A sound source positioning device based on multi-microphone and time difference algorithm

Abstract

A sound source positioning device based on multi-microphone and time difference algorithm, the device comprises a sound signal acquisition module, a signal processing module, a positioning and communication module, a display module and a power module; the sound signal acquisition module adopts multiple output analog signal microphones to realize omnidirectional acquisition of sound signals; the signal processing module takes a single-chip microcomputer as the core and integrates analog signal amplification, filtering, peak detection, analog-to-digital conversion and time difference algorithm processing functions; the positioning and communication module combines a Beidou positioning module and a mobile communication module to realize sound source position determination and remote data transmission; the power module stably supplies power to each module in multiple gears. The present application combines multi-microphone cooperative acquisition with the time difference algorithm to improve the accuracy and stability of local single-point sound source positioning and can be applied to security monitoring, environmental monitoring, industrial inspection and other scenes.

Description

Technical Field

[0001] This invention belongs to the field of sound source localization and signal processing technology, specifically relating to a sound source localization device based on multiple microphones and time difference algorithm. Background Technology

[0002] Sound source localization technology has wide applications in security monitoring, environmental monitoring, industrial equipment fault diagnosis, and emergency rescue. Traditional sound source localization devices mostly use single-microphone or dual-microphone structures, combined with simple phase difference or intensity difference algorithms, which have problems such as low positioning accuracy, small coverage area, and weak anti-interference ability.

[0003] In existing technologies, some positioning devices combine digital microphones with complex algorithms, but digital microphones are expensive and have strict requirements for communication protocol compatibility. Other devices use multi-sensor networking schemes, but these suffer from system complexity, high power consumption, and unsuitability for local single-point positioning scenarios. In addition, existing devices often suffer from insufficient noise suppression and signal distortion in the analog signal processing stage, affecting the input quality of the positioning algorithm and leading to large positioning errors. At the same time, there is also room for improvement in data transmission and device power supply stability.

[0004] Therefore, there is an urgent need to design a sound source localization device that is simple in structure, cost-controllable, highly accurate in positioning, and adaptable to local single-point positioning scenarios, in order to overcome the shortcomings of existing technologies. Summary of the Invention

[0005] To address the aforementioned issues, this invention provides a sound source localization device based on a multi-microphone and time difference algorithm, which improves the accuracy and stability of local single-point sound source localization and features a compact structure, high portability, and good compatibility.

[0006] To solve the above-mentioned technical problems, the technical solution adopted by the present invention is as follows: A sound source localization device based on multi-microphone and time difference algorithm includes a sound signal acquisition module, a signal processing module, a localization and communication module, a display module, and a power supply module for supplying power to each module, connected in sequence. The acoustic signal acquisition module includes multiple microphones that output analog signals, enabling omnidirectional acquisition of acoustic signals; The signal processing module includes an analog signal processing unit and a digital signal processing unit. It is based on a microcontroller and integrates functions such as analog signal amplification, filtering, peak detection, analog-to-digital conversion, and time difference algorithm processing. The positioning and communication module includes a BeiDou positioning unit and a mobile communication unit, which enables the determination of the sound source location and remote data transmission; The display module uses an LCD display unit, which is connected to the microcontroller via the I2C interface to display the three-dimensional coordinates of the sound source location, the signal spectrum, the microphone working status, the BeiDou positioning information, and the mobile communication status in real time. The power supply module provides stable multi-level power to the sound signal acquisition module, signal processing module, positioning and communication module, and display module.

[0007] The acoustic signal acquisition module includes multiple microphones that output analog signals. The peripheral microphones are evenly distributed in a ring and fixed to the front of the device housing to achieve omnidirectional acoustic signal acquisition and ensure that the monitoring area is covered without blind spots. Each microphone is equipped with independent limiting and amplification circuitry.

[0008] The analog signal processing unit includes a filtering circuit, a peak detection circuit, and a hysteresis comparator circuit. The filter circuit adopts an RC filter structure; The peak detection circuit uses an operational amplifier and diodes to capture the peak value of the input signal, ensuring that the output signal stably presents the peak information. Hysteresis comparator circuits adjust the upper and lower limits by changing the resistance ratio, thereby stabilizing the signal amplitude and determining the threshold, and avoiding false triggering caused by signal fluctuations.

[0009] The digital signal processing unit has a built-in time difference algorithm, and the specific implementation process is as follows: First, the microphone's acquisition time reference is calibrated using the NTP time synchronization mechanism to ensure timestamp consistency; The microcontroller obtains the audio signal acquisition timestamps of each microphone through the analog signal processing unit, calculates the arrival time difference of the audio signal between any two microphones, and removes outliers. By combining the microphone ring installation spacing with the standard sound speed, a spatial positioning model is established, and the three-dimensional coordinates of the sound source are derived through geometric calculations. Finally, the positioning error was corrected through an iterative optimization algorithm.

[0010] The digital signal processing unit also integrates Fast Fourier Transform (FFT) spectrum analysis function to perform frequency domain conversion on the acquired acoustic signal and extract characteristic parameters such as frequency distribution, amplitude peak value and harmonic components, providing data support for sound source type identification.

[0011] The positioning and communication module communicates bidirectionally with the signal processing module.

[0012] The positioning and communication module is equipped with current-limiting resistors and LED indicators to monitor the communication status in real time.

[0013] The Beidou positioning unit communicates with the microcontroller via the UART protocol to obtain the device’s own latitude and longitude location information in real time, providing reference coordinates for sound source localization. The mobile communication unit interacts with the microcontroller through the PA2 and PA3 interfaces, while the PA1 interface is used for positioning and communication module reset control. The positioning and communication module uploads the sound source localization results, spectrum analysis data, and the device's own location information to the IoT cloud platform in real time, supporting multi-terminal network access.

[0014] The power module includes a multi-level voltage conversion circuit and a charging protection circuit. The power module supplies power to the microcontroller, positioning and communication module, and display module through the first conversion circuit. The power module provides negative voltage support to the analog signal processing unit through the second conversion circuit.

[0015] The charging protection circuit is equipped with various capacitors of different specifications for power filtering, ensuring power supply stability and reducing the impact of voltage fluctuations on signal processing.

[0016] The main beneficial effects of this invention are as follows: 1. High positioning accuracy and strong stability: The device adapted to the core requirements of local single-point positioning adopts a multi-microphone ring uniform array design. The outer microphones cover the monitoring area without blind spots. Combined with the core Time Difference of Occurrence (TDOA) algorithm, the acquisition reference is calibrated through NTP time synchronization to ensure the consistency of microphone timestamps. A spatial positioning model is established by combining microphone installation spacing and standard sound velocity. The microphone installation error is compensated through iterative optimization algorithm and software calibration. At the same time, the sound velocity is dynamically corrected according to the ambient temperature, which is superior to traditional single / dual microphone positioning devices. Moreover, the algorithm has an outlier rejection mechanism to avoid invalid data interference, which greatly improves the accuracy and stability of local single-point sound source positioning and solves the core problem of large positioning errors in traditional technologies.

[0017] 2. Excellent sound signal processing performance: The anti-interference capability is significantly improved. The sound signal acquisition end is equipped with independent limiting and amplification circuits for each microphone, which not only avoids signal overload distortion, but also effectively amplifies weak sound signals and improves the signal-to-noise ratio. The analog signal processing unit integrates RC filtering, peak detection, and hysteresis comparison circuits to achieve high-frequency noise suppression, stable signal peak capture, and accurate amplitude threshold judgment at each level, completely solving the problems of insufficient noise suppression, signal distortion, and easy false triggering due to signal fluctuations in the analog signal processing stage of traditional devices. The digital signal processing unit also provides high-purity and high-quality signal input for subsequent algorithm calculations through high-precision acquisition by ADC and microsecond-level storage of system timestamps, thus improving the anti-interference capability of the device throughout the entire chain.

[0018] 3. Highly integrated functions: This device, integrating positioning, analysis, transmission, and visualization capabilities, is centered around a microcontroller and integrates the entire process of sound signal acquisition, processing, positioning, communication, and display. It not only achieves precise three-dimensional coordinate positioning of the sound source but also extracts characteristic parameters such as frequency distribution and peak amplitude of the sound signal through FFT spectrum analysis, providing data support for sound source type identification. The positioning and communication module combines BeiDou positioning and mobile communication to obtain the device's own precise reference coordinates and upload positioning results, spectrum data, and device status to the IoT cloud platform in real time, supporting multi-terminal network access. The display module uses an LCD unit to display the sound source's geographical coordinates, signal spectrum, and the working status of each module in real time, achieving local visualization and remote sharing of data, thus overcoming the shortcomings of traditional devices with limited functionality and weak data interaction capabilities.

[0019] 4. The power supply system is stable and reliable: The power module is designed with multi-level voltage conversion circuits to adapt to various operating scenarios, providing appropriate voltages for microcontrollers, communication modules, display modules, and analog signal processing units. It also has a dedicated negative voltage support for the analog signal processing unit to meet the power supply needs of different modules. The charging protection circuit integrates overcharge, over-discharge, and short-circuit protection functions, and is equipped with multi-specification capacitors for power filtering, effectively reducing the impact of voltage fluctuations on signal processing and ensuring stable power supply for each module. Attached Figure Description

[0020] The present invention will be further described below with reference to the accompanying drawings and embodiments: Figure 1 This is a block diagram of the overall module structure of the sound source localization device of the present invention; Figure 2 This is a schematic diagram of the amplitude limiting circuit of the sound signal acquisition module according to an embodiment of the present invention; Figure 3 This is a schematic diagram of the analog amplifier circuit of the sound signal acquisition module in an embodiment of the present invention; Figure 4 This is a schematic diagram of the peak detection circuit of the signal processing module in an embodiment of the present invention; Figure 5 This is a schematic diagram of the STM32L496 microcontroller circuit in an embodiment of the present invention. Figure 6 This is a schematic diagram of the power module circuit according to an embodiment of the present invention; Figure 7 This is a schematic diagram of the filter circuit according to an embodiment of the present invention; Figure 8 This is a schematic diagram of the hysteresis comparator circuit of the present invention; Figure 9 This is a flowchart of the acoustic signal acquisition and front-end preprocessing process of the present invention; Figure 10 This is a flowchart illustrating the overall operation of the sound source localization device of the present invention. Detailed Implementation

[0021] Example 1: As Figure 1 As shown, a sound source localization device based on multi-microphone and time difference algorithm includes a sound signal acquisition module, a signal processing module, a localization and communication module, a display module, and a power supply module for each module, which are connected in sequence. The sound signal acquisition module includes seven microphones that output analog signals, enabling omnidirectional sound signal acquisition; The signal processing module includes an analog signal processing unit and a digital signal processing unit. It is based on a microcontroller and integrates functions such as analog signal amplification, filtering, peak detection, analog-to-digital conversion, and time difference algorithm processing. The positioning and communication module includes a BeiDou positioning unit and a mobile communication unit, which enables the determination of the sound source location and remote data transmission; The display module uses an LCD display unit, which is connected to the microcontroller's I2C interface to display the three-dimensional coordinates of the sound source location, the signal spectrum, the microphone's working status, BeiDou positioning information, and 4G communication status in real time. The power supply module provides stable multi-level power to the sound signal acquisition module, signal processing module, positioning and communication module, and display module.

[0022] The sound signal acquisition module includes 7 WM-61A microphones that output analog signals. The outer 6 WM-61A microphones are evenly distributed in a ring and fixed to the front of the device housing. The last one is the central microphone. The central angle of the adjacent microphones is 60°, the installation spacing is 5cm, the array diameter is 10cm, and the microphone pickup direction is outward. They are soldered to the independent signal interface on the PCB board and connected to the analog signal processing unit through DuPont wires to achieve omnidirectional sound signal acquisition and ensure that the monitoring area is covered without blind spots. Each WM-61A microphone is equipped with independent limiting and amplification circuitry: like Figure 2 As shown, the limiting circuit consists of a 5.9K resistor, a 0.1uF and a 330nF capacitor, and two Schottky diodes. It limits the amplitude of the weak analog signal output by the microphone to 0-200mV to avoid signal overload distortion and the impact on the circuit after amplification. like Figure 3 As shown, the amplifier circuit uses an LM386 chip, and is configured with a 180Ω resistor and a 10uF capacitor to achieve a magnification factor of 91 times, effectively amplifying weak sound signals and improving the signal-to-noise ratio.

[0023] In the amplifier circuit, the limited signal from the microphone is input to the signal input terminal (IN+) of the LM386 chip via a coupling capacitor. The chip's GAIN pin is configured with a combination of resistors and capacitors to set the amplification parameters. The output terminal (VOUT) outputs the amplified and stable signal. The BYPASS pin is connected to a filter capacitor to reduce power supply noise interference. The attached diagram details the component specifications and pin connections of the amplifier circuit, clearly demonstrating its core role in enhancing the amplitude of weak audio signals and optimizing the signal-to-noise ratio.

[0024] like Figure 5 As shown, the core controller of the signal processing module uses the STM32L496ZGT3 microcontroller. This chip has a high-performance ARM Cortex-M4 core, supports high-speed analog signal processing and complex algorithm calculations. Through the GPIO ports (PB0-PB5) of the STM32L496ZGT3 microcontroller, synchronous trigger signals can be output to control the acquisition timing of the 6 microphones, ensuring that all microphones start signal acquisition at the same time point, with an acquisition synchronization error ≤10μs, laying the foundation for subsequent time difference calculations. The analog signal processing unit includes a filtering circuit, a peak detection circuit, and a hysteresis comparator circuit. like Figure 7 As shown, the filter circuit adopts an RC filter structure, consisting of a 10K resistor and 750pF and 620pF capacitors, which is used to suppress high-frequency environmental noise and ensure signal purity. like Figure 4 As shown, the peak detection circuit uses an operational amplifier and a diode, consisting of an OPA1611ADR operational amplifier, an IN4148W diode, a 1uF capacitor, and a 100K resistor, to capture the peak value of the input signal and ensure that the output signal stably presents the peak information. In the filtering circuit, the filtered analog signal is input to the non-inverting input of the OPA1611ADR operational amplifier. The output of the operational amplifier is connected to the energy storage capacitor via an IN4148W diode, and a bleed resistor is connected in parallel across the capacitor. When the input signal reaches its peak value, the operational amplifier outputs a high level, driving the diode to conduct and charging the capacitor to store the peak voltage. When the signal attenuates, the diode is turned off, and the capacitor slowly discharges through the bleed resistor, maintaining a stable output signal. The attached diagram clearly illustrates the component composition and connection logic of the peak detection circuit, visually demonstrating its functional design of capturing signal peak values ​​and preventing peak loss.

[0025] like Figure 8As shown, the hysteresis comparator circuit uses the TLV3501AIDBVR chip. By changing the resistor ratio, the upper and lower limits are adjusted to achieve signal amplitude stability and threshold judgment, avoiding false triggering caused by signal fluctuations. The filtered signal is input to the TLV3501AIDBVR hysteresis comparator. Its +IN pin is connected to the signal input, and the -IN pin is connected to the threshold adjustment circuit through a 16.5KΩ resistor and a 1KΩ adjustable resistor to set the lower threshold to 0.31V and the upper threshold to 0.51V. When the input signal is higher than 0.51V or lower than 0.31V, the hysteresis comparator outputs a high level (3.3V) or a low level (0V) to avoid missampling caused by small signal fluctuations.

[0026] The digital signal processing unit has a built-in time difference algorithm, and the specific implementation process is as follows: First, the acquisition time reference of the seven microphones is calibrated through the NTP time synchronization mechanism to ensure timestamp consistency; The microcontroller connects to the IoT cloud platform via a 4G communication module and calls the NTP time synchronization interface to synchronize the microcontroller's system time every 30 seconds, with a synchronization error of ≤1ms. After synchronization, the microcontroller records the current system time as a timestamp reference.

[0027] The system timestamp is read and stored in the corresponding microphone's data acquisition structure, with the ADC interrupt trigger time as the signal arrival time. The timestamp format is "year-month-day-hour-minute-second-microsecond", and the storage precision is ≤1μs to ensure that the timestamps of the 6 signals can be compared.

[0028] Let the timestamps of the 6 microphones be respectively Using microphone 0 as the reference, calculate the time difference between the other 5 microphones and the reference microphone. Remove outliers and retain valid time difference data; Outliers are time differences that exceed the range of 3σ, where σ is the standard deviation. By combining the microphone ring installation spacing with the standard sound speed, a spatial positioning model is established, and the three-dimensional coordinates of the sound source are derived through geometric calculations. In this embodiment, the radius of the microphone ring array is r=5cm, the standard speed of sound is v=340m / s, and it can be dynamically corrected by software according to the ambient temperature. For every 1℃ change in temperature, the speed of sound is corrected by ±0.6m / s. The latitude and longitude coordinates of the device itself are provided by the Beidou module, with a positioning accuracy of ±2m. Establish a two-dimensional coordinate system with the reference microphone 0 as the origin, and based on the time difference Calculate the distance difference from the sound source to each microphone. By combining the microphone's coordinates, a system of equations is established using the hyperbolic positioning principle:

[0029] The least squares method was used to solve the system of equations, with 10 iterations. The convergence condition was that the coordinate deviation between two iterations was ≤0.01m. Correcting microphone installation errors: Through software calibration, the positional deviation compensation value for each microphone is pre-stored in Flash; Finally, the relative coordinates of the sound source are combined with the latitude and longitude of the device itself to convert them into absolute geographic coordinates, ensuring that the positioning error is less than 0.1 meters.

[0030] The digital signal processing unit also integrates Fast Fourier Transform (FFT) spectrum analysis function to perform frequency domain conversion on the acquired acoustic signal and extract characteristic parameters such as frequency distribution, amplitude peak value and harmonic components, providing data support for sound source type identification.

[0031] The positioning and communication module communicates bidirectionally with the signal processing module.

[0032] The positioning and communication module is equipped with a 100Ω current-limiting resistor and LED indicator to monitor the communication status in real time.

[0033] The Beidou positioning unit communicates with the STM32L496ZGT3 microcontroller via the UART protocol to obtain the device’s own latitude and longitude location information in real time, providing reference coordinates for sound source positioning. The 4G communication unit interacts with the microcontroller through the PA2 (4G_RXD) and PA3 (4G_TXD) interfaces, while the PA1 interface is used for positioning and communication module reset control. The positioning and communication module uploads the sound source localization results, spectrum analysis data, and the device's own location information to the IoT cloud platform in real time, supporting multi-terminal network access.

[0034] The display module uses an LCD display unit, which is connected to the I2C interface of the STM32L496ZG microcontroller. The LCD display unit uses a 4.3-inch TFT display screen (resolution 480×272), and communicates with the STM32L496ZG microcontroller via an I2C interface (SCL=PB6, SDA=PB7) at a communication rate of 100kHz. Real-time display of the absolute geographic coordinates of the sound source (longitude accurate to 0.0001°, latitude accurate to 0.0001°), update frequency 2Hz; Plot a spectrum (horizontal axis: frequency 0-5kHz, vertical axis: amplitude 0-60dB), refreshed every 500ms; The device displays the microphone's operating status (whether the signal amplitude of the 6 microphones is within the normal range of 0.31-0.51V), BeiDou positioning status (positioning successful / failed), and 4G communication status (connection normal / disconnected), allowing users to easily view the device's operating data on-site.

[0035] The power module includes a multi-level voltage conversion circuit and a charging protection circuit. like Figure 6 As shown, in this embodiment, the power module is a 3.7V lithium battery. The 3.7V lithium battery is converted to 5V voltage through a 3.7V to 5V circuit, and then to 3.3V voltage through a 5V to 3.3V circuit (using an HT7833 chip) to power the microcontroller, positioning and communication module and display module. The 3.7V lithium battery is converted to 5V voltage through a 3.7V to 5V circuit, and then the 5V to -5V circuit provides negative voltage support for the analog signal processing unit. The charging protection circuit uses the TP5C00X4 chip to realize overcharge, over-discharge, and short circuit protection for lithium batteries. In addition, the charging protection circuit is equipped with capacitors of specifications such as 100nF, 1uF, and 220uF for power supply filtering to ensure power supply stability and reduce the impact of voltage fluctuations on signal processing.

Claims

1. A sound source localization device based on multi-microphone and time difference algorithm, characterized in that: It includes a sound signal acquisition module, a signal processing module, a positioning and communication module, a display module, and a power supply module that supplies power to each module, connected in sequence. The acoustic signal acquisition module includes multiple microphones that output analog signals, enabling omnidirectional acquisition of acoustic signals; The signal processing module includes an analog signal processing unit and a digital signal processing unit. It is based on a microcontroller and integrates functions such as analog signal amplification, filtering, peak detection, analog-to-digital conversion, and time difference algorithm processing. The positioning and communication module includes a BeiDou positioning unit and a mobile communication unit, which enables the determination of the sound source location and remote data transmission; The display module uses an LCD display unit, which is connected to the microcontroller via the I2C interface to display the three-dimensional coordinates of the sound source location, the signal spectrum, the microphone working status, the BeiDou positioning information, and the mobile communication status in real time. The power supply module provides stable multi-level power to the sound signal acquisition module, signal processing module, positioning and communication module, and display module.

2. The sound source localization device based on multi-microphone and time difference algorithm according to claim 1, characterized in that: The acoustic signal acquisition module includes multiple microphones that output analog signals. The peripheral microphones are evenly distributed in a ring and fixed to the front of the device housing to achieve omnidirectional acoustic signal acquisition and ensure that the monitoring area is covered without blind spots. Each microphone is equipped with independent limiting and amplification circuitry.

3. The sound source localization device based on multi-microphone and time difference algorithm according to claim 1, characterized in that: The analog signal processing unit includes a filtering circuit, a peak detection circuit, and a hysteresis comparator circuit. The filter circuit adopts an RC filter structure; The peak detection circuit uses an operational amplifier and diodes to capture the peak value of the input signal, ensuring that the output signal stably presents the peak information. Hysteresis comparator circuits adjust the upper and lower limits by changing the resistance ratio, thereby stabilizing the signal amplitude and determining the threshold, and avoiding false triggering caused by signal fluctuations.

4. The sound source localization device based on multi-microphone and time difference algorithm according to claim 1, characterized in that: The digital signal processing unit has a built-in time difference algorithm, and the specific implementation process is as follows: First, the microphone's acquisition time reference is calibrated using the NTP time synchronization mechanism to ensure timestamp consistency; The microcontroller obtains the audio signal acquisition timestamps of each microphone through the analog signal processing unit, calculates the arrival time difference of the audio signal between any two microphones, and removes outliers. By combining the microphone ring installation spacing with the standard sound speed, a spatial positioning model is established, and the three-dimensional coordinates of the sound source are derived through geometric calculations. Finally, the positioning error was corrected through an iterative optimization algorithm.

5. A sound source localization device based on multi-microphone and time difference algorithm according to claim 4, characterized in that: The digital signal processing unit also integrates Fast Fourier Transform (FFT) spectrum analysis function to perform frequency domain conversion on the acquired acoustic signal and extract characteristic parameters such as frequency distribution, amplitude peak value and harmonic components, providing data support for sound source type identification.

6. The sound source localization device based on multi-microphone and time difference algorithm according to claim 1, characterized in that: The positioning and communication module communicates bidirectionally with the signal processing module.

7. A sound source localization device based on multi-microphone and time difference algorithm according to claim 6, characterized in that: The positioning and communication module is equipped with current-limiting resistors and LED indicators to monitor the communication status in real time.

8. A sound source localization device based on multi-microphone and time difference algorithm according to claim 7, characterized in that: The Beidou positioning unit communicates with the microcontroller via the UART protocol to obtain the device’s own latitude and longitude location information in real time, providing reference coordinates for sound source localization. The mobile communication unit interacts with the microcontroller through the PA2 and PA3 interfaces, while the PA1 interface is used for positioning and communication module reset control. The positioning and communication module uploads the sound source localization results, spectrum analysis data, and the device's own location information to the IoT cloud platform in real time, supporting multi-terminal network access.

9. A sound source localization device based on multi-microphone and time difference algorithm according to claim 1, characterized in that: The power module includes a multi-level voltage conversion circuit and a charging protection circuit. The power module supplies power to the microcontroller, positioning and communication module, and display module through the first conversion circuit. The power module provides negative voltage support to the analog signal processing unit through the second conversion circuit.

10. A sound source localization device based on a multi-microphone and time difference algorithm according to claim 9, characterized in that: The charging protection circuit is equipped with various capacitors of different specifications for power filtering, ensuring power supply stability and reducing the impact of voltage fluctuations on signal processing.

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