Sound pickup system with adjustable beam directivity, method, equipment and medium
By adopting a linear super microphone array composed of omnidirectional and directivity microphones in the pickup device, and using audio encoding, decoding and digital signal processing technology, beam directionality is realized, solving the problem of pickup device picking up in the far field in the prior art The problem of unadjustable sound and beam directionality improves audio monitoring quality and customer satisfaction.
Patent Information
- Application Number
- CN202411532892.X
- Authority / Receiving Office
- CN · China
- Patent Type
- Applications(China)
- Current Assignee / Owner
- Filing Date
- 2024-10-30
- Publication Date
- 2025-02-07
AI Technical Summary
Existing pickup equipment cannot collect spatial orientation information of the sound field in far-field pickup, and the beam direction is unadjustable, resulting in poor sound acquisition effect in non-end shooting directions and cannot meet the audio monitoring quality and customer needs.
A linear super microphone array consisting of an omnidirectional pickup electret microphone and a directivity pickup electret microphone is used to perform signal processing through audio codec chips and digital signal processing DSP chips, so as to achieve adjustable beam direction.
The beam direction is adjustable, and the frequency beam response can be maintained within a specific frequency response range, maximize spatial direction, improve the ability to extract target sound signals in noisy acoustic environments, and meet audio monitoring quality and customer needs. .
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Figure CN119402803A_ABST
Abstract
Description
Technical Field
[0001] The invention belongs to the field of microphone equipment, and relates to a microphone system, method, equipment and medium with adjustable beam directivity. Background Art
[0002] With the development of society, the role of audio monitoring as a part of security monitoring is increasingly enhanced, especially in some blind spots of video monitoring and areas where it is inconvenient to install cameras. As the core equipment of audio monitoring, the performance of the microphone is directly related to the quality of audio monitoring. At present, the mainstream microphone devices on the market are mainly composed of a single microphone or multiple microphones. Among them, the microphone devices composed of a single microphone have inherent deficiencies in far-field sound pickup, and cannot collect the spatial orientation information of the sound field, and cannot accurately measure the location information of the sound source; the array-shaped products composed of multiple microphones are mainly presented in the form of uniform linear arrays in the field of security monitoring. This type of microphone device has inherent advantages in the end-fire direction, but has obvious deficiencies in other directions, that is, the beam directivity cannot be adjusted and can only be limited to a specific direction area. If it is necessary to obtain sound information in other directions, it is necessary to change the installation position of the microphone device, which is time-consuming, labor-intensive and unreliable. In addition, the microphone devices currently available on the market, whether composed of a single microphone or multiple microphones, all use the same type of microphone devices for pickup. Especially in the microphone devices composed of multiple microphones, most of the microphone devices used are omnidirectional microphone capsules, and very few are directional microphone capsules. The linear array microphones currently available on the market are basically composed of omnidirectional microphone capsule devices. The problem this brings is the existence of axial symmetry. Assuming that the 0° direction is the end-fire direction of the microphone, the pickup and positioning results in the 90° and 270° directions are the same, which is not conducive to beamforming and sound source positioning in the desired direction. Therefore, the microphone devices available on the market cannot guarantee the audio monitoring quality and customer needs. Summary of the invention
[0003] The purpose of the present invention is to overcome the shortcomings of the above-mentioned prior art and provide a microphone system, method, device and medium with adjustable beam directivity, which can ensure the audio monitoring quality and meet customer needs.
[0004] In order to achieve the above object, the present invention adopts the following technical scheme:
[0005] On the one hand, the present invention provides a microphone system with adjustable beam directivity, comprising a front-end sound signal acquisition module, a mid-end sound signal processing module and a back-end sound signal output module, wherein the front-end sound signal acquisition module comprises an audio codec chip, a plurality of omnidirectional sound pickup electret microphones and a plurality of directional sound pickup electret microphones, each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are staggered and distributed in sequence at equal intervals along a straight line, each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are connected to the input end of the audio codec chip, and the mid-end sound signal processing module is connected to the output end of the audio codec chip and the back-end sound signal output module.
[0006] The microphone pickup system with adjustable beam directivity of the present invention is further improved in that:
[0007] Furthermore, the mid-end sound signal processing module includes a digital signal processing DSP chip, and the digital signal processing DSP chip is connected to the output end of the audio codec Codec chip and the back-end sound signal output module.
[0008] Furthermore, the back-end sound signal output module includes an Ethernet chip, a DAC chip and a USB chip, wherein a digital signal processing DSP chip is connected to the Ethernet chip, the DAC chip and the USB chip.
[0009] Further, the DAC chip is connected to the DAC interface pin of the digital signal processing DSP chip;
[0010] The USB chip is connected to the USB interface pin of the digital signal processing DSP chip;
[0011] The Ethernet chip is connected to the Ethernet corresponding interface pin of the digital signal processing DSP chip.
[0012] In a second aspect of the present invention, the present invention provides a sound pickup method with adjustable beam directivity, comprising the following steps:
[0013] Acquire analog sound signals output by an omnidirectional sound pickup electret microphone and a directional sound pickup electret microphone, and convert the analog sound signals into multi-channel digital sound signals, wherein each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are alternately distributed in sequence at equal intervals along a straight line;
[0014] Preprocessing the multi-channel digital sound signal, filtering the preprocessed multi-channel sound signal through a beamforming filter to obtain processed sound signal data;
[0015] The processed sound signal data and the multi-channel digital sound signal are output.
[0016] The sound pickup method with adjustable beam directivity of the present invention is further improved in that:
[0017] Furthermore, the process of preprocessing the multi-channel digital sound signal by the mid-end sound signal processing module is:
[0018] The multi-channel digital sound signal is processed in sequence by a mid-end sound signal processing module using a direct current filter, an overlapping frame function, a window function filter and a short-time Fourier transform.
[0019] Furthermore, the method of obtaining the analog sound signals output by the omnidirectional sound pickup electret microphone and the directional sound pickup electret microphone through the audio codec chip and converting the analog sound signals into multi-channel digital sound signals also includes:
[0020] Perform data automatic gain adjustment on the multi-channel digital sound signal.
[0021] Furthermore, the parameter solution expression of the beamforming filter is:
[0022] h=B H (BB H ) -1 c(θ s )
[0023] in, The expressions are:
[0024]
[0025] in, The Nth-order optimal beam pattern of a uniform linear array of omnidirectional microphone capsules, is the Nth-order optimal beam pattern of a uniform linear array of directional microphone capsules, θ s The expected direction.
[0026] In a third aspect of the present invention, a computer device is provided, comprising a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the processor implements the steps of the sound pickup method with adjustable beam directivity when executing the computer program.
[0027] In a fourth aspect of the present invention, the present invention provides a computer-readable storage medium, wherein the computer-readable storage medium stores a computer program, and when the computer program is executed by a processor, the steps of the sound pickup method with adjustable beam directivity are implemented.
[0028] The present invention has the following beneficial effects:
[0029] In the specific operation of the microphone system, method, device and medium with adjustable beam directivity described in the present invention, each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are staggered and distributed at equal intervals in a straight line, and the preprocessed multi-channel sound signal is filtered through a beamforming filter, thereby making full use of the acoustic vector (AVS) characteristics of the omnidirectional microphone head and the directional microphone head to achieve adjustable beam directivity; in addition, within a specific frequency response range, the response of the beam can be uniform, that is, the frequency beam response has an invariant characteristic; finally, the spatial directivity can be maximized, which is helpful to extract the target sound signal in a noisy acoustic environment to ensure the audio monitoring quality and customer needs. BRIEF DESCRIPTION OF THE DRAWINGS
[0030] The accompanying drawings constituting a part of the present invention are used to provide a further understanding of the present invention. The exemplary embodiments of the present invention and their descriptions are used to explain the present invention and do not constitute an improper limitation of the present invention. In the accompanying drawings:
[0031] Figure 1 is a system structure diagram of the present invention;
[0032] Figure 2 This is the structural diagram of the front-end sound signal acquisition module;
[0033] Figure 3 This is a structural diagram of the mid-end sound signal processing module;
[0034] Figure 4 This is the structural diagram of the back-end sound signal output module;
[0035] Figure 5 is a schematic diagram of the linear super array plane;
[0036] Figure 6 It is a structural diagram of the present invention;
[0037] Figure 7 The present invention is a flow chart of the method. DETAILED DESCRIPTION
[0038] The following will be combined with the drawings in the embodiments of the present invention to clearly and completely describe the technical solutions in the embodiments of the present invention. Obviously, the described embodiments are part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the present invention.
[0039] In the description of the present invention, it should be understood that the terms “include” and “comprises” indicate the presence of described features, wholes, steps, operations, elements and / or components, but do not exclude the presence or addition of one or more other features, wholes, steps, operations, elements, components and / or collections thereof.
[0040] It should also be understood that the terms used in the present specification are only for the purpose of describing specific embodiments and are not intended to limit the present invention. As used in the present specification and the appended claims, unless the context clearly indicates otherwise, the singular forms "a", "an" and "the" are intended to include plural forms.
[0041] It should be further understood that the term "and / or" used in the present specification and the appended claims refers to any combination of one or more of the associated listed items and all possible combinations, and includes these combinations. For example, A and / or B can represent: A exists alone, A and B exist at the same time, and B exists alone. In addition, the character " / " in the present invention generally indicates that the associated objects are in an "or" relationship.
[0042] It should be understood that, although the terms first, second, third, etc. may be used to describe preset ranges, etc. in the embodiments of the present invention, these preset ranges should not be limited to these terms. These terms are only used to distinguish preset ranges from each other. For example, without departing from the scope of the embodiments of the present invention, the first preset range may also be referred to as the second preset range, and similarly, the second preset range may also be referred to as the first preset range.
[0043] The word "if" as used herein may be interpreted as "at the time of" or "when" or "in response to determining" or "in response to detecting", depending on the context. Similarly, the phrases "if it is determined" or "if (stated condition or event) is detected" may be interpreted as "when it is determined" or "in response to determining" or "when detecting (stated condition or event)" or "in response to detecting (stated condition or event)", depending on the context.
[0044] In order to make the purpose, technical solutions and advantages of the embodiments of the present invention clearer, the technical solutions in the embodiments of the present invention will be clearly and completely described below in conjunction with the drawings in the embodiments of the present invention. Obviously, the described embodiments are part of the embodiments of the present invention, rather than all of the embodiments. The components of the embodiments of the present invention described and shown in the drawings here can usually be arranged and designed in various different configurations. Therefore, the following detailed description of the embodiments of the present invention provided in the drawings is not intended to limit the scope of the claimed invention, but merely represents selected embodiments of the present invention. Based on the embodiments in the present invention, all other embodiments obtained by ordinary technicians in this field without making creative work are within the scope of protection of the present invention.
[0045] Various structural schematic diagrams of the embodiments disclosed in the present invention are shown in the accompanying drawings. These figures are not drawn to scale, and some details are magnified and some details may be omitted for the purpose of clear expression. The shapes of various regions and layers shown in the figures and the relative sizes and positional relationships therebetween are only exemplary, and may deviate in practice due to manufacturing tolerances or technical limitations, and those skilled in the art may additionally design regions / layers with different shapes, sizes, and relative positions according to actual needs.
[0046] Embodiment 1
[0047] refer to Figure 1 and Figure 6 The microphone system with adjustable beam directivity described in the present invention includes a front-end sound signal acquisition module, a mid-end sound signal processing module and a back-end sound signal output module.
[0048] like Figure 2 As shown, the front-end sound signal acquisition module includes an audio codec Codec chip, a plurality of omnidirectional pickup electret microphones and a plurality of directional pickup electret microphones, the omnidirectional pickup electret microphone and the directional pickup electret microphone are different in the pickup mode, and the sensitivity, signal-to-noise ratio, dynamic range, frequency response, equivalent noise level, etc. of the omnidirectional pickup electret microphone and the directional pickup electret microphone are the same; the omnidirectional pickup electret microphone and the directional pickup electret microphone are staggered and distributed in a straight line at equal intervals in sequence, for example, the number of omnidirectional pickup electret microphones is eight, and the number of directional pickup electret microphones is seven, wherein each omnidirectional pickup electret microphone is located at positions 1, 3, 5, 7, 9, 11, 13, and 15 respectively, and each directional pickup electret microphone is located at positions 2, 4, 6, 8, 10, 12, and 14 in sequence, and the omnidirectional pickup electret microphone and the directional pickup electret microphone form a linear super microphone array (LSA).
[0049] The output ends of the omnidirectional sound pickup electret microphones and the directional sound pickup electret microphones are connected to the input end of the audio codec chip. The audio codec chip has multi-channel input and output characteristics.
[0050] The working process of the audio codec chip is: converting the received analog sound signal into a multi-channel digital sound signal; then performing data automatic gain adjustment AGC on the multi-channel digital sound signal to amplify or reduce the multi-channel digital sound signal so that the multi-channel digital sound signal remains consistent in amplitude and ensures that the maximum difference between channels is within 3dB; finally, the processed signal is sent to the digital signal processing DSP chip.
[0051] like Figure 3As shown, the mid-end sound signal processing module includes a digital signal processing DSP chip, which receives the multi-channel digital sound signal sent by the audio codec Codec chip and pre-processes the multi-channel digital sound signal. Specifically, the multi-channel audio serial port (McASP port) multi-channel data transmission, clock synchronization management, programmable control, low latency, low power consumption and other characteristics are used to complete the task of receiving the multi-channel digital sound signal; in the process of receiving data, a multi-channel time division multiplexing (TDM) mode is adopted to ensure the accuracy, efficiency and stability of the multi-channel sound signal during the receiving process; the pre-processing mainly includes: DC filter , overlapping framing function, window function filter, short-time Fourier transform (STFT), wherein the DC filter is used to remove the DC component in the sound signal, so that the signal changes around the zero point, which helps to improve the dynamic range of the signal; the overlapping framing function is used to segment the continuous multi-channel data sound signal according to the preset frame length and overlap rate, which can process the multi-channel data synchronously on the one hand, and enhance the real-time efficiency of the system when it is running on the other hand; the window function filter reduces or controls the undesirable effects in the signal processing process by weighted processing of the multi-channel digital sound signal, such as reducing spectrum leakage, reducing the truncation effect of the sound signal, selectively enhancing the desired frequency response, improving the frequency resolution, etc. The short-time Fourier transform (STFT) is used to transform the framed data from the time domain to the frequency domain. The beamforming algorithm described in the present invention is performed in the frequency domain of the sound signal, and the frequency domain characteristics of the sound signal are used to enhance the desired sound source. In addition, the digital signal processing DSP chip performs parameter setting and program control on its interface pins. For example, the frame synchronization pin (FS), the bit clock pin (BCLK), and the master clock pin (MCLK) are used to control and manage the external audio codec Codec chip to ensure the stability of the system during the synchronous transmission of multi-channel data. Thirdly, the data algorithm processing function. The data algorithm described in the present invention is a directivity adjustable beamforming algorithm in the frequency domain. The functions of the directivity adjustable beamforming algorithm are: 1) making full use of the characteristics of the omnidirectional microphone head and the directional microphone head to achieve adjustable directivity of the beam; 2) the response of the beam can be uniform within a specific frequency response range, that is, the frequency beam response is unchanged; 3) the spatial directivity can be maximized, which is helpful to extract the target desired sound signal in a noisy acoustic environment; finally, the data output function uses the IO port or pin configuration function of the digital signal processing DSP chip, for example, the configuration of the Ethernet chip pin, the DAC chip pin, the USB chip pin, etc., to complete the output function of the sound signal processed by the algorithm and the original multi-channel sound signal data.
[0052] refer to Figure 4The back-end sound signal output module is used to complete the task of outputting the sound signal of the pickup device. The back-end sound signal output module includes an Ethernet chip, a DAC chip and a USB chip. Specifically, the Ethernet chip is connected to the Ethernet corresponding interface pin of the digital signal processing DSP chip, and the processed sound signal data and the original multi-channel data sound signal in the digital signal processing DSP chip are output. The Ethernet interface method can greatly meet the transmission problem in the physical space distance, which is convenient for the application of the present invention in engineering; secondly, the DAC chip is connected to the DAC interface pin of the digital signal processing DSP chip, and the sound signal data processed by the algorithm in the digital signal processing DSP chip is output, and the earphone and other external devices are driven, which can be used to evaluate the sound pickup effect of the pickup in real time, and can also complete the real-time monitoring task in special scenes; finally, the USB chip is connected to the USB interface pin of the digital signal processing DSP chip, and the sound signal data processed by the algorithm in the digital signal processing DSP and the original multi-channel data sound signal are output, and the data is transmitted in USB mode. On the one hand, the USB chip expands the diversity of the device interface in the present invention, which is convenient for the operation of the user, and on the other hand, the USB method can be used in short-distance scene communication, which is more convenient and efficient. The combination of multiple output modes can make the present invention have stronger scene application capabilities and meet the different application demands of different customers.
[0053] In addition, it should be noted that although the traditional beamforming algorithm, especially the linear differential beamforming method, is widely used in small-sized pickup devices, its biggest drawback is that the beam is constrained to a specific directional angle and cannot achieve the steering function, which is greatly limited in practical applications. Therefore, the present invention discloses a directivity-adjustable beamforming algorithm, which combines an omnidirectional microphone head and a directional microphone head to form a linear microphone array, which can simultaneously obtain the measurement values of sound pressure and sound velocity at a single point, that is, generate a new acoustic vector (AVS), which can significantly improve the directivity angle adjustable characteristics of the beam. For ease of description, the linear microphone array composed of the combination of an omnidirectional microphone head and a directional microphone head is called a linear super array (LSA), and the schematic diagram of the linear super array is shown as follows. Figure 5 As shown, assume that the number of omnidirectional microphone heads is M, the spacing between adjacent array elements is δ, the number of directional microphone heads is M′, the spacing between adjacent array elements is δ′, there is no overlap problem in the array element positions of the two uniform linear arrays, and the two uniform linear arrays are coaxial and have the same origin. Assume that the angle measured counterclockwise along the x-axis is θ. Since all array elements of the uniform linear array composed of directional microphone heads have the same directional pattern and have frequency-invariant characteristics, their observation direction is Then we have:
[0054] Α(θ)=a0+a1sinθ
[0055] Among them, the parameter value of a0, a1 = 1-a0 is determined by the type of directional microphone. The scene in which the pickup device of the present invention is used is the acoustic far field, and the distance between the sound source and the array is much larger than the size of the array itself. Therefore, the relationship between the phase delay vector of the linear super array and the measured angle θ, that is, the steering vector of the array is:
[0056]
[0057] Among them, d 1,f (θ) and d 2,f (θ) respectively represent the expression of the steering vector of the two arrays, namely:
[0058]
[0059] Where superscript T represents the transposition operator, j represents the imaginary unit, ω M and ω′ M′ The expressions are:
[0060]
[0061] Where f is the time frequency, c = 340m / s is the speed of sound in the air; the wavelength is In order to avoid spatial aliasing of the spectrum, the present invention requires that the two array element spacings δ and δ′ satisfy a requirement of being less than half of the minimum wavelength.
[0062] Assume that there is a sound source of interest in the far field, and the incident angle between it and the linear super array is denoted as θ s , then the signal received by the LSA array is expressed as:
[0063]
[0064] in
[0065] y1=[Y 1,1 … Y 1,M ] T =d1(θ s )X+v1
[0066] y2=[Y 2,1 … Y 2,M′ ] T =d2(θ s )X+v2
[0067] Among them, y1 and y2 are the signal vectors received by the two uniform linear arrays, X is the sound source signal of interest, and v is the additive noise source. The additive noise source v is expressed as:
[0068]
[0069] in
[0070] v1=[V 1,1 … V 1,M ] T
[0071] v2=[V 2,1 … V 2,M′ ] T
[0072] In order to obtain the desired signal of interest from the noisy environmental noise, a beamforming filter with a length of M+M′ is applied to the signal vector received by the array, and the signal after the filter is:
[0073] Z=h H y=h H d(θ s )X+h H v
[0074] The superscript H represents the conjugate transpose operator. Where h1=[H 1,1 … H 1,M ] T , h2=[H 2,1 … H 2,M′ ] T , so that the incident angle is θ s The desired signal of interest passes through the beamforming filter without distortion, and the beamforming filter needs to satisfy the distortion-free constraint condition: H d(θ s )=1.
[0075] The beam pattern is an important indicator for measuring the quality of beamforming filter design. The beam pattern refers to the spatial response of the beamformer, specifically:
[0076]
[0077] The superscript * represents the conjugate operator, and the beam pattern expression can be understood as the uniform linear array beam pattern composed of omnidirectional microphone heads. Uniform linear array beam pattern composed of directional microphone capsules The sum of in
[0078]
[0079] Using the least squares error criterion and The exponential part of and After optimization, the Jacobi-Anger expansion of the exponential part is obtained as follows:
[0080]
[0081] in, and The truncation order of the Jacobi-Anger expansion of the exponential part is N1 and N2 respectively, then:
[0082]
[0083] will be re-derived and The expression after Jacobi-Anger expansion is:
[0084]
[0085] Expressing the above two expressions in vector form, we get:
[0086]
[0087] Among them, β 1,n =[β n (ω1)…β n (ω M )] T ,
[0088] β 2,n =[β n (ω′1) … β n (ω′ M′ )] T .
[0089] In the desired direction θ s The expression of the optimal beam pattern of the upper N-order differential array with adjustable beam directivity is:
[0090]
[0091] Among them, α n , n=1,2,…N are real parameters, and Using the above formula, we can get the desired direction θ s N-order Optimal Beam Pattern of Upper Linear Superarray in, The Nth-order optimal beam pattern of a uniform linear array of omnidirectional microphone capsules, is the Nth-order optimal beam pattern of a uniform linear array of directional microphone capsules, and The expressions are:
[0092]
[0093] Using Euler's formula, Get it back and The expression is:
[0094]
[0095] Through the derivation and demonstration of the above formula, the linear super array in the desired direction θ is given. s The beamforming filter equation on is:
[0096] Bh=c(θ s )
[0097] in, The expressions are:
[0098]
[0099]
[0100] Finally, the filter parameters of the linear super array directivity adjustable beamformer are obtained as follows:
[0101] h=B H (BB H ) -1 c(θ s ).
[0102] Embodiment 2
[0103] In an embodiment of the present invention, the sampling frequency of the signal is 16000 Hz, the frame length is 4 ms, that is, the corresponding number of points is 64, and the frame overlap rate is 75%, that is, the number of frame overlap points is 48; the windowing function refers to window function filtering of each frame of audio signal, and the window function filter reduces or controls the undesirable effects in the signal processing process by weighting the sound signal, such as reducing spectrum leakage, reducing the truncation effect of the sound signal, selectively enhancing the desired frequency response, improving the frequency resolution, etc. In this embodiment, the window function filter used is a Hamming window, and the length of the window function is 256.
[0104] In this embodiment, the length of the short-time Fourier transform function is 256 points.
[0105] In this embodiment, h=B H (BB H ) -1 c(θ s ) solves the filter parameters and uses them in the digital signal processing DSP chip for each frame of multi-channel sound signal data. The sound signal is enhanced by the beam filter, and the enhanced sound signal data is a single-channel signal in the frequency domain. In this embodiment, the length of the directivity angle variable beamforming filter is 256.
[0106] In this embodiment, the single-channel frequency domain sound signal data processed by the beam filter is processed by the inverse short time Fourier transform (ISTFT) to transform it into a time domain signal. At this time, the length of the obtained signal is 64.
[0107] Embodiment 3
[0108] refer to Figure 7 The sound pickup method with adjustable beam directivity of the present invention comprises the following steps:
[0109] 1) obtaining analog sound signals output by each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone, and converting the analog sound signals into multi-channel digital sound signals, wherein each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are alternately distributed in sequence with equal intervals along a straight line;
[0110] 2) preprocessing the multi-channel digital sound signal, filtering the preprocessed multi-channel sound signal through a beamforming filter to obtain processed sound signal data;
[0111] 3) Outputting the processed sound signal data and the multi-channel digital sound signal.
[0112] In this embodiment, the process of preprocessing the multi-channel digital sound signal by the mid-end sound signal processing module is as follows:
[0113] The multi-channel digital sound signal is processed in sequence by a mid-end sound signal processing module using a direct current filter, an overlapping frame function, a window function filter and a short-time Fourier transform.
[0114] In this embodiment, the method of obtaining the analog sound signals output by the omnidirectional sound pickup electret microphone and the directional sound pickup electret microphone through the audio codec chip and converting the analog sound signals into multi-channel digital sound signals also includes:
[0115] Automatically adjust the gain of the multi-channel digital sound signal data.
[0116] In this embodiment, the parameter solution expression of the beamforming filter is:
[0117] h=B H (BB H ) -1 c(θ s )
[0118] in, The expressions are:
[0119]
[0120]
[0121] in, The Nth-order optimal beam pattern of a uniform linear array of omnidirectional microphone capsules, is the Nth-order optimal beam pattern of a uniform linear array of directional microphone capsules, θ s The expected direction.
[0122] Embodiment 4
[0123] A computer device includes a memory, a processor, and a computer program stored in the memory and executable on the processor. When the processor executes the computer program, the steps of implementing the sound pickup method with adjustable beam directivity are, for example, including: obtaining analog sound signals output by an omnidirectional sound pickup electret microphone and a directional sound pickup electret microphone, and converting the analog sound signals into multi-channel digital sound signals; preprocessing the multi-channel digital sound signals, filtering the preprocessed multi-channel sound signals through a beamforming filter to obtain processed sound signal data; outputting the processed sound signal data and the multi-channel digital sound signals, wherein each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are alternately distributed in a straight line at equal intervals. The memory may include a memory, such as a high-speed random access memory, and may also include a non-volatile memory, such as at least one disk memory, etc. The processor, the network interface, and the memory are interconnected through an internal bus, which may be an industrial standard architecture bus, a peripheral component interconnection standard bus, an extended industrial standard architecture bus, etc. The bus may be divided into an address bus, a data bus, a control bus, etc. The memory is used to store programs, specifically, the programs may include program codes, and the program codes include computer operation instructions. The memory may include internal memory and non-volatile memory, and provide instructions and data to the processor.
[0124] Embodiment 5
[0125] A computer-readable storage medium stores a computer program, and when the computer program is executed by a processor, the steps of implementing the sound pickup method with adjustable beam directivity are, for example, including: obtaining analog sound signals output by an omnidirectional sound pickup electret microphone and a directional sound pickup electret microphone, and converting the analog sound signals into multi-channel digital sound signals; preprocessing the multi-channel digital sound signals, filtering the preprocessed multi-channel sound signals through a beamforming filter to obtain processed sound signal data; outputting the processed sound signal data and the multi-channel digital sound signals, wherein each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are staggered and distributed in sequence along a straight line at equal intervals. Specifically, the computer-readable storage medium includes, but is not limited to, for example, volatile memory and / or non-volatile memory. The volatile memory may include a random access memory (RAM) and / or a cache memory (cache), etc. The non-volatile memory may include a read-only memory (ROM), a hard disk, a flash memory, an optical disk, a magnetic disk, etc.
[0126] Those skilled in the art will appreciate that the embodiments of the present application may be provided as methods, systems, or computer program products. Therefore, the present application may adopt the form of a complete hardware embodiment, a complete software embodiment, or an embodiment in combination with software and hardware. Moreover, the present application may adopt the form of a computer program product implemented in one or more computer-usable storage media (including but not limited to disk storage, CD-ROM, optical storage, etc.) that include computer-usable program code.
[0127] The present application is described with reference to the flowcharts and / or block diagrams of the methods, devices (systems), and computer program products according to the embodiments of the present application. It should be understood that each process and / or box in the flowchart and / or block diagram, as well as the combination of the processes and / or boxes in the flowchart and / or block diagram, can be implemented by computer program instructions. These computer program instructions can be provided to a processor of a general-purpose computer, a special-purpose computer, an embedded processor, or other programmable data processing device to generate a machine, so that the instructions executed by the processor of the computer or other programmable data processing device generate instructions for implementing the processes in the flowchart and / or block diagram. Figure 1 A process or multiple processes and / or boxes Figure 1 A device that provides the functions specified in a block or multiple blocks.
[0128] These computer program instructions may also be stored in a computer-readable memory capable of directing a computer or other programmable data processing device to operate in a specific manner, so that the instructions stored in the computer-readable memory produce an article of manufacture comprising an instruction device, which implements the process Figure 1 A process or multiple processes and / or boxes Figure 1A function specified in one or more boxes.
[0129] These computer program instructions can also be loaded onto a computer or other programmable data processing device so that a series of operating steps are executed on the computer or other programmable device to produce a computer-implemented process, thereby providing instructions for implementing the process. Figure 1 A process or multiple processes and / or boxes Figure 1 The steps for the functions specified in one or more boxes.
[0130] Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention rather than to limit it. Although the present invention has been described in detail with reference to the above embodiments, ordinary technicians in the relevant field should understand that the specific implementation methods of the present invention can still be modified or replaced by equivalents. Any modification or equivalent replacement that does not depart from the spirit and scope of the present invention should be covered within the scope of protection of the claims of the present invention.
Claims
1. A microphone system with adjustable beam directivity, characterized in that: It includes a front-end sound signal acquisition module, a mid-end sound signal processing module and a back-end sound signal output module, wherein the front-end sound signal acquisition module includes an audio codec Codec chip, a plurality of omnidirectional sound pickup electret microphones and a plurality of directional sound pickup electret microphones, each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are staggered and distributed in sequence at equal intervals along a straight line, each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are connected to the input end of the audio codec Codec chip, and the mid-end sound signal processing module is connected to the output end of the audio codec Codec chip and the back-end sound signal output module.
2. The microphone system with adjustable beam directivity according to claim 1, characterized in that: The mid-end sound signal processing module includes a digital signal processing DSP chip, and the digital signal processing DSP chip is connected to the output end of the audio codec Codec chip and the back-end sound signal output module.
3. The microphone system with adjustable beam directivity according to claim 2, characterized in that: The back-end sound signal output module comprises an Ethernet chip, a DAC chip and a USB chip, wherein a digital signal processing DSP chip is connected to the Ethernet chip, the DAC chip and the USB chip.
4. The microphone system with adjustable beam directivity according to claim 3, characterized in that: The DAC chip is connected to the DAC interface pin of the digital signal processing DSP chip; The USB chip is connected to the USB interface pin of the digital signal processing DSP chip; The Ethernet chip is connected to the Ethernet corresponding interface pin of the digital signal processing DSP chip.
5. A sound pickup method with adjustable beam directivity, characterized in that: The following steps are involved: Acquire analog sound signals output by an omnidirectional sound pickup electret microphone and a directional sound pickup electret microphone, and convert the analog sound signals into multi-channel digital sound signals, wherein each omnidirectional sound pickup electret microphone and each directional sound pickup electret microphone are alternately distributed in sequence at equal intervals along a straight line; Preprocessing the multi-channel digital sound signal, filtering the preprocessed multi-channel sound signal through a beamforming filter to obtain processed sound signal data; The processed sound signal data and the multi-channel digital sound signal are output.
6. The sound pickup method with adjustable beam directivity according to claim 5, characterized in that: The process of preprocessing the multi-channel digital sound signal is as follows: The multi-channel digital sound signal is processed in sequence by a direct current filter, an overlapping frame function, a window function filter and a short-time Fourier transform.
7. The sound pickup method with adjustable beam directivity according to claim 5, characterized in that: The method further includes: obtaining analog sound signals output by the omnidirectional sound pickup electret microphone and the directional sound pickup electret microphone, and converting the analog sound signals into multi-channel digital sound signals: Perform data automatic gain adjustment on the multi-channel digital sound signal.
8. The sound pickup method with adjustable beam directivity according to claim 5, characterized in that: The parameter solution expression of the beamforming filter is: h=B H (BB H ) -1 c(θ s ) in, The expressions are: in, The Nth-order optimal beam pattern of a uniform linear array of omnidirectional microphone capsules, is the Nth-order optimal beam pattern of a uniform linear array of directional microphone capsules, θ s The expected direction.
9. A computer device comprising a memory, a processor, and a computer program stored in the memory and executable on the processor, characterized in that: When the processor executes the computer program, the steps of the sound pickup method with adjustable beam directivity as described in any one of claims 5 to 8 are implemented.
10. A computer-readable storage medium storing a computer program, characterized in that: When the computer program is executed by a processor, the steps of the sound pickup method with adjustable beam directivity as described in any one of claims 5 to 8 are implemented.
Citation Information
Cited By
Directive MEMS microphone, beam forming method and device, and electronic equipment
CN121462957A