Directional pickup method of microphone

Abstract

The invention discloses a directional pickup method for a microphone. The microphone comprises a left microphone core, a middle microphone core and a right microphone core. Wherein the middle microphone core is located on the central axis of the microphone, and the left microphone core and the right microphone core are symmetrically distributed on the left side and the right side of the middle microphone core; sound of a sound source is synchronously collected through a left microphone core, a middle microphone core and a right microphone core of the microphone, and three paths of sound signals are obtained; respectively filtering the three paths of sound signals to obtain left, middle and right microphone core signals; calculating an included angle between a connecting line of a sound source and the middle microphone core and the central axis of the microphone based on the time difference and the distance difference of the sound reaching the left, middle and right microphone cores to obtain a sound source direction; based on the left microphone core signal, the middle microphone core signal, the right microphone core signal and the sound source direction, through time delay compensation and gain adjustment, the sound signal in the target direction is enhanced, the sound signal in the non-target direction is suppressed, the final output signal is obtained, and the directionality and the sound quality of pickup are improved.

Description

Technical Field

[0001] This invention relates to a method for directional microphone pickup, which is particularly suitable for scenarios requiring precise sound source localization, such as conference systems, educational sound reinforcement, and stage performances, and belongs to the field of audio acquisition equipment. Background Technology

[0002] In the field of audio acquisition, traditional microphones generally employ omnidirectional or cardioid pickup patterns, which lack sufficient selectivity for the direction of sound sources. This results in the simultaneous recording of environmental noise (such as reverberation, background voices, and equipment interference) and the target signal, severely impacting the signal-to-noise ratio and clarity of the audio signal. Especially in professional scenarios such as conference systems, educational sound reinforcement, and stage performances, pickup devices need to accurately lock onto sound sources in specific directions (such as speakers or performers) while effectively suppressing interference from other directions. This places higher demands on audio quality and spatial positioning accuracy.

[0003] While existing directional microphone technology can partially improve the problem of insufficient directional selectivity, it still has technical defects: (1) Mechanical structure limitations: schemes that enhance directivity through physical structure (such as supercardioid diaphragms) are prone to frequency response curve distortion and have limited lateral suppression capabilities; (2) Digital beamforming technology relies on multi-microphone arrays and complex algorithms, which leads to increased system latency and higher costs, making it difficult to meet the needs of mobile devices or real-time interactive scenarios; (3) Insufficient application adaptability: existing equipment is difficult to balance noise suppression and sound quality fidelity, and cannot simultaneously meet the differentiated needs of conference voice clarity, dynamic range of instrument recording, and spatial sense of stage performance.

[0004] In summary, the performance of current directional microphones in complex acoustic environments falls significantly short of user expectations. Therefore, there is an urgent need to develop a new directional sound pickup solution. Summary of the Invention

[0005] To address the aforementioned technical problems, this invention provides a method for directional microphone pickup. By optimizing the microphone layout and signal processing circuit, it enhances sound signals from a specific direction and suppresses sound signals from non-target directions, thereby achieving the technical objective of improving the directionality of sound pickup and sound quality.

[0006] To achieve the above technical objectives, the present invention provides a method for directional microphone pickup, wherein the microphone includes left, center, and right microphones; wherein the center microphone is located on the central axis of the microphone, and the left and right microphones are symmetrically distributed on the left and right sides of the center microphone; The sound from the sound source is collected simultaneously by the left, center, and right microphone cores to obtain three sound signals; The three audio signals are filtered separately to obtain the left, center, and right microphone signals; Based on the time difference and distance difference of sound reaching the left, center and right microphones, the angle between the line connecting the sound source and the center microphone and the microphone's central axis is calculated to obtain the direction of the sound source. Based on the left, center, and right microphone signals, as well as the direction of the sound source, the sound signal in the target direction is enhanced and the sound signal in the non-target direction is suppressed through time delay compensation and gain adjustment to obtain the final output signal.

[0007] Furthermore, the step of filtering the three audio signals to obtain left, center, and right microphone signals includes: The FFT algorithm is used to convert each audio signal from a time-domain signal to a frequency-domain signal; For each frequency domain signal, retain the frequency band from 20Hz to 20kHz to obtain the filtered frequency domain signal; The IFFT algorithm is used to convert each filtered frequency domain signal into a time domain signal to obtain the left, middle and right microphone signals.

[0008] Furthermore, the present invention includes calculating the angle between the line connecting the sound source and the center microphone and the microphone's central axis based on the time difference and distance difference of sound arrival at the left, center, and right microphones to obtain the sound source direction, comprising: Let the time difference between the sound reaching the middle microphone and the left microphone be... The time difference between the sound reaching the middle microphone and the right microphone is Assuming the speed of sound in air is 340 m / s, the distance difference between the sound reaching the middle microphone core and the left microphone core is 340 m / s. The distance difference between the middle microphone core and the right microphone core is 340. ; With the center microphone as the origin, the straight lines containing the left, center, and right microphones are... The axis, the microphone's center axis is The axis, and the spacing between adjacent microchannels is Then the coordinates of the left, middle, and right microphones are respectively (- ,0), (0,0), ( ,0); Let the coordinates of the sound source be ( , The angle between the line connecting the sound source and the microphone core and the microphone's central axis is... Establish the following system of equations: Solve for the coordinates of the sound source ( , ) and included angle ,in: Furthermore, the present invention further includes, based on the left, center, and right microphone signals and the direction of the sound source, enhancing the sound signal in the target direction through time delay compensation and gain adjustment, while suppressing the sound signal in the non-target direction, to obtain the final output signal, comprising: set up For the current moment, For delay compensation of the left microphone core; To compensate for the time delay of the right microphone, the left, middle, and right microphone signals sampled at the current time are respectively , , ,but This is the left microphone signal after time delay compensation. This is the right microphone signal after time delay compensation; Also set , , These are the gain coefficients for the left, middle, and right microphone cores, respectively. If the gain function is the angle, then the final output signal The calculation formula is as follows: Furthermore, the present invention, the Take 1, the , All values ​​are -0.5.

[0009] Furthermore, the present invention, the The expression is as follows: in, The effective pickup angle threshold.

[0010] In summary, the microphone used in this invention adopts a linear three-microphone array layout (center microphone + symmetrically distributed left and right microphones), and is equipped with a specialized signal processing algorithm. First, the three audio signals are bandpass filtered (20Hz-20kHz) and time-domain aligned. Then, the direction of the sound source is calculated based on the time difference of the sound arriving at the three microphones. Through time delay compensation and gain adjustment, the sound signal in the target direction is enhanced, while the sound signal in the non-target direction is suppressed. This achieves the effect of improving the directionality of sound pickup and sound quality, and is suitable for various occasions that require directional sound pickup, such as conferences, teaching, stage performances, and recording.

[0011] Compared with the prior art, the present invention has the following technical advantages: 1) Improve directional sound pickup performance: By calculating the angle between the line connecting the sound source and the microphone core and the microphone's central axis through an algorithm, the direction of the sound source can be accurately determined, and the sound signal directly in front of the microphone can be enhanced, while the sound signals on the left and right sides of the microphone can be suppressed, effectively improving the microphone's directional sound pickup performance.

[0012] 2) Improve sound quality: The phase problem is solved by delay compensation, and the amplitude problem is solved by gain adjustment. Gain adjustment and delay compensation (phase alignment) together form a complete processing chain for beamforming, which enhances directionality, making the picked-up sound clearer and purer, reducing background noise interference, and improving sound quality.

[0013] 3) Adaptable to various application scenarios: This directional microphone is suitable for various scenarios that require directional sound pickup, such as meetings, teaching, stage performances, recording, etc., and can meet the needs of different users for directional sound pickup. Attached Figure Description

[0014] Figure 1 A structural principle block diagram of a microphone provided in an embodiment of the present invention; Figure 2 An electrical schematic diagram of a signal processing circuit board provided in an embodiment of the present invention; Figure 3 A flowchart of a method provided in an embodiment of the present invention. Detailed Implementation

[0015] To make the objectives, technical solutions, and advantages of the present invention clearer, the technical solutions of the present invention will be clearly and completely described below in conjunction with the embodiments of the present invention.

[0016] In the description of this application, terms such as "connection" or "linked" are not limited to physical or mechanical connections, but can include electrical connections, whether direct or indirect. Those skilled in the art can understand the specific meaning of these terms in this application according to the specific circumstances. Terms such as "upper," "lower," "left," "right," and "middle" are used only to indicate relative positional relationships; when the absolute position of the described object changes, the relative positional relationship may also change accordingly.

[0017] like Figure 1 As shown, this embodiment provides a method for directional microphone pickup. The microphone used includes a central microphone mounted on its central axis, and left and right microphones symmetrically mounted on either side of the central microphone. Sound from a designated sound source is acquired through a linear three-microphone array mounted in the microphone. Furthermore, based on the orientation of the three microphones, the central microphone is used to acquire sound directly in front of the microphone, while the left and right microphones are used to acquire sound from the left and right sides of the microphone, respectively.

[0018] In practice, the microphone includes a housing, three microphones installed inside the housing, and a signal processing circuit board. The three microphones are located on a straight line perpendicular to the microphone's central axis and are equally spaced. One microphone is located on the microphone's central axis and is designated as the center microphone; the other two microphones are symmetrically located to the left and right of the center microphone and are designated as the left and right microphones, respectively. Furthermore, a structure is provided inside the microphone housing to secure the three microphones, facilitating the equal-spaced fixing of the left, center, and right microphones in their respective positions and ensuring the stability and accuracy of the three microphones.

[0019] like Figure 1 , Figure 2 As shown, the microphone also includes a signal processing circuit board electrically connected to the left, center, and right microphone cores. The signal processing circuit board includes three operational amplifier circuits, three ADCs (analog-to-digital converters), a DSP (digital signal processor), and a DAC (digital-to-analog converter); the left, center, and right microphone cores are electrically connected to their respective operational amplifier circuits; the three operational amplifier circuits are electrically connected to their respective ADCs; the three ADCs are electrically connected to the DSP; and the DSP is electrically connected to the DAC.

[0020] In practical implementation, the signal processing circuit board uses existing technology to amplify, analyze, and process the sound signals collected by the three microphones. Each microphone is equipped with an independent operational amplifier circuit to avoid crosstalk between channels. The three operational amplifier circuits are high-performance and amplify the analog signals collected by the three microphones, ensuring the signal strength meets the requirements of subsequent processing. Furthermore, the gain of the operational amplifier circuits can be adjusted according to actual needs to ensure the signal strength meets the requirements of subsequent processing. Three ADCs convert the amplified analog signals into digital signals that can be processed by the DSP, and the three independent ADCs ensure phase consistency. The DSP has built-in signal processing algorithms that can perform noise reduction, time delay compensation, and, based on the direction of the sound source, amplify sound signals from the target direction while suppressing sound signals from other directions. The DAC converts the DSP-processed signal back into an analog signal and can drive the speaker to achieve directional enhancement.

[0021] like Figure 3 As shown, the method of the present invention uses a signal processing circuit board and a signal processing algorithm to analyze the analog signals collected by the three microphones. Based on the time difference characteristics of the sound arriving at the three microphones from different directions, the direction of the sound source is determined. Then, based on the direction of the sound source, the sound signal from the target direction (e.g., directly in front of the microphone) is enhanced, while the sound signal from non-target directions (e.g., to the left and right sides of the microphone) is suppressed to obtain the final output signal. The details are as follows.

[0022] S1. The sound from the sound source is collected simultaneously by the left, center and right microphone cores to obtain three sound signals.

[0023] Specifically, such as Figure 2 As shown, for a given sound source, the left, center, and right microphones respectively collect sound from the left, front, and right sides of the microphone, thus obtaining three analog signals. These signals are then sent to the signal processing circuit board, where they are first amplified by the corresponding amplifier circuit, then converted into digital signals by the corresponding ADC, and finally sent to the DSP. The digital signal output by the ADC is, by default, a discrete-time sequence.

[0024] S2. Filter the three audio signals respectively to obtain the left, center and right microphone signals.

[0025] Specifically, such as Figure 2 , Figure 3 As shown, the FFT (Fast Fourier Transform) algorithm is first used to convert the three acquired audio signals from the time domain to the frequency domain. The Fast Fourier Transform (FFT) is a core tool in digital signal processing, used to convert discrete-time signals from the time domain to the frequency domain.

[0026] Then, the 20Hz to 20kHz frequency bands of the three frequency domain signals are retained, and other frequency bands are removed to obtain the filtered frequency domain signals. In practice, the effective frequency band of 20Hz-20kHz (the range audible to the human ear) of the three frequency domain signals is retained, and components outside the target frequency band are directly filtered out by the "zeroing method" (such as setting the coefficients <20Hz or >20kHz in the frequency domain to zero), eliminating meaningless components and avoiding waste of resources. Moreover, compared with time-domain filtering (such as FIR / IIR), frequency-domain filtering can more accurately separate frequency bands and avoid phase distortion.

[0027] The three filtered frequency domain signals are then converted back into time domain signals using the IFFT (Inverse Fast Fourier Transform) algorithm, yielding the left, center, and right microphone signals. At this point, the digital signal output by the DSP has been freed from noise and redundant frequency bands, providing clean audio data for subsequent DAC conversion.

[0028] S3. Based on the time difference and distance difference of the sound reaching the left, center and right microphone cores, calculate the angle between the line connecting the sound source and the center microphone core and the microphone's central axis to obtain the direction of the sound source.

[0029] like Figure 1 , Figure 3 As shown, let the time difference between the sound reaching the middle microphone and the left microphone be . The time difference between the sound reaching the middle microphone and the right microphone is Assuming the speed of sound in air is 340 m / s, the distance difference between the sound reaching the middle microphone core and the left microphone core is 340 m / s. The distance difference between the middle microphone core and the right microphone core is 340. .

[0030] Furthermore, for ease of calculation, the center microphone core is taken as the origin, and the straight lines containing the left, center, and right microphone cores are... The axis, the microphone's center axis is Construct a rectangular coordinate system with axes, and the distance between adjacent microphone cores is . Then the coordinates of the left, middle, and right microphones are respectively (- ,0), (0,0), ( ,0).

[0031] Let the coordinates of the sound source be ( , The angle between the line connecting the sound source and the microphone core (in the direction of the incident sound) and the microphone's central axis (reference direction) is... Based on geometric relationships, the following system of equations can be established: Solve for the coordinates of the sound source in the above system of equations. , ) and included angle and the included angle That is, the direction of the sound source, we get: S4. Based on the left, center, and right microphone signals and the direction of the sound source, the sound signal in the target direction is enhanced through time delay compensation and gain adjustment, while the sound signal in the non-target direction is suppressed to obtain the final output signal.

[0032] Let the time difference between the sound reaching the middle microphone and the left microphone be... The time difference between the sound reaching the middle microphone and the right microphone is , For the current moment, For delay compensation of the left microphone core; To compensate for the time delay of the right microphone; and assuming that the left, middle, and right microphone signals sampled at the current time are respectively , , ,but This is the left microphone signal after time delay compensation. This is the right microphone signal after time delay compensation.

[0033] Furthermore, the weights are adjusted according to the direction of the sound source and the position of the microphone core. , , These are the gain coefficients for the left, middle, and right microphone cores, respectively. If the gain function is the angle, then the final output signal The calculation formula is as follows: Specifically This indicates that the left microphone signal is time-shifted to compensate for the phase difference caused by the sound source deviating from the central axis; This involves time-shifting the right microphone signal to compensate for the phase difference caused by the sound source deviating from the central axis. When the sound source deviates from the central axis, the time difference in arrival time of the sound at the left and right microphones leads to phase mismatch. Time delay compensation aligns the three signals in the time domain through time shifting, satisfying the coherent superposition condition for beamforming. Furthermore, with the compensated information in phase alignment, the signal processing algorithm can effectively enhance the sound in the target direction.

[0034] Furthermore, by adjusting the gain coefficient of each microphone signal, the superposition effect of sound signal amplitudes from different directions can be controlled. It should be noted that this embodiment aims to capture sound directly in front of the microphone and suppresses sound from other directions. Take 1, , All values ​​are set to -0.5. In other situations, where the goal is for the microphone to pick up sound from other directions, the gain coefficient can be adjusted as needed.

[0035] in, The angle gain function is expressed as follows: Specifically, such as Figure 1 As shown, The effective pickup angle threshold can be set to a fixed angle according to product requirements, for example: Set to ±30°. When The absolute value is less than When (i.e., the sound source is within the effective range), then ;when The absolute value is greater than When (i.e., the sound source is outside the effective range), then .

[0036] As shown above, this step solves the phase problem through time delay compensation and the amplitude problem through gain adjustment, thus obtaining the final output signal. It enhances sound signals from specific directions and suppresses sound signals from non-target directions, thereby improving the directionality and sound quality of sound pickup. It is suitable for various occasions that require directional sound pickup, such as meetings, teaching, stage performances, and recording.

[0037] The technical solutions provided by the embodiments of the present invention have been described in detail above. Specific examples have been used to illustrate the principles and implementation methods of the present invention. The description of the above embodiments is only for the purpose of helping to understand the method and core ideas of the present invention. At the same time, for those skilled in the art, based on the ideas of the present invention, modifications can be made to the technical solutions described in the foregoing embodiments, or equivalent substitutions can be made to some of the technical features. These modifications or substitutions do not cause the essence of the corresponding technical solutions to deviate from the ideas and scope of the technical solutions of the embodiments of the present invention.

Claims

1. A method for directional microphone pickup, characterized in that, The microphone used includes left, center, and right microphones; the center microphone is located on the central axis of the microphone, and the left and right microphones are symmetrically distributed on the left and right sides of the center microphone. The sound from the sound source is collected simultaneously by the left, center, and right microphone cores to obtain three sound signals; The three audio signals are filtered separately to obtain the left, center, and right microphone signals; Based on the time difference and distance difference of sound reaching the left, center and right microphones, the angle between the line connecting the sound source and the center microphone and the microphone's central axis is calculated to obtain the direction of the sound source. Based on the left, center, and right microphone signals, as well as the direction of the sound source, the sound signal in the target direction is enhanced and the sound signal in the non-target direction is suppressed through time delay compensation and gain adjustment to obtain the final output signal.

2. The method for directional microphone pickup according to claim 1, characterized in that, The filtering process for the three audio signals to obtain left, center, and right microphone signals includes: The FFT algorithm is used to convert each audio signal from a time-domain signal to a frequency-domain signal; For each frequency domain signal, retain the frequency band from 20Hz to 20kHz to obtain the filtered frequency domain signal; The IFFT algorithm is used to convert each filtered frequency domain signal into a time domain signal to obtain the left, middle and right microphone signals.

3. The method for directional microphone pickup according to claim 2, characterized in that, The method of calculating the angle between the line connecting the sound source and the center microphone and the microphone's central axis based on the time difference and distance difference of sound arrival at the left, center, and right microphones to obtain the sound source direction includes: Let the time difference between the sound reaching the middle microphone and the left microphone be... The time difference between the sound reaching the middle microphone and the right microphone is Assuming the speed of sound in air is 340 m / s, the distance difference between the sound reaching the middle microphone core and the left microphone core is 340 m / s. The distance difference between the middle microphone core and the right microphone core is 340. ; With the center microphone as the origin, the straight lines containing the left, center, and right microphones are... The axis, the microphone's center axis is The axis, and the spacing between adjacent microchannels is Then the coordinates of the left, middle, and right microphones are respectively (- ,0), (0,0), ( ,0); Let the coordinates of the sound source be ( , The angle between the line connecting the sound source and the microphone core and the microphone's central axis is... Establish the following system of equations: ； Solve for the coordinates of the sound source ( , ) and included angle ,in:

4. The method for directional microphone pickup according to claim 3, characterized in that, The method involves enhancing the sound signal from the target direction and suppressing the sound signal from non-target directions based on the left, center, and right microphone signals and the direction of the sound source through time delay compensation and gain adjustment, to obtain the final output signal, including: set up For the current moment, For delay compensation of the left microphone core; To compensate for the time delay of the right microphone, the left, middle, and right microphone signals sampled at the current time are respectively , , ,but This is the left microphone signal after time delay compensation. This is the right microphone signal after time delay compensation; Also set , , These are the gain coefficients for the left, middle, and right microphone cores, respectively. If the gain function is the angle, then the final output signal The calculation formula is as follows:

5. A method for directional microphone pickup according to claim 4, characterized in that, The Take 1, the , All values ​​are -0.

5.

6. A method for directional microphone pickup according to claim 4, characterized in that, The The expression is as follows: ； in, The effective pickup angle threshold.

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