Digital array microphone based on triangular configuration and digital microphone system

Abstract

According to the digital array microphone based on the triangular configuration and the digital microphone system, three piezoelectric sensing modules are integrated in one digital microphone in a triangular array mode, and a beamforming and orientation algorithm which is designed in advance is integrated into the microphone. The design not only retains high definition, low power consumption and strong compatibility of the digital microphone, but also realizes more accurate sound source localization and higher-quality sound capture through the triangular array layout. Meanwhile, due to the integrated design, the manufacturing process is simplified, the cost is reduced, and the integration convenience of the equipment is improved. The built-in beam forming and orientation algorithm further improves the usability and the performance stability of the system, and the high-grade sound processing function can be enjoyed without complex configuration. In addition, the algorithm is directly integrated at the acquisition end, so that the processing time delay is optimized, and the sound definition and accuracy are greatly improved.

Description

Technical Field

[0001] The present invention relates to the technical field of audio data processing, and in particular to a digital array microphone and a digital microphone system based on a triangle configuration. Background Art

[0002] Digital microphones are an important development in modern audio technology. They convert sound signals directly from analog to digital form through a built-in analog-to-digital converter (ADC), thereby optimizing the processing and transmission of audio signals. Compared with traditional analog microphones, digital microphones have shown many significant advantages, including higher signal-to-noise ratio, lower total harmonic distortion (THD), and wider dynamic range. These features enable digital microphones to more accurately retain the quality of the original signal when capturing sound, and maintain good recording effects even in environments with high background noise or a large sound dynamic range.

[0003] Another key advantage of digital microphones is their strong resistance to electromagnetic interference. Since the signal is converted into a digital format inside the microphone, the sensitivity of digital signals to external interference during long-distance transmission is significantly reduced compared to analog signals. This is especially important for application scenarios that require long cable transmission or are used in environments with strong electromagnetic interference.

[0004] In terms of application, digital microphones are widely used in various occasions, from consumer electronics (such as smartphones, tablets, smart speakers, etc.) to professional audio recording equipment, and even remote conferencing systems and security monitoring systems. Their high degree of integration and excellent performance make digital microphones an ideal choice for improving audio input quality.

[0005] Although digital microphones are widely used in various devices and systems, providing clear sound quality and high digital compatibility, they face obvious performance limitations in complex environments because they rely on a single piezoelectric sensor module to capture sound. This is mainly manifested in the difficulty of effectively separating multiple sound sources and controlling background noise, especially in applications that require precise sound source positioning and high-quality audio capture, such as multi-person conference systems and smart home devices. These limitations reduce the competitiveness of digital microphones in the high-end audio application market. The specific manifestations are as follows:

[0006] 1. Performance limitations of a single sensor module: Digital microphones on the market are generally equipped with only one piezoelectric sensor module, which means that their sound capture capabilities are limited to single-point signal processing. In scenarios where multiple sound sources need to be distinguished or sound needs to be clearly captured in a noisy background, microphones with this configuration are difficult to meet the needs of high-quality audio processing. As users' requirements for audio performance increase, digital microphones with a single sensor module have shown their limitations in many applications.

[0007] 2. Cost and complexity of array configuration: Although combining multiple independent single-sensor module microphones to form an array is a feasible solution, doing so will introduce additional complexity and cost. Building a microphone array requires complex hardware configuration and sophisticated synchronization mechanisms, which increases the overall cost and operational difficulty of the system. In addition, this approach also requires more complex signal processing algorithms, which do not have much advantage over traditional analog microphone systems in terms of cost and technical implementation.

[0008] 3. Increased software development and maintenance costs: The use of multi-microphone arrays also involves the custom development of software and algorithms, including the development of beamforming and sound source localization algorithms. The implementation of these technologies not only requires a high level of expertise, but also requires significant time and financial investment, especially during the debugging and optimization process, the cost will rise further. Summary of the invention

[0009] In view of the above-mentioned shortcomings of the prior art, an object of the present invention is to provide a digital array microphone and a digital microphone system based on a triangular configuration, so as to solve the above-mentioned problems of the prior art.

[0010] To achieve the above-mentioned purpose and other related purposes, the present invention provides a digital array microphone based on a triangular configuration, comprising: three MEMS modules arranged in an equilateral triangle array; each MEMS module is used to convert the sound pressure signal received by each MEMS module into an analog voltage signal; three analog-to-digital conversion modules correspondingly connected to each MEMS module are used to convert the analog voltage signal output by each MEMS module into a digital signal; a signal processing module connected to each analog-to-digital conversion module is used to perform beamforming and sound source orientation operations according to the three digital signals output by the three analog-to-digital conversion modules and in combination with the control signal from the main control chip, to obtain an enhanced signal and send the orientation result to the main control chip at the same time; a modulation module is connected to the signal processing module and is used to perform multi-bit to 1-bit digital modulation processing on the enhanced signal output by the signal processing module to obtain a modulated signal; an interface module is connected to the modulation module and is used to convert the modulated signal output by the modulation module into a PDM signal and send it to the main control chip.

[0011] In one embodiment of the present invention, the signal processing module is used to form beams pointing in three different directions from the three input digital signals, and determine the direction of the sound source based on the control signal of the main control chip, and output the beam signal in the corresponding direction as an enhanced signal, and send the directional result to the main control chip.

[0012] In one embodiment of the present invention, the three digital signals inputted are used to form beams pointing in three different directions, and the direction of the sound source is determined based on the control signal of the main control chip, including: converting the three digital signals into frequency domains respectively to obtain three frequency domain signals; forming filtered signals pointing in three different directions based on the three frequency domain signals inputted at the same time; converting the filtered signals pointing in three different directions into time domains respectively to obtain beam signals pointing in three different directions; and determining the direction of the sound source based on the beam signals pointing in three different directions based on the control signal of the main control chip.

[0013] In one embodiment of the present invention, the signal processing module includes: three STFT processing units, which are used to convert each input digital signal into a frequency domain signal; three beam formers, which are respectively connected to the three STFT processing units, and are used to form filter signals pointing to corresponding directions according to the three frequency domain signals input at the same time; wherein the filter signals formed by the three beam formers point to three different directions; three ISTFT processing units, which are respectively connected to a beam former, and are used to convert each input filter signal pointing to the corresponding direction into the time domain, and obtain a beam signal pointing to the direction; the beam selection and orientation unit is connected to each ISTFT processing unit, and is used to input beam signals pointing to three different directions, and determine the sound source direction in combination with the control signal sent in real time by the main control chip, and output the beam signal in the corresponding direction as an enhanced signal, and at the same time send the determined sound source direction to the main control chip as a orientation result.

[0014] In one embodiment of the present invention, each beamformer is pre-set with a beamforming coefficient that matches the direction of the filtering signal formed by it; each beamformer calculates the filtering signal pointing to the corresponding direction using the pre-set beamforming coefficient.

[0015] In one embodiment of the present invention, the beamforming coefficients preset in each of the three beamformers are calculated based on the relative positions of the three MEMS modules and the relative arrival time differences of the sound waves in the three directions reaching each MEMS module.

[0016] In one embodiment of the present invention, each beamformer weights three frequency domain signals input simultaneously by a preset beamforming coefficient and then sums them to obtain a filtered signal pointing to a corresponding direction.

[0017] In one embodiment of the present invention, the beam selection and orientation unit is used to directly output the beam signal of the specified direction as an enhanced signal when receiving a control signal of a specified beam direction, and send the specified direction as an orientation result to the main control chip; when receiving a control signal of a non-specified beam direction, the optimal beam direction is determined using the optimal beam direction determination process, and the beam signal of the corresponding direction is output as an enhanced signal, and the determined optimal beam direction is sent to the main control chip as an orientation result.

[0018] In one embodiment of the present invention, the optimal beam direction determination process includes: calculating the power of the beamforming output in each direction based on the beam signal pointing in each direction; determining the direction that maximizes the power of the beamforming output based on the power of the beamforming output in each direction, and using this direction as the optimal beam direction.

[0019] To achieve the above-mentioned object and other related objects, the present invention provides a digital microphone system, which includes: a main control chip and a digital array microphone based on a triangular configuration; wherein the main control chip and the digital array microphone are communicatively connected via an interface.

[0020] As described above, the present invention provides a digital array microphone and a digital microphone system based on a triangular configuration, which have the following beneficial effects: the present invention integrates three piezoelectric sensor modules in a digital microphone in the form of a triangular array, and integrates the pre-designed beamforming and directional algorithms into the microphone. This design not only retains the high definition, low power consumption and strong compatibility of the digital microphone, but also achieves more accurate sound source positioning and higher quality sound capture through the triangular array layout. At the same time, this integrated design simplifies the manufacturing process, reduces costs, and improves the ease of integration of the equipment. The built-in beamforming and directional algorithms further enhance the ease of use and performance stability of the system, and advanced sound processing functions can be enjoyed without complex configuration. In addition, since the algorithm is directly integrated at the acquisition end, the processing delay is optimized, and the sound clarity and accuracy are greatly improved. BRIEF DESCRIPTION OF THE DRAWINGS

[0021] Figure 1 Shown is a schematic structural diagram of a triangular-configuration digital array microphone in one embodiment of the present invention.

[0022] Figure 2 Shown is a schematic diagram of the layout of three MEMS modules in one embodiment of the present invention.

[0023] Figure 3 It is a schematic diagram of a second-order structure in one embodiment of the present invention.

[0024] Figure 4 Shown is a structural schematic diagram of a signal processing module in an embodiment of the present invention.

[0025] Figure 5 Shown is a structural schematic diagram of a digital microphone system in one embodiment of the present invention. DETAILED DESCRIPTION

[0026] The following describes the embodiments of the present invention by specific examples, and those skilled in the art can easily understand other advantages and effects of the present invention from the contents disclosed in this specification. The present invention can also be implemented or applied through other different specific embodiments, and the details in this specification can also be modified or changed in various ways based on different viewpoints and applications without departing from the spirit of the present invention. It should be noted that the following embodiments and features in the embodiments can be combined with each other without conflict.

[0027] It should be noted that in the following description, reference is made to the accompanying drawings, which describe several embodiments of the present invention. It should be understood that other embodiments may also be used, and that mechanical composition, structural, electrical and operational changes may be made without departing from the spirit and scope of the present invention. The following detailed description should not be considered restrictive, and the scope of the embodiments of the present invention is limited only by the claims of the published patents. The terms used herein are only for describing specific embodiments and are not intended to limit the present invention. Spatially related terms, such as "upper", "lower", "left", "right", "below", "below", "lower", "above", "upper", etc., may be used in the text to facilitate the description of the relationship between an element or feature shown in the figure and another element or feature.

[0028] Throughout the specification, when a part is said to be "connected" to another part, this includes not only the case of "direct connection" but also the case of "indirect connection" by placing other elements therebetween. In addition, when a part is said to "include" a certain constituent element, unless otherwise stated, it does not exclude other constituent elements, but means that other constituent elements may be included.

[0029] The terms first, second and third mentioned herein are used to describe various parts, components, regions, layers and / or segments, but are not limited thereto. These terms are only used to distinguish a certain part, component, region, layer or segment from other parts, components, regions, layers or segments. Therefore, the first part, component, region, layer or segment described below may refer to the second part, component, region, layer or segment within the scope of the present invention.

[0030] Furthermore, as used herein, the singular forms "a", "an" and "the" are intended to include the plural forms as well, unless there is an indication to the contrary in the context. It should be further understood that the terms "comprise", "include" indicate the presence of the described features, operations, elements, components, items, kinds, and / or groups, but do not exclude the presence, occurrence or addition of one or more other features, operations, elements, components, items, kinds, and / or groups. The terms "or" and "and / or" used herein are interpreted as inclusive, or mean any one or any combination. Therefore, "A, B or C" or "A, B and / or C" means "any of the following: A; B; C; A and B; A and C; B and C; A, B and C". Exceptions to this definition will only occur when the combination of elements, functions or operations is inherently mutually exclusive in some way.

[0031] The digital array microphone based on a triangular configuration of the present invention integrates three piezoelectric sensor modules in a triangular array into a digital microphone, and integrates the pre-designed beamforming and directional algorithms into the microphone. This design not only retains the high definition, low power consumption and strong compatibility of the digital microphone, but also achieves more accurate sound source positioning and higher quality sound capture through the triangular array layout. At the same time, this integrated design simplifies the manufacturing process, reduces costs, and improves the convenience of equipment integration. The built-in beamforming and directional algorithms further enhance the ease of use and performance stability of the system, and advanced sound processing functions can be enjoyed without complex configuration. In addition, since the algorithm is directly integrated at the acquisition end, the processing delay is optimized, and the sound clarity and accuracy are greatly improved.

[0032] The following is a detailed description of the embodiments of the present invention with reference to the accompanying drawings so that those skilled in the art can easily implement the present invention. The present invention can be embodied in many different forms and is not limited to the embodiments described herein.

[0033] like Figure 1 A structural schematic diagram of a triangular-configuration digital array microphone in an embodiment of the present invention is shown.

[0034] The system comprises:

[0035] Three MEMS modules are arranged in an equilateral triangle array, namely, MEMS module 11, MEMS module 12 and MEMS module 13; in the layout of the equilateral triangle array, MEMS module 11, MEMS module 12 and MEMS module 13 are respectively placed at the three vertices of the equilateral triangle to form an equilateral triangle array with a side length of d. The specific layout is as follows Figure 2As shown in the figure, each MEMS module has independent functions and can convert the sound pressure signals it receives into analog voltage signals. This layout enables the system to use these three modules to capture sound waves from different directions.

[0036] The three analog-to-digital conversion modules corresponding to each MEMS module, namely, the analog-to-digital conversion module 21, the analog-to-digital conversion module 22 and the analog-to-digital conversion module 23, are used to convert the analog voltage signals output by the MEMS module 11, the MEMS module 12 and the MEMS module 13 into digital signals respectively, forming three digital signals; specifically, the analog-to-digital conversion module 21 converts the analog voltage signal 1 output by the MEMS module 11 into a digital signal 1, the analog-to-digital conversion module 22 converts the analog voltage signal 2 output by the MEMS module 12 into a digital signal 2, and the analog-to-digital conversion module 23 converts the analog voltage signal 3 output by the MEMS module 13 into a digital signal 3.

[0037] The signal processing module 3 is connected to the analog-to-digital conversion module 21, the analog-to-digital conversion module 22 and the analog-to-digital conversion module 23, and is used to perform beamforming and sound source orientation operations according to the three digital signals output by the analog-to-digital conversion module 21, the analog-to-digital conversion module 22 and the analog-to-digital conversion module 23, and in combination with the control signal from the main control chip, to obtain an enhanced signal and simultaneously send the orientation result to the main control chip;

[0038] The modulation module 4 is connected to the signal processing module 3 and is used to perform a multi-bit to 1-bit digital modulation process on the enhanced signal output by the signal processing module 3 to obtain a modulated signal;

[0039] The interface module 5 is connected to the modulation module 4 and is used to convert the modulation signal output by the modulation module 4 into a PDM (Pulse Density Modulation) signal and send it to the main control chip.

[0040] The specific implementation principles of the MEMS module, the analog-to-digital conversion module, the modulation module and the interface module are now described in conjunction with the following embodiments.

[0041] In a specific embodiment, the MEMS (Micro-Electro-Mechanical Systems) module adopts a capacitive MEMS structure, including a thin and elastic acoustic diaphragm and a rigid back plate. The diaphragm, the back plate and the air gap between them together form a parallel plate capacitor, so there is,

[0042] V=Q / C,C=εS / x; (Formula 1)

[0043] Where C is the capacitance, S is the area of ​​the plate, Q is the amount of charge stored when the voltage between the plates is V, ε is the dielectric constant of the medium (air) between the plates, and x is the distance between the two plates.

[0044] When the sound pressure changes on the diaphragm, it will cause the elastic displacement of the film, and the distance x between the two plates will change, C will change, Q remains unchanged using an external constant voltage power supply, then the voltage between the two plates will change. This process completes the conversion process of the sound signal to the voltage signal.

[0045] In a specific embodiment, the analog-to-digital conversion module used in the present invention is mainly responsible for converting analog signals into digital signals. The following is a detailed description of the working principle of the module:

[0046] First, in order to optimize the dynamic range of the digital array microphone and meet the requirements of different sensitivities, the analog voltage signal output by the MEMS module is subjected to gain matching processing to improve the dynamic range of the system to adapt to the requirements of different sensitivities. The specific expression is:

[0047] Vg = g * V; (Formula 2)

[0048] In the above formula, g is the adjustable gain coefficient, V is the analog voltage signal output by the MEMS module, and Vg is the adjusted analog voltage signal.

[0049] Secondly, the signal Vg after gain adjustment is converted into digital signal based on the sigma-delta modulation principle. Sigma-delta modulation is based on the principle of integral loop filtering and shaping, which moves the noise signal energy from low frequency to high frequency to improve the signal-to-noise ratio performance of low frequency. Figure 3 As shown, taking the second-order structure as an example, its mathematical model can be expressed as:

[0050] V(z) = z -1 *U(z)+(1-z -1 ) 2 *E(z); (Formula 3)

[0051] In the above formula, the noise transfer function is (1-z -1 ) 2 , which is used to shape the noise signal E(z). The higher the order, that is, the more cascaded integrators there are and the larger the power, the more effective the shaping will be. The signal transfer function is z -1 , which delays the signal U(z) by one sample point without doing any other processing on the signal, thus retaining the original characteristics of the signal. Finally, the quantizer in the structure accurately quantizes the shaped signal into a digital signal, that is, converts it into the digital signal that the analog-to-digital conversion module finally outputs.

[0052] In a specific embodiment, the modulation module 4 is responsible for converting the encrypted multi-bit digital signal into a 1-bit signal. The modulation module 4 performs multi-bit to 1-bit digital modulation processing on the enhanced signal output by the signal processing module 3, thereby simplifying the interface, saving chip interaction pin resources, and maintaining high performance.

[0053] The working principle of the modulation module 4 is based on the sigma-delta modulation principle, which is also used in the analog-to-digital conversion module, but in the modulation module 4, its structural design and implementation method are different. The modulation module 4 adopts a CIFB structure of multi-stage integrator cascade, which completes the function of converting multiple bits to 1 bit while transferring low-frequency noise modulation to high frequency, maintaining high performance at low frequency. Taking the second-order structure as an example, its mathematical model can be expressed as:

[0054] V(z) = z -1 *U(z)+(1-z -1 ) 2 *E(z); (Formula 3)

[0055] In the above formula, although it is the same as the modulation module 4, the input U(z) of the modulation module 4 is a multi-bit digital signal, and the output of the quantizer is a 1-bit signal, which has only two values ​​0 and 1, so it is also called a binary modulator. This design allows the modulation module to focus only on the modulation function between digital signals without involving the analog-to-digital conversion process in the analog-to-digital conversion module.

[0056] In a specific embodiment, the interface module 5 is responsible for converting the modulated signal into a PDM signal. The input of the interface module 5 is a modulated signal, and the output is a PDM signal. The principle of this module is described as follows:

[0057] The 1-bit high-speed PDM code stream signal output by the modulation module 4 is controlled by timing logic. The basic principle is to generate an OEN enable signal that controls DATA IO through timing logic, so that it generates a low-level signal when outputting the data of the current digital microphone, and maintains a high-impedance signal when not outputting the digital microphone data. Specifically expressed as:

[0058] Dout = (oen == 0) ? data: high-z; (Formula 5)

[0059] In the above formula, this logic means that when oen is low level (0), data is output, and when oen is high level (1), high configuration high-z is output.

[0060] The signal processing module 3 is the core of the present invention, and is mainly responsible for executing beamforming and directional algorithms to achieve efficient processing of the received three-way signals. The working principle of the signal processing module 3 is now described in detail in conjunction with the following embodiments.

[0061] In one embodiment, the signal processing module 3 receives three digital signals from three MEMS sensor modules, and the three signals represent the sound wave information in three different directions; the signal processing module 3 can combine the signals in these three directions into three independent beams by applying a beamforming algorithm, and each beam points to a specific direction; after forming the beam, the signal processing module 3 will determine the direction of the sound source according to the control signal sent by the main control chip. Once the direction of the sound source is determined, the signal processing module 3 will select the beam signal in the corresponding direction as the enhanced signal output, and send the directional result to the main control chip.

[0062] In this example, if Figure 2 , since the three MEMS modules are arranged in an equilateral triangle array, the relative positions of the three MEMS modules are known, and the three different directions are in the directions perpendicular to the three sides of the triangle, namely 0°, 120°, -120° (or 240°), and the angle difference between them is 120 degrees. Assuming that MEMS module 11 is located at the origin and the direction pointing to MEMS module 12 is 0°, then we can define the three directions as follows: the direction from MEMS module 11 to MEMS module 12 is 0°, the direction from MEMS module 11 to MEMS module 13 is 120°, and the direction from MEMS module 11 to the opposite direction of MEMS module 12 (or the opposite direction of the extension line of MEMS module 12) is -120° (or 240°).

[0063] In a specific embodiment, in the signal processing module, beams facing three different directions are formed based on three digital signals, and the direction of the sound source is determined according to the control signal of the main control chip, which specifically includes the following steps:

[0064] The three digital signals from the three MEMS sensing modules are converted from the time domain to the frequency domain to obtain three frequency domain signals. The frequency domain signals provide the amplitude and phase information of the signals at different frequencies, which is crucial for the subsequent beamforming processing.

[0065] Based on the three frequency domain signals input simultaneously, filtered signals pointing in three different directions are formed.

[0066] The filtered signals pointing in three directions are converted from the frequency domain back to the time domain to obtain three beam signals pointing in different directions.

[0067] Based on the control signal of the main control chip, the direction of the sound source is determined according to the beam signals pointing in three different directions.

[0068] In a specific embodiment, if Figure 4 As shown, the signal processing module includes:

[0069] Three STFT (short-time Fourier transform) processing units, namely STFT processing unit 311, STFT processing unit 312, and STFT processing unit 313; these three STFT processing units are respectively connected to an analog-to-digital conversion module, respectively receive a time domain digital signal from the MEMS sensor, and convert the digital signal into a frequency domain signal.

[0070] STFT (Short-time Fourier Transform) is a commonly used time-frequency analysis method that can perform localized analysis of signals in the time domain and frequency domain. For each MEMS module, its corresponding time domain signal is converted to the frequency domain through STFT to obtain the frequency domain representation X n,m (f), specifically expressed as:

[0071]

[0072] In the formula, x n,m (t) is the time domain signal of the mth frame of the nth MEMS module, Stands for Fourier Transform.

[0073] Three beamformers, namely beamformer 321, beamformer 322, and beamformer 323, are respectively connected to three STFT processing units, respectively receive frequency domain signals from the three STFT processing units simultaneously, and apply a beamforming algorithm to form filter signals pointing in three different directions; for example, a filter signal pointing in the 0° direction, a filter signal pointing in the 120° direction, and a filter signal pointing in the -120° direction.

[0074] Three ISTFT (Inverse Short Time Fourier Transform) processing units, namely ISTFT processing unit 331, ISTFT processing unit 332, and ISTFT processing unit 333; the three ISTFT processing units are respectively connected to a beamformer, respectively receive the filtered signals from the connected beamformer, and convert them back to time domain signals. ISTFT (Inverse Short Time Fourier Transform) is the inverse process of STFT, which can convert the frequency domain signal back into the time domain signal. These time domain signals are beam signals pointing in three different directions, namely beam signals pointing to 0°, 120°, and -120°.

[0075] The specific process of forming a beam signal pointing in the direction θ is as follows:

[0076]

[0077] Where y(t,θ) is the beam signal pointing in the direction of θ, Y m (f,θ) is the input frequency domain signal pointing in the direction of θ.

[0078] The beam selection and orientation unit 34 is connected to each ISTFT processing unit, and is used to input beam signals pointing in three different directions, and determine the direction of the sound source in combination with the control signal sent in real time by the main control chip, and output the corresponding beam signal as an enhanced signal, and at the same time send the determined sound source direction as a orientation result to the main control chip.

[0079] In a specific embodiment, before beamforming, each beamformer needs to pre-set a beamforming coefficient that matches the direction of the filter signal formed by it. Each beamformer uses the correspondingly set beamforming coefficient to calculate the filter signal pointing to the corresponding direction according to the three frequency domain signals input simultaneously.

[0080] The beamforming coefficients pre-set in the three beamformers are calculated based on the relative positions of the three MEMS modules and the relative arrival time differences of the sound waves in the three directions reaching each MEMS module. The specific implementation is as follows:

[0081] The beamforming coefficient W for each beamformer setting n,m (f,θ), which is determined by the relative positions of the three MEMS modules and the relative arrival time difference (or equivalent phase difference Δφ) of the sound waves arriving at each MEMS module in the three directions. n,m (f,θ)) is calculated. Its mathematical expression is:

[0082]

[0083] Considering that the distance from each vertex to the center of an equilateral triangle is We can calculate the phase difference from the direction θ and the speed of sound c.

[0084] Taking the 0° direction as an example, assuming that the beamforming coefficient corresponding to the MEMS module 11 is used as a reference, the beamforming coefficient of the MEMS module 11 is:

[0085] W 1,m (f,0°)=1; (Formula 9)

[0086] For MEMS modules 12 and 13, considering the relative path difference of the sound wave arriving from the 0° direction, we have:

[0087]

[0088] Accordingly, the phase difference Δφ n,m (f,0°) is:

[0089]

[0090] Therefore, the beamforming coefficients corresponding to the MEMS modules 12 and 13 are:

[0091]

[0092] It should be noted that the step of solving the beamforming coefficients here is completed in advance. In actual applications, since the configuration of the microphone array is fixed, the coefficients do not need to be solved individually each time, but only need to be solved once during the design phase.

[0093] In one embodiment, each beamformer weights three frequency domain signals input simultaneously by a preset beamforming coefficient and then sums them to obtain a filtered signal pointing to a corresponding direction.

[0094] The specific formula is as follows:

[0095]

[0096] Where Y m (f,θ) represents the beamformed signal at frequency f and direction θ, X n,m (f) represents the frequency domain signal of the nth MEMS module at frequency f and direction θ, W n,m (f,θ) represents the beamforming coefficient corresponding to the nth MEMS module.

[0097] In one embodiment, the beam selection and orientation unit has two beam selection modes according to the type of control signal sent in real time by the main control chip:

[0098] The first mode is: when a control signal pointing to a specific beam direction is received, the unit directly selects and outputs the beam signal of the specified beam direction as an enhanced signal; and sends this specified beam direction as a directional result to the main control chip for further processing or decision-making.

[0099] The second mode is: when receiving a control signal of a non-specified beam direction, the unit will start the optimal beam direction determination process, and calculate or identify the optimal beam direction through an algorithm or a preset program. Once the optimal beam direction is determined, the unit will output the corresponding beam signal as an enhanced signal, and send the calculated optimal beam direction as the orientation result to the main control chip.

[0100] This design enables the beam selection and directionality unit to flexibly respond to different control signals, whether it is a directly specified beam direction for compatibility with certain applications that require single-directional beamforming results, or an optimal beam direction that needs to be determined through calculation. Signal processing and result output can be effectively performed to meet different application scenarios and requirements.

[0101] In one embodiment, the optimal beam direction determination process includes:

[0102] The power of the beamforming output in each direction is calculated based on the beam signal pointing in each direction; specifically, for each direction θ, we obtain the beamforming output signal y(t,θ) through STFT and beamforming algorithm, where t refers to the time index. For each direction, the power of the beamforming output P(θ) is calculated as follows:

[0103]

[0104] Here, N is the total number of sampling points in the observation window, and y(t,θ) is the beamforming output signal at time t.

[0105] According to the power of the beamforming output in each direction, the direction θ that maximizes the power of the beamforming output is determined opt , and take this direction as the optimal beam direction, expressed by the following formula:

[0106]

[0107] The optimal beam direction θ is determined opt After that, the corresponding optimal beamforming output is the beamforming output y(t,θ opt ).

[0108] In one embodiment, in addition to the optimal beam direction determination process described in the above embodiment, the optimal beam direction determination process may also include the following method: determining the optimal beam direction based on the relative position of the MEMS module, which may include time difference measurement, signal strength analysis, beamforming algorithm and other technologies.

[0109] The present invention innovatively proposes to integrate three MEMS modules in a single digital microphone, but the design is not limited to the configuration of three MEMS modules, and two or more MEMS modules can be flexibly integrated according to actual application requirements. By integrating different numbers of MEMS modules in a single digital microphone, the system can be adjusted and optimized according to specific application scenarios. For example, in scenarios where high-precision sound positioning is required, the number of MEMS modules can be increased to improve the accuracy of sound source orientation. This flexibility is not only reflected in the number of modules, but also in the configuration and layout of the modules. By adjusting the relative positions between the MEMS modules, the performance of sound collection and processing can be further optimized. In addition, no matter how the number of the MEMS modules changes, the preset beamforming and sound source orientation algorithm logic in the signal processing module remains unchanged. This feature ensures the stability and reliability of the algorithm and reduces the potential risks introduced by algorithm changes. At the same time, according to the changes in the number of MEMS modules and their relative positions, we only need to make appropriate adaptive modifications to the algorithm to achieve performance optimization. This design not only improves the scalability and flexibility of the system, but also reduces the complexity of development and maintenance, providing convenience for future technology upgrades and functional expansion.

[0110] like Figure 4 A structural schematic diagram of a digital microphone system in an embodiment of the present invention is shown.

[0111] The system includes: a digital array microphone 100 based on a triangular configuration and a main control chip 200; since the digital array microphone 100 based on a triangular configuration has the same structure and implementation principle as the digital array microphone of the above embodiment, its structure and principle are not described in detail.

[0112] The digital array microphone 100 is communicatively connected to the main control chip 200 via a PDM interface and a control interface; specifically, the interface module 5 of the digital array microphone 100 is communicatively connected to the main control chip 200 via a PDM interface, and the signal processing module 3 of the digital array microphone 100 is communicatively connected to the main control chip 200 via a control interface.

[0113] The digital array microphone 100 is responsible for receiving external sound pressure signals, performing array signal voice enhancement processing on audio signals, and finally converting them into high-performance 1-bit PDM digital code streams, and sending them to the main control chip 200 for further processing. While transmitting the digital code stream, the digital array microphone 100 also sends the sound source orientation results to the main control chip 20 in real time. During the processing, the digital array microphone 100 receives control signals from the main control chip through the control interface.

[0114] The main control chip 200 is responsible for receiving the 1-bit digital code stream and directional results sent by the digital array microphone 100, and can send a real-time algorithm parameter control signal to the digital array microphone 100. Through this signal, it can control which beamforming signal the digital array microphone 100 finally outputs.

[0115] In response to the urgent need for a new digital microphone design that can improve audio capture and processing capabilities without adding too much complexity and cost, we proposed a digital array microphone design based on a triangular configuration. This design cleverly integrates multiple sensor modules in a single digital microphone, which not only significantly optimizes the sound capture quality, but also greatly simplifies the system configuration and maintenance process. The implementation of this design will be a major breakthrough in digital microphone technology, which can greatly improve the performance and efficiency of sound processing.

[0116] The digital array microphone design based on the triangular configuration not only solves the performance limitations of traditional single-sensor digital microphones in complex environments, but also brings a series of significant advantages to users by using its unique structure and integrated algorithm. The main advantages of the new microphone provided by the present invention are specifically described below:

[0117] 1. Improved performance and convenience

[0118] The triangular configuration-based digital array microphone of the present invention not only retains the advantages of digital microphones, such as clarity of digital signal processing, low power consumption and high compatibility, but also significantly improves the performance ceiling. In addition, the triangular array design allows the microphone to capture sound from different angles, greatly enhancing the capture accuracy and sound quality of the sound source.

[0119] 2. Easy integration and cost reduction

[0120] The present invention integrates three piezoelectric sensors into one digital microphone unit. This design not only retains the small size, low cost and easy integration characteristics of digital microphones, but also improves the processing upper limit. This integrated design simplifies the manufacturing process, reduces manufacturing costs, and is easy to be widely used in various devices.

[0121] 3. Efficient built-in algorithm

[0122] The beamforming and directional algorithms of the three-microphone array of the present invention have been pre-designed and integrated into the microphone, making it extremely convenient to use while ensuring stable performance. Users can enjoy advanced sound processing functions without complex algorithm configuration or additional software installation.

[0123] 4. Optimized processing latency and algorithm synchronization

[0124] Since all processing algorithms are integrated at the acquisition end, there is no need for a system with multiple digital microphones working together, and this design significantly reduces processing latency. This not only makes synchronization between microphones easier, but also because the algorithm is processed directly at the sound acquisition point, the upper limit of its effect is also higher, greatly optimizing the clarity and accuracy of the sound.

[0125] This triangular configuration-based digital array microphone technology, through its innovative design and technical implementation, provides the market with an efficient and economical solution, and is expected to demonstrate its strong competitive advantages in consumer electronics, enterprise conference systems, smart homes and a wider range of applications.

[0126] In summary, the digital array microphone and digital microphone system based on the triangular configuration of the present invention integrate three piezoelectric sensor modules in a digital microphone in the form of a triangular array, and integrate the pre-designed beamforming and directional algorithms into the microphone. This design not only retains the high definition, low power consumption and strong compatibility of the digital microphone, but also achieves more accurate sound source positioning and higher quality sound capture through the triangular array layout. At the same time, this integrated design simplifies the manufacturing process, reduces costs, and improves the convenience of equipment integration. The built-in beamforming and directional algorithms further enhance the ease of use and performance stability of the system, and advanced sound processing functions can be enjoyed without complex configuration. In addition, since the algorithm is directly integrated at the acquisition end, the processing delay is optimized, and the sound clarity and accuracy are greatly improved. Therefore, the present invention effectively overcomes the various shortcomings in the prior art and has a high industrial utilization value.

[0127] The above embodiments are merely illustrative of the principles and effects of the present invention, and are not intended to limit the present invention. Anyone familiar with the technology may modify or change the above embodiments without violating the spirit and scope of the present invention. Therefore, all equivalent modifications or changes made by a person of ordinary skill in the art without departing from the spirit and technical ideas disclosed by the present invention shall still be covered by the claims of the present invention.

Claims

1. A digital array microphone based on a triangular configuration, characterized in that: include: Three MEMS modules arranged in an equilateral triangle array; each MEMS module is used to convert the sound pressure signal received by each module into an analog voltage signal; Three analog-to-digital conversion modules connected to each MEMS module are used to convert the analog voltage signal output by each MEMS module into a digital signal; A signal processing module, connected to each analog-to-digital conversion module, is used to perform beamforming and sound source orientation operations according to the three digital signals output by the three analog-to-digital conversion modules and in combination with the control signal from the main control chip, to obtain an enhanced signal and simultaneously send the orientation result to the main control chip; A modulation module, connected to the signal processing module, configured to perform a multi-bit to 1-bit digital modulation process on the enhanced signal output by the signal processing module to obtain a modulated signal; The interface module is connected to the modulation module and is used to convert the modulation signal output by the modulation module into a PDM signal and send it to the main control chip.

2. The triangular configuration-based digital array microphone according to claim 1, characterized in that: The signal processing module is used to form beams pointing in three different directions from the three input digital signals, and determine the direction of the sound source according to the control signal of the main control chip, and output the beam signal in the corresponding direction as an enhanced signal, and send the directional result to the main control chip.

3. The triangular configuration-based digital array microphone according to claim 2, characterized in that: The three digital signals inputted form beams pointing to three different directions, and the direction of the sound source is determined according to the control signal of the main control chip, including: Convert the three digital signals into frequency domain respectively to obtain three frequency domain signals; Based on the three frequency domain signals input simultaneously, filtering signals pointing to three different directions are formed; The filtered signals pointing in three different directions are respectively converted into time domains to obtain beam signals pointing in three different directions; Based on the control signal of the main control chip, the direction of the sound source is determined according to the beam signals pointing in three different directions.

4. The triangular configuration-based digital array microphone according to claim 3, characterized in that: The signal processing module comprises: Three STFT processing units, used for converting respective input digital signals into frequency domain signals; Three beamformers are connected to the three STFT processing units respectively, and are used to form filter signals pointing to corresponding directions according to the three frequency domain signals input simultaneously; wherein the filter signals formed by the three beamformers point to three different directions; The three ISTFT processing units are respectively connected to a beam former, and are used to convert the filter signals pointed to the corresponding directions inputted by the respective units into the time domain, so as to obtain the beam signals pointed to the directions; The beam selection and orientation unit is connected to each ISTFT processing unit, and is used to input beam signals pointing in three different directions, and determine the direction of the sound source in combination with the control signal sent in real time by the main control chip, and output the beam signal in the corresponding direction as an enhanced signal, and at the same time send the determined sound source direction as a orientation result to the main control chip.

5. The triangular configuration-based digital array microphone according to claim 4, characterized in that: Each beamformer is pre-set with a beamforming coefficient that matches the pointing direction of the filtered signal formed by it; each beamformer calculates the filtered signal pointing in the corresponding direction using the pre-set beamforming coefficient.

6. The triangular configuration-based digital array microphone according to claim 5, characterized in that: The beamforming coefficients preset in each of the three beamformers are calculated based on the relative positions of the three MEMS modules and the relative arrival time differences of the sound waves in the three directions reaching each MEMS module.

7. The triangular configuration-based digital array microphone according to claim 5, characterized in that: Each beamformer weights the three frequency domain signals input simultaneously by a preset beamforming coefficient and then sums them to obtain a filtered signal pointing in the corresponding direction.

8. The triangular configuration-based digital array microphone according to claim 4, characterized in that: The beam selection and orientation unit is used to directly output the beam signal in the specified direction as an enhanced signal when receiving a control signal for specifying a beam direction, and at the same time send the specified direction as an orientation result to the main control chip; When a control signal of a non-specified beam direction is received, the optimal beam direction is determined using the optimal beam direction determination process, and the beam signal of the corresponding direction is output as an enhanced signal, and the determined optimal beam direction is sent to the main control chip as a directional result.

9. The triangular configuration-based digital array microphone according to claim 8, characterized in that: The optimal beam direction determination process includes: Calculate the power of the beamforming output in each direction according to the beam signal pointed in each direction; A direction that maximizes the power of the beamforming output is determined according to the power of the beamforming output in each direction, and the direction is taken as the optimal beam direction.

10. A digital microphone system, characterized in that: The system comprises: a main control chip and a digital array microphone based on a triangle configuration as described in claims 1 to 9; wherein the main control chip is communicatively connected with the digital array microphone via an interface.

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