WO2020151133A1

WO2020151133A1 - Sound acquisition system having distributed microphone array, and method

Info

Publication number: WO2020151133A1
Application number: PCT/CN2019/086768
Authority: WO
Inventors: 郑能恒; 丁俊豪; 文衍晖
Original assignee: 深圳大学
Priority date: 2019-01-25
Filing date: 2019-05-14
Publication date: 2020-07-30
Also published as: CN109788382A

Abstract

Disclosed are a sound acquisition system having a distributed microphone array, and a method, applicable for acquiring sound to control a device, and solving the existing problems of poor sound acquisition of voice assistants and resource wastage caused by multiple devices each being bound to microphone sound acquisition modules and voice assistants. The system comprises: a distributed microphone array used to collect sound when indoors; a sound source location processing module used to locate the position of a sound source produced by a person speaking; a voice processing module used to perform processing according to the position of the sound source and the sound collected by the distributed microphone array; and a microphone data acquisition module used to acquire a signal transmitted by the distributed microphone array and synchronize the signal transmitted by the distributed microphone array to a data acquisition terminal. The system provides superior sound acquisition, and enables voice control of multiple apparatuses without the need to fit and install an independent voice acquisition module and voice assistant to each apparatus, thereby reducing resource wastage.

Description

Distributed microphone array pickup system and method Technical field

[0001] The present invention relates to the field of speech recognition technology, and more particularly to a distributed microphone array pickup system and method.

Background technique

[0002] With the development of artificial intelligence technology, the technology of controlling equipment through human-computer interaction is becoming more and more common. In human-computer interaction technology, the technology of controlling equipment through voice has been widely used. In application scenarios such as smart homes, voice assistants have almost become a standard configuration for future development.

[0003] However, it is not easy for a device to have human-like hearing capabilities. In addition to the back-end voice recognition, semantic understanding and many other links, the first step is to allow the device to obtain clear voice signals, that is, to make the device hear better. clear.

[0004] 5 See some devices equipped with voice assistants, the installation location is relatively limited to only one point, the sound pickup effect is poor for longer distances, and the existing devices are all bound with the voice assistant, when in a room When multiple devices are placed, multiple voice assistants need to be configured, which causes a waste of resources.

Summary of the invention

technical problem

[0005] The main purpose of the present invention is to provide a distributed microphone array pickup system and method, which aims to solve the problem of poor pickup effect of voice assistants in 5 existing technologies, and the devices are bound to the voice assistant. The technical problem of waste of resources.

The solution to the problem

Technical solutions

[0006] In order to achieve the above objective, the first aspect of the present invention provides a distributed microphone array pickup system, including: a distributed microphone array for collecting sound information indoors; a sound source location processing module for locating speakers The position of the sound source emitting the sound information; a voice processing module for processing according to the sound source position and the sound information collected by the distributed microphone array to obtain a clean and smooth voice signal; a microphone data collection module for collecting the distributed microphone The array is transmitted after being processed by the voice processing module And synchronize the signal emitted by the distributed microphone array to a data acquisition terminal, and the data acquisition terminal controls the device through the signal emitted by the distributed microphone array.

[0007] Further, the distributed microphone array includes microphones installed on the indoor roof and four walls; by adjusting the positions of the microphones, the microphones on the walls and the microphones on the roof are connected to each other. In space, it appears in the shape of a quadrangular pyramid.

[0008] Further, the sound source localization processing module includes: a microphone coordinate determination unit, configured to set the three-dimensional coordinates of one of the reference microphones in the distributed microphone array, and set the three-dimensional coordinates according to the set three-dimensional coordinates The three-dimensional coordinate system and the three-dimensional coordinates of other microphones in the distributed microphone array; a time delay estimation unit for estimating the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signals received by the microphones; sound source position coordinates The determining unit is configured to calculate the coordinates of the sound source position according to the time, delay, sound velocity between the sound source reaching each microphone, and the coordinates of each microphone.

[0009] Further, the voice processing module includes: a noise reduction unit, configured to reduce noise on the sound information collected by the distributed microphone array to obtain a clean voice signal; and a weighted average unit, configured to combine each microphone The signals are delayed and aligned, and then weighted and averaged to obtain a clean and smooth speech signal

[0010] Further, the noise reduction unit includes: a main microphone signal setting subunit, which is used to set the signal of the main microphone to be composed of a human voice signal and a noise signal; a microphone determination subunit, which is used to determine the sound source The distance to each microphone determines that the microphone closest to the sound source is the main microphone, and the remaining microphones are secondary microphones, and the microphone farthest from the sound source is the noise signal reference microphone; the noise estimation subunit is used to use the noise signal reference microphone The initial silent signal is used as a noise estimate, and the noise estimate is used as the noise signal of the distributed microphone array; a power spectrum calculation subunit, configured to set the subunit and the noise estimation subunit according to the main microphone signal Calculate the power spectrum of the main microphone signal; a factor introduction subunit, used to introduce an over-subtraction factor and a spectrum lower limit compensation factor into the power spectrum calculation subunit; an inverse Fourier transform subunit, used to pass Fourier The inverse transform transforms the signal of the power spectrum calculation subunit that introduces the factor introduction unit into a time domain signal, thereby obtaining a clean speech signal.

[0011] Further, the weighted average unit includes: a time delay estimation subunit, configured to estimate the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signal received by Kefeng; The sub-unit is used to delay and align the voice signals received by all microphones according to the delay time of each signal, so as to perform a weighted average of each voice signal that has undergone noise reduction processing according to the distance from the sound source position to obtain a clean Smooth voice signal.

[0012] Further, the microphone data collection module includes: a data collection unit, configured to collect sound information collected by all microphones in the distributed microphone array and processed by the voice processing module; a data sending unit, configured to transfer the The sound information collected by the data collection unit is sent to the background data collector; the equipment control unit is used to connect to the equipment and control the equipment according to the sound information received by the data collector.

[0013] The second aspect of the present invention provides a distributed microphone array pickup method, including: collecting sound information indoors through the distributed microphone array; locating the speaker's sound source position; collecting according to the sound source position and the distributed microphone array Process the sound information of the distributed microphone array to obtain a clean and smooth voice signal; collect the signals emitted by the distributed microphone array, and synchronize the signals emitted by each microphone in the distributed microphone array to a data collection terminal, and the data collection terminal passes all The signal control equipment emitted by the distributed microphone array.

[0014] A third aspect of the present invention provides an electronic device, including: a memory, a processor, and a computer program stored on the memory and capable of running on the processor, wherein the processor executes all When the computer program is described, the above method is realized.

[0015] A fourth aspect of the present invention provides a computer-readable storage medium on which a computer program is stored, which is characterized in that, when the computer program is executed by a processor, the foregoing method is implemented.

The beneficial effects of the invention

Beneficial effect

[0016] The present invention provides a distributed microphone array pickup system and method. The beneficial effects are: Array layout of multiple microphones indoors through a distributed structure, so that there are multiple locations in the room that can collect voice information, thereby It eliminates the need for users to go to a specific device for voice interaction and emit sound information, thereby improving the sound pickup effect; and binding multiple devices with the microphone data collection module, that is, combining a distributed microphone array After the transmitted signal is synchronized to a data collection terminal, the system can control multiple devices through voice information without having to install an independent pickup module and voice assistant for each device, thereby reducing waste of resources.

Brief description of the drawings Description of the drawings

[0017] In order to explain the embodiments of the present invention or the technical solutions in the prior art more clearly, the following will briefly introduce the accompanying drawings needed in the technical description of the embodiment or 5, and it is obvious that in the following description The drawings are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained from these drawings without creative work.

[0018] FIG. 1 is a schematic block diagram of the structure of a distributed microphone array pickup system according to an embodiment of the present invention;

[0019] FIG. 2 is a schematic block diagram of the structure of a sound source localization processing module of a distributed microphone array sound pickup system according to an embodiment of the present invention;

[0020] FIG. 3 is a schematic block diagram of the structure of a voice processing module of a distributed microphone array pickup system according to an embodiment of the present invention;

[0021] FIG. 4 is a schematic block diagram of a structure of a noise reduction unit of a distributed microphone array sound pickup system according to an embodiment of the present invention;

[0022] FIG. 5 is a schematic block diagram of the structure of an electronic device according to an embodiment of the present invention.

Invention embodiment

Embodiments of the invention

[0023] In order to make the objectives, features, and advantages of the present invention more obvious and understandable, the technical solutions in the embodiments of the present invention will be described clearly and completely in conjunction with the accompanying drawings in the embodiments of the present invention. Obviously The described embodiments are only a part of the embodiments of the present invention, rather than all the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative work shall fall within the protection scope of the present invention.

[0024] Please refer to FIG. 1, which is a distributed microphone array pickup system, including: a distributed microphone array 1, used to collect sound information indoors; and a sound source localization processing module 2, used to locate the speaker emitting sound information The position of the sound source; the voice processing module 3 is used to process the sound information collected by the sound source position and the distributed microphone array 1 to obtain a clean and smooth voice signal; the microphone data collection module 4 is used to collect the distributed microphone array 1 The transmitted signal is processed by the voice processing module 3, and the signal transmitted by the distributed microphone array 1 is synchronized to a data acquisition terminal, and the data acquisition terminal controls the equipment through the signal transmitted by the distributed microphone array 1.

[0025] The distributed microphone array 1 includes microphones installed on the indoor roof and four walls; by adjusting the position of each microphone, the microphones on the walls and the microphones on the roof are connected to each other. The line takes the shape of a quadrangular pyramid in space.

[0026] Please refer to FIG. 2, the sound source localization processing module 2 includes: a microphone coordinate determination unit 21, configured to set the three-dimensional coordinates of one of the reference microphones in the distributed microphone array 1, and set the three-dimensional coordinates according to the set three-dimensional coordinates The coordinate system and the three-dimensional coordinates of other microphones in the distributed microphone array 1; The time delay estimation unit 22 is used to estimate the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signals received by the microphones; The sound source position coordinates are determined The unit 23 is configured to calculate the coordinates of the sound source position according to the time, delay, sound speed between the sound source reaching each microphone, and the coordinates of each microphone.

[0027] Specifically, the dimensions of a rectangular room along the x-axis, y-axis, and z-axis directions in the three-dimensional space coordinate system are set as a, b, and c, respectively, and a microphone is formed on each of the four walls and ceiling of the room to form a distribution Microphone array 1, set the coordinates of one of the microphones as

fru yu %)

, Marked as the first microphone, the coordinates of the other microphones are

Among them, m=2, 3,4, 5, set the sound source position of the speaker's sound information

, And set the time required for the sound signal to reach each microphone from the sound source position as

, Set the speed of sound to V

, The distance from the sound source position to each microphone is

4 , You can get:

[0028]

[0029] Therefore, the time delay between the sound signal reaching the mth microphone of the distributed microphone array and the first microphone from the sound source position can be calculated as:

wt m

[0041] Formula 3 is obtained by subtracting Formula 1 from Formula 2, and Formula 3 is expressed as follows: [0042]

(h ^: m-vision. wi).

m: 2

[0043] Since the position of the sound source and the coordinates of the microphone are respectively

m v v%

Jwt (tn J

, The distance from the sound source position to the m-th microphone can be expressed as:

[0046] Similarly, the distance from the sound source position to the first microphone can be expressed as:

[0048] So use

[0050] Then, formula 4 is substituted into formula 3 to obtain formula 5, which is expressed as follows:

[0051]

[0052] At this time, the four variables are set as

Mm a tn

/ m

And i m

, And set the expression of four variables as follows:

[0059] For the parameters in formula 6

It can be calculated by the time delay estimation (TDOA) method based on the generalized cross-correlation (GCC-PHAT) algorithm. Since the system provided by this application uses a distributed microphone array 1, the distance between the microphones is relatively large, and the spatial resolution of the array signal is very high. The delay estimated by the GCC-PHAT algorithm

7 ^'

f TO

It is also more accurate. Therefore, when m=3,4,5, formula 6 is used to form a ternary linear equation system and expressed as follows: [0062] Referring to FIGS. 3 and 4, the speech processing module 3 includes: a noise reduction unit 31, configured to reduce the noise of the sound information collected by the distributed microphone array IJ1 to obtain a clean speech signal; a weighted average unit 32, It is used to align the delay of each microphone signal, and then perform a weighted average on it to obtain a clean and smooth voice signal.

[0063] The noise reduction unit 31 includes: a main microphone signal setting sub-unit 311, which is used to set the signal of the main microphone to be composed of human voice signals and noise signals; a microphone determining sub-unit 312, which is used to arrive at various The distance of the microphone determines that the microphone closest to the sound source is the main microphone, and the remaining microphones are secondary microphones, and the microphone furthest from the sound source is determined to be the noise signal reference microphone; noise estimation subunit 3 13 is used to use the noise signal reference microphone The initial silent signal is used as the noise estimate, and the noise estimate is used as the noise signal of the distributed microphone array; the power spectrum calculation subunit 314 is configured to calculate the signal of each microphone signal according to the main microphone signal setting subunit 311 and the noise estimation subunit 313 Power spectrum; factor introduction subunit 315, used to introduce over-subtraction factor and spectrum lower limit compensation factor in power spectrum calculation subunit 314; inverse Fourier transform subunit 316, used to introduce factors through inverse Fourier transform The signal of the power spectrum calculation subunit 314 of the lead-in unit is transformed into a time domain signal to obtain a clean speech signal.

[0064] After the coordinates of the sound source position are obtained, the distance between the sound source position and each microphone is calculated

, Determine the microphone closest to the sound source as the main microphone, and other microphones as the secondary microphones, and use the microphone furthest from the sound source as the noise signal reference microphone.

[0065] Specifically, the main microphone signal is set

, among them

X(11)

Is the target voice signal,

¥(11〕

Noise signal, let the noise signal refer to the initial silent signal of the microphone as . Since 5 microphones work at the same time, the noise signal received by all microphones is the same. Use the noise signal to refer to the initial silent section of the microphone as the noise estimation, so

y(n):: x(n)! ¥ (ii)

, The corresponding power spectrum expression is:

[0068] In the process of using spectral subtraction to obtain the power spectrum of the speech signal, the power spectrum may be negative due to oversubtraction, so an oversubtraction factor is introduced

[0070] Since human hearing is not sensitive to the phase of the speech, the amplitude spectrum of the speech signal is obtained through spectral subtraction and noise reduction.

It can be combined with the phase spectrum of the original signal received by the microphone to estimate the frequency spectrum of the voice signal, and then the clean voice signal can be restored by inverse Fourier transform

t(ll)

[0071] The weighted average unit 32 includes: a time delay estimation subunit 321, configured to estimate the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signal received by the microphone; and a delay alignment subunit 322, configured to Delay alignment of the voice signals received by all microphones according to the delay time of each signal In this way, the weighted average of the voice signals that have undergone noise reduction processing is performed according to the distance from the sound source position to obtain a clean and smooth voice signal.

[0072] After the above-mentioned noise reduction processing, each microphone channel can obtain a relatively clean voice signal. In order to enhance the voice signal in the direction of the speaker, it is assumed that the position of the speaker does not change in a short period of time. Delay, align the five microphone signals with delay, and then perform a weighted average according to their respective distances from the speaker to obtain a clean and smooth voice signal.

[0073]

[0074] The microphone data collection module 4 includes: a data collection unit 41 for collecting sound signals collected by all microphones in the distributed microphone array 1 and processed by the voice processing module 3; a data sending unit 42 for sending the data collection unit 41 The collected sound information is sent to the background data collector; the device control unit 43 is used to connect to the device and control the device according to the sound information received by the data collector.

[0075] In this embodiment, the distributed microphone array 1 is connected to the data collection unit 41 through a wireless network, and after the voice signal received by the microphone is processed by the voice processing module 3, a clean and smooth voice signal is obtained, and then a clean and smooth voice signal is obtained through the wireless network. The voice signal is transmitted to the data collection unit 41, and the data sending unit 42 transmits the voice signal received by the data collection unit 41 to the background data collector. After the device control unit 43 is connected to the device, it can be controlled by the voice signal received by the data collector equipment.

[0076] By connecting the data collection unit 41 and the distributed microphone array 1 through a wireless network, it is not only convenient for networking, but also increased flexibility in smart home voice control. At the same time, a unified device control unit 43 is used as a central processing platform. A common hardware input interface can be provided for each intelligent voice interaction device, so that the system can control multiple devices through voice information without having to install a pickup module and voice assistant for each device, which reduces the waste of resources.

[0077] The present application also provides a method for collecting sound from a distributed microphone array, including: collecting sound information indoors through the distributed microphone array; locating the speaker's sound source position; according to the sound source position and the sound collected by the distributed microphone array The information is processed to obtain a clean and smooth voice signal; collect the signal emitted by the distributed microphone array, and synchronize the signal emitted by the distributed microphone array microphone to a data acquisition terminal, and the data acquisition terminal controls the equipment through the signal emitted by the distributed microphone array . [0078] The distributed microphone array includes microphones arranged on the indoor roof and four walls; by adjusting the position of each microphone, the microphones installed on the walls and the microphones installed on the roof are connected to each other and present a quadrangular pyramid shape in space. .

[0079] Locating the speaker's sound source position includes: setting the three-dimensional coordinates of one of the reference microphones in the distributed microphone array, and setting the three-dimensional coordinate system and the three-dimensional coordinates of other microphones in the distributed microphone array according to the set three-dimensional coordinates ; Estimate the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signal received by the microphone; calculate the coordinates of the sound source position according to the time, delay, sound velocity and the coordinates of each microphone between the sound source reaching each microphone.

[0080] Processing according to the sound source position and the sound information collected by the distributed microphone array to obtain a clean and smooth voice signal includes: denoising the sound information collected by the distributed microphone array to obtain a clean voice signal; The voice signals of each microphone in the array undergo delay alignment and weighted average to obtain a clean and smooth voice signal.

[0081] Denoising the sound information collected by the distributed microphone array to obtain a clean voice signal includes: setting the signal of the main microphone to be composed of a voice signal and a noise signal emitted by a person; and determining the distance according to the distance from the sound source to each microphone The microphone with the closest sound source is the main microphone, and the remaining microphones are secondary microphones. The microphone farthest from the sound source is determined as the noise signal reference microphone; the initial silent signal of the noise signal reference microphone is used as the noise estimate, and the noise estimate is used as the distribution The noise signal of a microphone array; calculate the power spectrum of the main microphone signal according to the main microphone signal setting sub-unit and the noise estimation sub-unit; In the process of calculating the power spectrum of the main microphone signal, an over-subtraction factor and a spectrum lower limit compensation factor are introduced; The inverse inner transform transforms the signal of the power spectrum calculation subunit that introduces the factor introduction unit into a time domain signal, thereby obtaining a clean speech signal.

[0082] Processing the voice signal of each microphone in the distributed microphone array to obtain a clean and smooth voice signal includes: estimating the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signal received by Kefeng; The delay time of each channel of signals aligns the delays of the voice signals received by all microphones, so that the voice signals of each channel that have undergone noise reduction processing are weighted and averaged according to the distance from the sound source position to obtain a clean and smooth voice signal.

[0083] Collect the signals emitted by the distributed microphone array, and synchronize the signals emitted by the distributed microphone array microphones to a data collection terminal, and the data collection terminal is controlled by the signals emitted by the distributed microphone array. The control equipment includes: Collecting the sound information collected by all microphones in the distributed microphone array and processed by the voice processing module; Sending the collected sound information to the background data collector after processing; Connecting the data collector to the equipment and collecting data according to The sound information received by the receiver controls the device.

[0084] An embodiment of the present application provides an electronic device. Please refer to 5. The electronic device includes a memory 601, a processor 602, and a computer program stored in the memory 601 and running on the processor 602. The processor 602 When the computer program is executed, the distributed microphone array pickup method described above is realized.

[0085] Further, the electronic device further includes: at least one input device 603 and at least one output device 604.

[0086] The aforementioned memory 601, processor 602, input device 603, and output device 604 are connected through a bus 605.

[0087] The input device 603 may specifically be a camera, a touch panel, a physical button or a mouse, etc. The output device 604 may specifically be a display screen.

[0088] The memory 601 may be a high-speed random access memory (RAM, Random Access Memory) memory, or may be a non-volatile memory (non-volatile memory), such as a disk memory. The memory 601 is used to store a group of executable program codes, and the processor 602 is coupled with the memory 601.

[0089] Further, an embodiment of the present application also provides a computer-readable storage medium. The computer-readable storage medium may be the electronic device provided in the foregoing embodiments, and the computer-readable storage medium may be the aforementioned的Memory 601. A computer program is stored on the computer-readable storage medium, and when the program is executed by the processor 602, the distributed microphone array pickup method described in the foregoing method embodiment is implemented.

[0090] Further, the computer storage medium may also be a U disk, a mobile hard disk, a read-only memory 601 (ROM, Read-Only Memory), RAM, a magnetic disk, or an optical disk, and other media that can store program codes.

[0091] In the several embodiments provided in this application, it should be understood that the disclosed apparatus and method may be implemented in other ways. For example, the device embodiments described above are merely illustrative, for example, the division of the modules is only a logical function division, and there may be other division methods in actual implementation, for example, multiple modules or components may be combined or It can be integrated into another system, or some features can be ignored or not implemented. On the other hand, the displayed or discussed mutual coupling or direct coupling or communication connection may be indirect coupling or communication connection through some interfaces, devices or modules, and may be electrical , Mechanical or other forms.

[0092] The modules described as separate components may or may not be physically separated, and the components displayed as modules may or may not be physical modules, that is, they may be located in one place, or they may be distributed to multiple networks On the module. Some or all of the modules can be selected according to actual needs to achieve the objectives of the solution of the embodiment.

[0093] In addition, the functional modules in the various embodiments of the present invention may be integrated into one processing module, or each module may exist alone physically, or two or more modules may be integrated into one module. The above-mentioned integrated modules can be implemented in the form of hardware or software functional modules.

[0094] If the integrated module is implemented in the form of a software function module and sold or used as an independent product, it can be stored in a computer readable storage medium. Based on this understanding, the technical solution of the present invention essentially or the part that contributes to the prior art or all or part of the technical solution can be embodied in the form of a software product, and the computer software product is stored in a storage medium. , Including several instructions to enable a computer device (which may be a personal computer, a server, or a network device, etc.) to perform all or part of the steps of the methods described in the various embodiments of the present invention.

[0095] It should be noted that for the foregoing method embodiments, for simplicity of description, they are all expressed as a series of action combinations, but those skilled in the art should know that the present invention is not subject to the described sequence of actions. The limitation is because according to the present invention, some steps can be performed in other order or simultaneously. Secondly, those skilled in the art should also know that the embodiments described in the specification are all preferred embodiments, and the actions and modules involved are not necessarily all required by the present invention.

[0096] In the foregoing embodiments, the description of each embodiment has its own focus, and for a part that is not described in detail in an embodiment, reference may be made to related descriptions of other embodiments.

[0097] The above is a description of a distributed microphone array pickup system and method provided by the present invention. For those skilled in the art, according to the ideas of the embodiments of the present invention, there will be specific implementations and application scopes. In summary, the content of this specification should not be construed as limiting the present invention.

Claims

[Claim 1] A distributed microphone array sound pickup system, characterized in that it comprises:

Distributed microphone array, used to collect sound information indoors; Sound source localization processing module, used to locate the sound source position of the speaker's sound information; Voice processing module, used to collect sound according to the sound source position and the distributed microphone array The information is processed to obtain a clean and smooth voice signal;

The microphone data collection module is used to collect the signals emitted by the distributed microphone array after being processed by the voice processing module, and synchronize the signals emitted by the distributed microphone array to a data collection terminal, and the data collection terminal passes through the distributed Signal control equipment emitted by the microphone array.

[Claim 2] The distributed microphone array pickup system according to claim 1, wherein the distributed microphone array includes microphones installed on the indoor roof and four walls;

By adjusting the position of each microphone, the interconnection between the microphones set on the wall and the microphones set on the roof presents a quadrangular pyramid shape in space.

[Claim 3] The distributed microphone array sound pickup system of claim 1, wherein the sound source localization processing module comprises:

The microphone coordinate determination unit is used to set the three-dimensional coordinates of one of the reference microphones in the distributed microphone array, and to set the three-dimensional coordinate system and the three-dimensional coordinates of other microphones in the distributed microphone array according to the set three-dimensional coordinates. Coordinates; a time delay estimation unit for estimating the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signal received by the microphone;

The sound source position coordinate determination unit is used to calculate the coordinates of the sound source position according to the time, delay, sound speed between the sound source reaching each microphone, and the coordinates of each microphone.

[Claim 4] The distributed microphone array sound pickup system of claim 1, wherein the voice processing module comprises:

A noise reduction unit, configured to reduce noise on the sound information collected by the distributed microphone array to obtain a clean voice signal; The weighted average unit is used to align the delays of the microphone signals of each channel, and then perform a weighted average on them to obtain a clean and smooth voice signal.

[Claim 5] The distributed microphone array sound pickup system of claim 4, wherein the noise reduction unit comprises:

The main microphone signal setting sub-unit, which is used to set the signal of the main microphone to consist of human voice signals and noise signals;

The microphone determining subunit is used to determine the microphone closest to the sound source as the main microphone according to the distance from the sound source to each microphone, and the other microphones as secondary microphones, and to determine the microphone furthest from the sound source as the noise signal reference microphone;

A noise estimation subunit, configured to use a noise signal reference microphone's initial silent signal as a noise estimation, and use the noise estimation as a noise signal of the distributed microphone array;

The power spectrum calculation subunit is used to calculate the power spectrum of each microphone signal according to the main microphone signal setting subunit and the noise estimation subunit; the factor introduction subunit is used for the power spectrum calculation subunit Introduce over-subtraction factor and spectrum lower limit compensation factor;

The inverse Fourier transform subunit is used to transform the signal introduced into the power spectrum calculation subunit of the factor introduction unit into a time domain signal through an inverse Fourier transform, thereby obtaining a clean voice signal.

[Claim 6] The distributed microphone array pickup system according to claim 5, wherein the weighted average unit comprises:

The time delay estimation subunit is used to estimate the time delay of each microphone relative to the reference microphone according to the cross-correlation function of the signal received by Kefeng;

The delay alignment subunit is used to delay alignment of the voice signals received by all microphones according to the delay time of each signal, so as to perform a weighted average of the voice signals that have undergone noise reduction processing according to the distance from the sound source position. , Get a clean and smooth voice signal.

[Claim 7] The distributed microphone array pickup system according to claim 1, wherein the microphone data collection module comprises: A data collection unit, configured to collect sound signals collected by all microphones in the distributed microphone array and processed by a voice processing module;

A data sending unit, configured to send the sound information collected by the data collection unit to the background data collector;

The device control unit is used to connect the device and control the device according to the sound information received by the data collector.

[Claim 8] A method for collecting sound from a distributed microphone array, characterized in that it comprises:

Collect sound information indoors through a distributed microphone array;

Locate the speaker's sound source position;

Perform processing according to the sound source location and the sound information collected by the distributed microphone array to obtain a clean and smooth voice signal;

Collect the signals emitted by the distributed microphone array, and synchronize the signals emitted by the microphones in the distributed microphone array to a data collection terminal, and the data collection terminal controls the equipment through the signals emitted by the distributed microphone array.

[Claim 9] An electronic device, comprising: a memory, a processor, and a computer program stored in the memory and capable of running on the processor, wherein the processor executes the computer program when , To implement the method in claim 8.

[Claim 10] A computer-readable storage medium having a computer program stored thereon, wherein the computer program is executed by a processor to implement the method in claim 8.