speaker in an output direction corresponding to a sound source direction by a sound source tracker and outputting an acoustic signal suppressing echo components based on sound output directions of the sound source directions which are predetermined for each sound source direction in advance;
And an audio amplifier for amplifying the audio signal. 5. The method of claim 4,
Wherein the echo learning processor models each of the echo channels formed by each microphone as a filter to obtain a microphone-specific filter coefficient,
Wherein the sound source tracker tracks the direction of the sound source by correlating and analyzing the echo components separately for each acoustic signal by applying a filter coefficient for each microphone. The method according to claim 4 or 5,
Further comprising a beam former for processing an acoustic signal acquired from at least two microphones to acquire a beam-shaped acoustic signal in a sound source direction,
Wherein the echo canceler obtains a sound signal to be output to a speaker by suppressing echo components of the beam-formed acoustic signal. The method according to claim 6,
Wherein the echo learning processor is configured to model an echo channel generated in each beam forming direction by varying a beam forming direction in advance and obtain a filter coefficient for each beam forming direction,
Wherein the echo canceller suppresses an echo component based on a filter coefficient corresponding to a beam forming direction with respect to a beamformed acoustic signal. 8. The method of claim 7,
And a communication module for short-range wireless communication with the smartphone,
Wherein the execution of the echo learning processor for acquiring the filter coefficient for the echo channel and the selection of the acoustic output direction for each sound source direction are performed by a smartphone. 5. The method of claim 4,
Wherein the switch selects a speaker disposed at least in a direction opposite to a sound source direction, and outputs a sound signal.

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