Underground magnetic induction voice communication system based on Zynq architecture

Abstract

The application provides an underground magnetic induction voice communication system based on a Zynq architecture, belongs to the underground magnetic induction communication field, and constructs a complete software and hardware cooperative processing platform, which comprises a Zynq chip, a voice module, a radio frequency module and an automatic alignment module; the application makes full use of the advantages of the heterogeneous platform of the Zynq chip, and efficiently executes the complex system control and signal processing tasks on the PS end and the PL end respectively; at the same time, the radio frequency module adopts a direct sampling architecture, which not only effectively improves the data processing efficiency, but also makes the system highly integrated to meet the requirements of portable devices; on the basis, the automatic alignment module can automatically ensure that the coil always keeps in the best magnetic field coupling in the working process of the system, and improves the reliability of the communication link. Based on this, the application realizes a magnetic induction voice communication system which is specially designed for the underground complex environment, and has high reliability, high integration and high convenience.

Description

Technical Field

[0001] This invention belongs to the field of underground magnetic induction communication technology, and more specifically, relates to an underground magnetic induction voice communication system based on the Zynq architecture. Background Technology

[0002] In underground emergency scenarios such as earthquakes, mine collapses, and tunnel accidents, establishing a stable and reliable communication link is crucial for rapid and effective rescue operations and improving the survival rate of trapped personnel. It is the "lifeline" for ensuring the safety of trapped personnel and the "information line" for implementing scientific rescue.

[0003] In above-ground environments, wired and electromagnetic wave communication are common and mature technologies. However, they face challenges in underground emergency communication: wired communication is costly to deploy and prone to breakage during complex construction, geological disasters, or mine collapses, leading to communication interruptions; electromagnetic wave communication suffers from severe signal attenuation and time-varying channel instability due to the high conductivity of underground media, and the applicable low-frequency signals require large antennas, making effective deployment in underground environments difficult. Therefore, traditional wired and radio electromagnetic wave communication cannot meet the reliability, deployability, and low-cost requirements of underground emergency communication. In contrast, magnetic induction communication, with its low sensitivity to underground media, stable and predictable channel response, strong penetration, strong anti-interference ability, and flexible adjustment of coil antenna size, has become the best choice to solve the problem of underground emergency communication, ensuring the reliability and robustness of the link.

[0004] However, current research on underground magnetic induction communication mainly focuses on basic work such as model building, theoretical derivation, and experimental verification. The design and engineering implementation of practical systems still face many challenges: 1) Existing research mostly focuses on building simple point-to-point underground magnetic induction communication systems around a single objective, lacking comprehensive design for practical applications. For example, when analyzing the channel capacity and bit error rate of underground magnetic induction communication, only system models are built under ideal conditions for theoretical derivation, without involving the implementation of actual systems. When verifying communication distance, simple platforms consisting of signal sources, transceiver coils, and spectrum analyzers are often built in the laboratory, without involving the transmission of effective data; even if there are systems that can transmit data, most of them only support the transmission of a small amount of sensor data, using simple 2-ASK modulation, lacking mechanisms such as channel coding to improve communication performance. Especially in the area of ​​underground magnetic induction voice communication, there is no practical system that can support high-quality voice transmission, and there is a severe lack of mature equipment support. The overall system development is still in the exploratory stage. 2) The system has weak anti-interference capabilities and low link transmission rates. The transceiver coils rely on manual adjustment of their direction, which is inefficient and cannot guarantee that the coils maintain optimal magnetic field coupling, affecting the reliability of the link; 3) The system has low integration and poor portability. Existing equipment is mostly built from discrete components, resulting in a large equipment size, which makes it difficult to meet the miniaturization requirements of scenarios such as underground emergency rescue.

[0005] In summary, there is an urgent need to design a magnetic induction voice communication system that is specifically designed for complex underground environments and possesses high reliability, high integration, and high convenience. Summary of the Invention

[0006] In view of the above-mentioned defects or improvement needs of the existing technology, the present invention provides an underground magnetic induction voice communication system based on the Zynq architecture. The purpose is to provide a magnetic induction voice communication system with high reliability, high integration and high convenience, specifically designed for complex underground environments.

[0007] To achieve the above objectives, this invention provides an underground magnetic induction voice communication system based on the Zynq architecture, comprising: a Zynq chip, a voice module, a radio frequency module, and an automatic alignment module; wherein, the PS terminal of the Zynq chip integrates a main control unit, and the PL terminal integrates a polarization encoder, a polarization decoder, a modulator, and a demodulator; the voice module includes: a voice acquisition unit, a voice output unit, and an audio compiler; the automatic alignment module includes: a coil antenna and a signal strength detector; the radio frequency module includes: a DAC unit, an ADC unit, a first conditioning circuit, a second conditioning circuit, and a power amplifier;

[0008] Before the system performs voice signal transmission or reception tasks, the main control unit controls the coil antenna to rotate at a constant speed until it covers all preset postures, and records the posture of the coil antenna in real time. At the same time, it acquires the signal strength received by the coil antenna as detected by the signal strength detector; and fixes the posture of the coil antenna at the posture corresponding to the maximum signal strength.

[0009] When the system performs the voice signal transmission task, the audio compiler converts the voice signal collected by the voice acquisition unit into a digital voice signal; the digital voice signal is then polarized and modulated by a polar encoder and a modulator to obtain a baseband signal; the baseband signal is then converted from digital to analog by a DAC unit, and then conditioned and amplified by a first conditioning circuit and a power amplifier; the amplified signal is then transmitted through a coil antenna.

[0010] When the system performs the voice signal reception task, the voice signal received by the coil antenna is conditioned and converted from analog to digital by the second conditioning circuit and the ADC unit in sequence, and then converted into a baseband signal. The baseband signal is demodulated and polarized decoded by the demodulator and polarization decoder in sequence to recover the digital voice signal. The digital voice signal is converted into an analog voice signal by the audio compiler and then output by the voice output unit.

[0011] More preferably, the above-mentioned magnetic induction voice communication system operates in full-duplex mode; the center frequencies of the filters in the first conditioning circuit and the second conditioning circuit are different, but the bandwidths are the same;

[0012] The aforementioned automatic alignment module also includes: a parasitic coil with the same number of turns as the coil antenna; the parasitic coil and the coil antenna have parallel cross-sections and are placed coaxially to form a magnetic antenna with a parallel parasitic structure;

[0013] Resistance of coil antenna and the axial spacing between the coil antenna and the parasitic coil. These are the resistance and spacing values ​​that maximize the bandwidth of the magnetic antenna, selected from their respective preset candidate sets through optimization algorithms.

[0014] The expression for the magnetic antenna bandwidth is:

[0015]

[0016] in, and For the equation The two real roots of ; The operating angular frequency of the coil antenna The transmit power at the given time is expressed as:

[0017]

[0018] in, The impedance of the parasitic coil, ; The resistance of the parasitic coil; j The imaginary unit; The inductance of the parasitic coil; The capacitance of the parasitic coil; This represents the voltage amplitude of the coil antenna; The impedance of the coil antenna, ; The inductance of the coil antenna; The capacitance of the coil antenna; ; Permeability of free space; The number of turns of the coil antenna or parasitic coil; Where is the coil radius of the coil antenna; The radius of the parasitic coil is denoted as .

[0019] More preferably, the above-mentioned magnetic induction voice communication system operates in half-duplex mode; the center frequencies and bandwidths of the filters in the first conditioning circuit and the second conditioning circuit are the same;

[0020] The radio frequency module also includes a T / R switch, which connects the power amplifier and the coil antenna when the system performs a voice signal transmission task, at which time the second conditioning circuit is disconnected from the coil antenna; and connects the second conditioning circuit and the coil antenna when the system performs a voice signal reception task, at which time the power amplifier and the coil antenna are disconnected.

[0021] More preferably, the above-mentioned polarization encoder performs polarization encoding on the digital speech signal in the following manner:

[0022] Will Channel merging is performed on 1 independent but differently distributed sub-channels to obtain a channel of length 1. The vector channel; where the transfer function of the vector channel is... ; and They represent The input and output sequences of this transmission; It is the first i The transfer function of each sub-channel; The code length is the polar code length.

[0023] Channel decomposition is performed on the vector channel to obtain There are 3 complex polarization sub-channels; among which, the 1st... i The transfer function of the complex polarization subchannel is:

[0024]

[0025] in, This represents the input bit sequence of the polarization encoder; for The first in i 1 bit; for The subsequence is middle Previous i A sequence consisting of -1 bits; for The subsequence of represents middle After A sequence consisting of bits; For length is The set consisting of all binary sequences.

[0026] Calculate the Parsons parameters for each complex polarization subchannel;

[0027] The Bartholomew's parameters of each complex polarization subchannel are sorted, and the indices of the complex polarization subchannels corresponding to the top NK Bartholomew's parameters with the largest values ​​are used as frozen bit indices, while the indices of the complex polarization subchannels corresponding to the remaining Bartholomew's parameters are used as information bit indices; K is the preset number of information bits.

[0028] The transmit bits under the freeze bit index are set to 0 and the transmit bits under the information bit index are set to 1 in order to encode the digital speech signal and obtain the encoding vector. The encoding vector is then multiplied by the polar code generator matrix GN of length N to obtain the polar coding result of the digital speech signal.

[0029] More preferably, the main control unit is provided with a mapping table; the mapping table includes: the signal-to-noise ratio of the system channel. The correspondence between modulation methods; modulation methods include: the first to the fourth modulation methods; the first modulation method is BPSK modulation; the second modulation method is 4-QAM modulation; the third modulation method is 16-QAM modulation; the fourth modulation method is 64-QAM modulation; when When, the corresponding modulation method is BPSK modulation; when When, the corresponding modulation method is 4-QAM modulation; when When, the corresponding modulation method is 16-QAM modulation; when At this time, the corresponding modulation method is 64-QAM modulation; All are switching thresholds;

[0030] The above It is calculated in the following way:

[0031] The system is constructed such that its average bit error rate is less than or equal to a preset average bit error rate. The objective function is defined by constraints to maximize the spectral efficiency of the system.

[0032] The objective function mentioned above is:

[0033]

[0034] The constraints are:

[0035]

[0036] in, For the first i The rate of each modulation scheme; It is denoted as infinity; Let be the probability distribution function of the system channel signal-to-noise ratio; The bit error rate function of the system;

[0037] The Lagrange algorithm is used to solve the above objective function, and the above results are obtained. ;

[0038] The main control unit is also used to perform channel estimation on the baseband signal input to the demodulator when the system is performing voice signal reception tasks, and then calculate the signal-to-noise ratio of the current system channel; determine the modulation method corresponding to the signal-to-noise ratio of the current system channel based on the mapping table, and make the demodulator adopt the corresponding demodulation method, while feeding back to the transmitter so that the modulator in the transmitter adopts the modulation method.

[0039] More preferably, the main control unit is connected to the audio compiler via an I2C interface; the main control unit is connected to the polar encoder, polar decoder, modulator, and demodulator via an AXI bus interface; the audio compiler is connected to the polar encoder and polar decoder via an I2S interface; and the coil antenna is connected to the RF module via a coaxial cable.

[0040] More preferably, the above-mentioned voice communication system is mounted on the gimbal; the main control unit is used to control the gimbal to drive the coil antenna to rotate at a constant speed before the system performs voice signal transmission or reception tasks.

[0041] More preferably, the main control unit uses the Linux operating system.

[0042] More preferably, the voice acquisition unit is a microphone; the voice output unit is a speaker.

[0043] In summary, the above-described technical solutions conceived in this invention can achieve the following beneficial effects:

[0044] 1. This invention provides an underground magnetic induction voice communication system based on the Zynq architecture. Using a Zynq architecture chip as the core, a complete hardware and software co-processing platform is constructed, including: a Zynq chip, a voice module, an RF module, and an automatic alignment module. The Zynq chip's PS end integrates a main control unit, while the PL end integrates a polarization encoder, polarization decoder, modulator, and demodulator. This invention fully utilizes the advantages of the Zynq chip's heterogeneous platform, efficiently executing complex system control and signal processing tasks on the PS and PL ends respectively. Simultaneously, the RF module adopts a direct sampling architecture, eliminating the need for multiple components such as local oscillators, mixers, and intermediate frequency filters required in superheterodyne architectures. This not only effectively improves data processing efficiency and ensures real-time voice communication but also significantly reduces system complexity, cost, and size, enabling a highly integrated system to meet the requirements of portable devices. Furthermore, the automatic alignment module automatically ensures that the coil maintains optimal magnetic field coupling during system operation, improving the reliability of the communication link. Based on this, the present invention realizes a magnetic induction voice communication system that is designed specifically for complex underground environments, and has high reliability, high integration and high convenience.

[0045] 2. Furthermore, the underground magnetic induction voice communication system based on the Zynq architecture provided by this invention can expand the system's communication modes by simply adjusting the components of the radio frequency module, enabling the system to flexibly switch between full-duplex and half-duplex. This design provides different solutions for different application scenarios while keeping the core signal processing algorithm unchanged, thus enhancing the flexibility of the system architecture.

[0046] 3. Furthermore, the underground magnetic induction voice communication system based on the Zynq architecture provided by this invention, through a parallel passive parasitic coil structure, induces the coil to generate a continuous and flat extended bandwidth, providing the necessary spectrum resources for frequency division duplexing. This enables both communicating parties to simultaneously send and receive voice signals, achieving full-duplex voice communication and improving the system's communication efficiency and interactive experience.

[0047] 4. Furthermore, the underground magnetic induction voice communication system based on the Zynq architecture provided by this invention, through an automatic coil alignment mechanism and a polar code construction method based on the underground magnetic induction channel, endows the system with extremely strong error correction capabilities while ensuring optimal coupling efficiency at the physical level, significantly reducing the bit error rate. Combined with a CSI-based adaptive modulation and demodulation method, the transmission rate is maximized while ensuring system reliability. The synergistic effect of these three elements enables the communication link to exhibit excellent robustness and stability in the face of complex and time-varying underground media environments, effectively extending the system's communication distance and ensuring communication performance. Attached Figure Description

[0048] Figure 1 A schematic diagram of the structure of an underground magnetic induction voice communication system based on the Zynq architecture provided in an embodiment of the present invention;

[0049] Figure 2 A flowchart of underground magnetic induction voice communication based on Zynq architecture is provided for embodiments of the present invention.

[0050] Figure 3 This is a schematic diagram of underground magnetic induction channel polarization provided in an embodiment of the present invention;

[0051] Figure 4 This is a schematic diagram of an adaptive modulation and demodulation model based on underground magnetic induction CSI provided in an embodiment of the present invention;

[0052] Figure 5 The diagram illustrates the bandwidth extension effect of a magnetic antenna with a parallel parasitic structure, provided for an embodiment of the present invention. Detailed Implementation

[0053] To make the objectives, technical solutions, and advantages of this invention clearer, the invention will be further described in detail below with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative and not intended to limit the invention. Furthermore, the technical features involved in the various embodiments of this invention described below can be combined with each other as long as they do not conflict with each other.

[0054] To achieve the above objectives, this invention provides an underground magnetic induction voice communication system based on the Zynq architecture, comprising: a Zynq chip, a voice module, a radio frequency module, and an automatic alignment module; wherein, the PS terminal of the Zynq chip integrates a main control unit, and the PL terminal integrates a polarization encoder, a polarization decoder, a modulator, and a demodulator; the voice module includes: a voice acquisition unit, a voice output unit, and an audio compiler; the automatic alignment module includes: a coil antenna and a signal strength detector; the radio frequency module includes: a DAC unit, an ADC unit, a first conditioning circuit, a second conditioning circuit, and a power amplifier;

[0055] Before the system performs voice signal transmission or reception tasks, the main control unit controls the coil antenna to rotate at a constant speed until it covers all preset postures, and records the posture of the coil antenna in real time. At the same time, it acquires the signal strength received by the coil antenna as detected by the signal strength detector; and fixes the posture of the coil antenna at the posture corresponding to the maximum signal strength.

[0056] When the system performs the voice signal transmission task, the audio compiler converts the voice signal collected by the voice acquisition unit into a digital voice signal; the digital voice signal is then polarized and modulated by a polar encoder and a modulator to obtain a baseband signal; the baseband signal is then converted from digital to analog by a DAC unit, and then conditioned and amplified by a first conditioning circuit and a power amplifier; the amplified signal is then transmitted through a coil antenna.

[0057] When the system performs the voice signal reception task, the voice signal received by the coil antenna is conditioned and converted from analog to digital by the second conditioning circuit and the ADC unit in sequence, and then converted into a baseband signal. The baseband signal is demodulated and polarized decoded by the demodulator and polarization decoder in sequence to recover the digital voice signal. The digital voice signal is converted into an analog voice signal by the audio compiler and then output by the voice output unit.

[0058] In one alternative implementation, the above-described magnetic induction voice communication system operates in full-duplex mode; the center frequencies of the filters in the first conditioning circuit and the second conditioning circuit are different, but the bandwidths are the same.

[0059] The aforementioned automatic alignment module also includes: a parasitic coil with the same number of turns as the coil antenna; the parasitic coil and the coil antenna have parallel cross-sections and are placed coaxially to form a magnetic antenna with a parallel parasitic structure;

[0060] Resistance of coil antenna and the axial spacing between the coil antenna and the parasitic coil. These are the resistance and spacing values ​​that maximize the bandwidth of the magnetic antenna, selected from their respective preset candidate sets through optimization algorithms.

[0061] The expression for the magnetic antenna bandwidth is:

[0062]

[0063] in, and For the equation The two real roots of ; The operating angular frequency of the coil antenna The transmit power at the given time is expressed as:

[0064]

[0065] in, The impedance of the parasitic coil, ; The resistance of the parasitic coil; j The imaginary unit; The inductance of the parasitic coil; The capacitance of the parasitic coil; This represents the voltage amplitude of the coil antenna; The impedance of the coil antenna, ; The inductance of the coil antenna; The capacitance of the coil antenna; ; Permeability of free space; The number of turns of the coil antenna or parasitic coil; Where is the coil radius of the coil antenna; The radius of the parasitic coil is denoted as .

[0066] It should be noted that there are various optimization algorithms that can be used, such as sequential quadratic programming, continuous quadratic programming, interior point method, augmented Lagrangian function method, etc. There is no limitation here, but sequential quadratic programming is preferred.

[0067] In one alternative implementation, the magnetic induction voice communication system operates in half-duplex mode; the center frequencies and bandwidths of the filters in the first conditioning circuit and the second conditioning circuit are the same.

[0068] The radio frequency module also includes a T / R switch, which connects the power amplifier and the coil antenna when the system performs a voice signal transmission task, at which time the second conditioning circuit is disconnected from the coil antenna; and connects the second conditioning circuit and the coil antenna when the system performs a voice signal reception task, at which time the power amplifier and the coil antenna are disconnected.

[0069] Preferably, in an optional embodiment, the polarization encoder performs polarization encoding on the digital speech signal in the following manner:

[0070] Will Channel merging is performed on 1 independent but differently distributed sub-channels to obtain a channel of length 1. The vector channel; where the transfer function of the vector channel is... ; and They represent The input and output sequences of this transmission; It is the first i The transfer function of each sub-channel; The code length is the polar code length.

[0071] Channel decomposition is performed on the vector channel to obtain There are 3 complex polarization sub-channels; among which, the 1st... i The transfer function of the complex polarization subchannel is:

[0072]

[0073] in, This represents the input bit sequence of the polarization encoder; for The first in i 1 bit; for The subsequence is middle Previous i A sequence consisting of -1 bits; for The subsequence of represents middle After A sequence consisting of bits; For length is The set consisting of all binary sequences.

[0074] Calculate the Parsons parameters for each complex polarization subchannel;

[0075] The Bartholomew's parameters of each complex polarization subchannel are sorted, and the indices of the complex polarization subchannels corresponding to the top NK Bartholomew's parameters with the largest values ​​are used as frozen bit indices, while the indices of the complex polarization subchannels corresponding to the remaining Bartholomew's parameters are used as information bit indices; K is the preset number of information bits.

[0076] The transmit bits under the freeze bit index are set to 0 and the transmit bits under the information bit index are set to 1 in order to encode the digital speech signal and obtain the encoding vector. The encoding vector is then multiplied by the polar code generator matrix GN of length N to obtain the polar coding result of the digital speech signal.

[0077] Preferably, in one optional implementation, the main control unit is provided with a mapping table; the mapping table includes: the signal-to-noise ratio of the system channel. The correspondence between modulation methods; modulation methods include: the first to the fourth modulation methods; the first modulation method is BPSK modulation; the second modulation method is 4-QAM modulation; the third modulation method is 16-QAM modulation; the fourth modulation method is 64-QAM modulation; when When, the corresponding modulation method is BPSK modulation; when When, the corresponding modulation method is 4-QAM modulation; when When, the corresponding modulation method is 16-QAM modulation; when At this time, the corresponding modulation method is 64-QAM modulation; All are switching thresholds;

[0078] The above It is calculated in the following way:

[0079] The system is constructed such that its average bit error rate is less than or equal to a preset average bit error rate. The objective function is defined by constraints to maximize the spectral efficiency of the system.

[0080] The objective function mentioned above is:

[0081]

[0082] The constraints are:

[0083]

[0084] in, For the first i The rate of each modulation scheme; It is denoted as infinity; For system channel signal-to-noise ratio

[0085] The probability distribution function of the ratio; The bit error rate function of the system;

[0086] The Lagrange algorithm is used to solve the above objective function, and the above results are obtained. ;

[0087] The main control unit is also used to perform channel estimation on the baseband signal input to the demodulator when the system is performing voice signal reception tasks, and then calculate the signal-to-noise ratio of the current system channel; determine the modulation method corresponding to the signal-to-noise ratio of the current system channel based on the mapping table, and make the demodulator adopt the corresponding demodulation method, while feeding back to the transmitter so that the modulator in the transmitter adopts the modulation method.

[0088] In one alternative implementation, the main control unit is connected to the audio compiler via an I2C interface; the main control unit is connected to the polar encoder, polar decoder, modulator, and demodulator via an AXI bus interface; the audio compiler is connected to the polar encoder and polar decoder via an I2S interface; and the coil antenna is connected to the RF module via a coaxial cable.

[0089] In one alternative implementation, the aforementioned voice communication system is mounted on a gimbal; the main control unit is used to control the gimbal to rotate the coil antenna at a constant speed before the system performs voice signal transmission or reception tasks.

[0090] In one alternative implementation, the main control unit uses the Linux operating system.

[0091] In one alternative implementation, the voice acquisition unit is a microphone; the voice output unit is a speaker.

[0092] To further illustrate the underground magnetic induction voice communication system based on the Zynq architecture provided by this invention, a specific embodiment is described in detail below:

[0093] This embodiment provides an underground magnetic induction voice communication system based on the Zynq architecture, constructing a complete hardware and software collaborative processing platform that supports full-duplex / half-duplex communication modes. Specifically, as shown... Figure 1 As shown, this system uses the Zynq UltraScale+ MPSoC XCZU3EG as the core controller and includes 5 main modules. The specific composition and connection of each module are as follows:

[0094] The PS-side Linux interactive module (main control unit) is the "brain" of the system. It includes a Linux operating system, custom drivers, applications, and an LCD screen. It is responsible for system control, protocol stack, and human-computer interaction (such as driving the display) and other upper-layer applications, ensuring coordinated operation between all system modules. Specifically, a Linux operating system is run on the ARM Cortex-A53 core of the XCZU3EG PS, responsible for system control, protocol stack, and human-computer interaction interface, ensuring coordinated operation between all system modules. It should be noted that the Linux operating system in this embodiment is customized using the PetaLinux tool, drivers are loaded into the device tree to identify underlying hardware peripherals, and the interactive interface is developed using the QT tool.

[0095] The automatic alignment module, including a coil antenna and a signal strength detector, is responsible for ensuring real-time alignment between the transceiver coils before the system performs voice signal transmission or reception tasks, thereby guaranteeing effective coupling between them. Specifically, it controls the coil antenna to rotate at a constant speed until it covers all preset postures, and records the coil antenna's posture in real time. Simultaneously, it acquires the signal strength received by the coil antenna as detected by the signal strength detector. The coil antenna's posture is then fixed at the posture corresponding to the maximum signal strength. In this embodiment, the signal strength detector is an RSSI detection circuit. The RSSI detection circuit is implemented using Verilog programming and transmits the data to the PS terminal.

[0096] The PL-side signal processing module includes a polar encoder, polar decoder, modulator, and demodulator designed for underground environments. It is responsible for encoding and decoding the baseband signal, constellation mapping, and demapping. Upon receiving a digital speech signal from the audio compiler, it performs polar encoding, then maps it into symbols via the modulator before transmitting it to the RF module. Upon receiving a baseband signal from the RF module, it performs demapping and then restores the digital speech signal through polar decoding. More specifically, this embodiment uses Verilog programming to implement the above functions. The polar encoder employs a design with independent and unequal distribution of sub-channels, supporting code lengths... The code rate is adjustable; the decoder uses an SCL decoder; the modulator / demodulator supports adaptive switching between BPSK, 4-QAM, 16-QAM and 64-QAM modulation modes.

[0097] The voice module includes a voice acquisition unit, a voice output unit, and an audio compiler. In this embodiment, the voice acquisition unit is a microphone; the voice output unit is a speaker; the audio compiler is responsible for analog-to-digital conversion of the voice signal. When it receives an analog voice signal from the microphone, it converts it into a digital voice signal; when it receives a digital voice signal from the polarization decoder, it converts it into an analog voice signal to drive the speaker. More specifically, this embodiment uses an analog MEMS microphone (sensitivity of -26dBV) and a speaker with a rated power of 1W and an impedance of 8Ω; the audio compiler uses the TLV320AIC23, whose built-in ADC and DAC support variable sampling rates from 8kHz to 96kHz, and supports both I2C and SPI interfaces to configure internal registers while also allowing audio data transmission via the I2S interface.

[0098] RF Module: This module includes a DAC unit, an ADC unit, a first conditioning circuit, a second conditioning circuit, and a power amplifier, responsible for the mutual conversion between digital baseband signals and analog RF signals. Upon receiving a baseband signal from the polarization decoder, it is loaded onto the analog RF signal via the DAC. After conditioning by the uplink transmit link's pre-stage driver circuit (i.e., the first conditioning circuit), it is finally transmitted by the coil antenna, completing signal transmission. Upon receiving a voice signal from the coil antenna, the voice signal is conditioned by the downlink receive link (i.e., the second conditioning circuit), sampled by the ADC to obtain a digital baseband signal, and input to the demodulator. More specifically, this embodiment selects MAX5875 as the DAC unit (16-bit resolution, maximum update rate 200 MSPS) and LTC2217 as the ADC unit (16-bit resolution, maximum sampling rate 105 MSPS); the bandpass filter in the first conditioning circuit ensures that the spectral energy of the modulated signal is strictly limited to the uplink frequency band; the bandpass filter in the second conditioning circuit is responsible for accurately extracting the effective signal in the downlink frequency band from the wideband signal sampled by the ADC unit and suppressing strong interference from the local transmission link.

[0099] To prevent the coaxial cable from twisting and tangling during pan-tilt rotation, which could lead to mechanical damage over time, this embodiment integrates the entire system into a compact unit that rotates synchronously with the pan-tilt, thus eliminating relative cable twisting. Specifically, the voice communication system in this embodiment is mounted on the pan-tilt; the PS-side Linux interactive module controls the pan-tilt to rotate the coil antenna at a uniform speed. The pan-tilt is driven by a TB6600 motor, with a rotation range of 0-360°.

[0100] In this embodiment, the PS-side Linux interaction module configures the audio compiler TLV320AIC23 through the I2C interface, configures the polar encoder, polar decoder, modulator and demodulator on the PL side through the AXI bus interface, and drives the pan-tilt unit to rotate through the GPIO interface (PWM output).

[0101] In this embodiment, the audio compiler TLV320AIC23 transmits data to the PL-side signal processing module via an I2S interface; the RF module transmits signals to the PL-side baseband signal processing module via a high-speed parallel interface. The coil antenna in the automatic alignment module transmits signals to the RF module via a low-loss coaxial cable.

[0102] In this embodiment, the Zynq chip, voice module, and DAC, ADC, first conditioning circuit, and second conditioning circuit in the RF module are mounted on the same PCB board, serving as the controller board (i.e., the motherboard). The power amplifier in the RF module is mounted on a separate PCB board, denoted as the power amplifier board. This achieves physical isolation of the power amplifier in the RF module, effectively preventing the high-intensity RF energy generated by the power amplifier from coupling to the extremely sensitive receiving front-end on the motherboard through conduction or radiation. This avoids the receiver being "blocked" or "blinded," thereby maximizing the protection and utilization of the system's receiving sensitivity and dynamic range.

[0103] In this embodiment, the radio frequency module has good scalability, enabling the system to support both full-duplex and half-duplex transceiver modes.

[0104] When the magnetic induction voice communication system is operating in full-duplex mode, the center frequencies of the filters in the first conditioning circuit and the second conditioning circuit are different, but the bandwidths are the same.

[0105] The aforementioned automatic alignment module also includes: a parasitic coil with the same number of turns as the coil antenna; the parasitic coil and the coil antenna have parallel cross-sections and are placed coaxially to form a magnetic antenna with a parallel parasitic structure;

[0106] Resistance of coil antenna and the axial spacing between the coil antenna and the parasitic coil. These are the resistance and spacing values ​​that maximize the bandwidth of the magnetic antenna, selected from their respective preset candidate sets through optimization algorithms.

[0107] The expression for the magnetic antenna bandwidth is:

[0108]

[0109] in, and For the equation The only two real roots; The operating angular frequency of the coil antenna The transmit power at the given time is expressed as:

[0110]

[0111] in, The impedance of the parasitic coil, ; The resistance of the parasitic coil; j The imaginary unit; The inductance of the parasitic coil; The capacitance of the parasitic coil; This represents the voltage amplitude of the coil antenna; The impedance of the coil antenna, ; The inductance of the coil antenna; The capacitance of the coil antenna; ; Permeability of free space; The number of turns of the coil antenna or parasitic coil; Where is the coil radius of the coil antenna; The radius of the parasitic coil is denoted as .

[0112] In this embodiment, the coil antennas are all wound with high-purity copper wire with a wire diameter of 1mm, a coil radius of 0.2m, and 30 turns. The parasitic coil is an independent passive closed loop.

[0113] Through the aforementioned concurrent processing mechanism, this invention effectively transforms the spectrum resources gained from antenna bandwidth expansion into stable and reliable two-way communication capabilities. The frequency domain isolation and synchronous operation of the transmit and receive links make it possible to achieve smooth, natural interaction, much like a ground telephone call, in complex underground environments, significantly improving the feasibility and reliability of emergency communications.

[0114] When the above magnetic induction voice communication system is operating in half-duplex mode, the center frequencies and bandwidths of the filters in the first and second conditioning circuits are the same.

[0115] The radio frequency module also includes a T / R switch for time-division control of the coil antenna connecting to the power amplifier (i.e., the transmit link) or the second conditioning circuit (i.e., the receive link); specifically, when the system performs a voice signal transmission task, the power amplifier and the coil antenna are connected, and the second conditioning circuit is disconnected from the coil antenna; when the system performs a voice signal reception task, the second conditioning circuit is connected to the coil antenna, and the power amplifier and the coil antenna are disconnected.

[0116] At this point, the system can utilize only a single coil and, under narrow bandwidth conditions, alternately transmit and receive signals by time-division switching via a T / R switch, thus realizing half-duplex voice communication under narrow bandwidth conditions with a single coil.

[0117] Through the hardware architecture described above combined with software control and management, the system provided in this embodiment can achieve full-duplex communication and has the flexibility to be reconfigured to half-duplex mode.

[0118] The following description uses full-duplex mode as an example. The workflow of the entire system is as follows:

[0119] S1. System Initialization. After both devices start up, the Linux operating system runs the main control application, manages the system's operation flow, and handles human-machine interaction. Users input configuration parameters for each module through the interactive interface, such as the pan-tilt speed and direction, the TLV320AIC23 operating mode, the polar code encoding rate, and the initial modulation scheme of the adaptive modem (default is BPSK). The system completes the initial configuration of each module based on the user input.

[0120] S2. Automatic Coil Alignment. After system initialization, before device A and device B, equipped with the aforementioned underground magnetic induction voice communication system, formally establish communication, automatic alignment of the transceiver coils is first performed. Device A continuously generates a frequency using a MAX5875. A single-frequency test signal with fixed power is transmitted by the parasitic structure coil after being driven by the uplink transmit link of device A. Device B outputs a PWM signal via GPIO to drive the gimbal to rotate the parasitic structure coil at a constant speed of 180°. During the rotation, the receive link of device B monitors the signal strength value and feeds it back to the PS terminal in real time, synchronously recording the rotation angle of the gimbal. After the rotation is completed, the PS terminal analyzes the recorded signal strength data sequence and performs peak retrieval to find the maximum RSSI value and its corresponding precise angle, driving the gimbal to position itself at this point, completing the alignment of the transceiver coils.

[0121] S3, Full-duplex voice communication. With the coil aligned, the system's full-duplex voice communication function begins. Its signal processing flow is as follows: Figure 2 As shown (using device A sending and device B receiving as an example):

[0122] Device A's microphone first captures the user's voice signal and converts it into an analog electrical signal (300Hz~3.4kHz), which is then fed into the TLV320AIC23 chip. The chip's internal ADC... The sampling rate is used to sample and convert the analog speech signal into a digital speech signal, which is then transmitted to the PL terminal in real time via an I2S interface. The PL terminal compresses and encodes the digital speech data before sending the compressed data to the polarization encoder. This embodiment uses a code length... Information bits Encoding is performed using sub-channel independent unequal distribution polar codes (i.e., coding code rate of 0.5). The encoded bit stream, preamble sequence, and CSI estimated by the receive link form a transmission frame. The transmission frame is sent to the modulator, mapped into corresponding I / Q symbols, digitally filtered, and then digitally up-converted to shift the symbol stream to the uplink transmit sub-band (center frequency). In this embodiment, the modulator dynamically selects the modulation scheme (initially BPSK) based on the CSI information fed back from the receiving link of device A. The PL terminal transmits the digital I / Q signal to the MAX5875 chip of the RF module via a high-speed parallel interface, converting it into an analog signal. The signal then passes through the first conditioning circuit (including an uplink bandpass filter and a center frequency...). ,bandwidth After being driven by the power amplifier, the signal is transmitted to the coil antenna via a coaxial cable, and the antenna converts the electrical signal into a magnetic field signal and radiates it into space.

[0123] The coil antenna of device B induces a change in the magnetic field, generating a weak electrical signal (center frequency). ) and locally transmitted leakage signals (center frequency) The mixed signal enters the radio frequency module and passes through the second conditioning circuit (including a receiving bandpass filter and a center frequency filter). ,bandwidth The system suppresses strong interference from the local transmit subband of device B and accurately extracts weak useful signals from the receive subband. These signals are sampled by the LTC2217 chip, converted into digital signals, and transmitted to the PL terminal via a high-speed parallel interface. After digital filtering, the signals undergo digital down-conversion and demodulation by the demodulator to output a transmission frame. Subsequently, the received CSI is transmitted to the PS terminal to switch the modulation mode of device B's transmit link; the preamble sequence is used by device B for channel estimation; the CSI indicates the demodulation mode of device B's receive link; the voice bitstream enters the polarization decoder for error correction decoding to recover the compressed voice data. This example uses the least squares (LS) method and the continuous deletion list (SCL) algorithm for channel estimation and decoding, respectively. After decompression, the voice data is sent to the TLV320AIC23 chip via the I2S interface. The chip's internal DAC converts it into an analog signal, which is then used by an internal amplifier to drive the speaker, thus completing the voice reception.

[0124] The following are the polar encoding methods of the polar encoder in the above workflow:

[0125] Figure 3 The polarization process of four independent but differently distributed (idd) sub-channels is shown, illustrating the complex channel polarization of sub-channels. and channel output It conforms to idd. In this embodiment, the polar code length is... Information bits .

[0126] The aforementioned polar encoder performs polarization encoding on digital speech signals in the following manner:

[0127] 1) Complex channel polarization: In order to obtain the information bit index set ,right The composite channel is obtained by merging the idd sub-channels. The synthesized channel is then split into a series of polarization sub-channels with sequential dependencies. .

[0128] Assume the input symbol set is The output symbol set is The channel has Distribution states, , It is a positive integer, and further assumes that each type of channel appears Next. The first Secondary use of channel Recorded as ,in, , .

[0129] Will Divided into By performing single-step complex channel polarization on each polarization pair and combining the results, a channel of length [length missing] can be synthesized. new channel Repeat this process for each channel use. Next, get Synthetic Channels Finally, use Get the length as Synthetic Channel .

[0130] The general formulas for channel combining and splitting in complex channel polarization are as follows:

[0131]

[0132]

[0133] in, It is a channel The transfer function, and They represent The input and output sequences of the next transmission. This represents the input of the polarization encoder.

[0134] 2) Bartholomew parameter calculation: Calculate the Bartholomew parameters for each complex polarization subchannel;

[0135] In polar codes, the Bach parameter is used to evaluate the reliability of subchannels. The Bach parameter has a clear physical meaning; it represents the upper bound of the error probability of the maximum likelihood decision in a single channel transmission and is commonly used as an important indicator of channel performance. Its calculation formula is:

[0136]

[0137] By utilizing the results of complex channel polarization, the problem of determining the information bit index set is transformed into the problem of calculating the Bach parameters. After complex channel polarization, the synthetic channel can be obtained. and polarization channel From the transfer function, the formula for calculating the Bartholomew's parameters of complex channel polarization can be derived as follows:

[0138]

[0139] Initial received signal The likelihood ratio (LR) and log-likelihood ratio (LLR) are expressed as follows:

[0140]

[0141] in, For the first i Subchannel coefficients; For noise variance; and The first i Likelihood ratio and log-likelihood ratio of a complex polarimetric subchannel.

[0142] Furthermore, the formula for calculating the Bartholomew's parameters for complex channel polarization can be further transformed into the following form:

[0143]

[0144] It should be noted that the above calculation of the Bartholomew's parameters involves multiple dependent symbol calculations, and its computational complexity increases exponentially with the code length. Therefore, directly calculating the accurate value of the Bartholomew's parameters for each polarization sub-channel becomes extremely difficult. The Monte Carlo algorithm (MC algorithm) can effectively solve this problem. The MC algorithm can be used to statistically analyze... The expectation is used to approximate the Bavarian parameters of the polarimetric subchannel.

[0145] This embodiment uses the Monte Carlo algorithm (MC algorithm) to approximate the Bartholomew's parameters of the polarization subchannel.

[0146] Code length Symbolic energy The channel coefficient vector is Signal-to-noise ratio is Initialize key parameters as input. In this iteration, the algorithm follows the formula Generate a received signal for each bit, and according to the formula Initialize the log-likelihood ratio for each bit. Then update the log-likelihood ratio for each bit, and calculate the Parshall parameters for each polarization sub-channel using the Parshall parameter calculation formula. Finally, for... The results of each iteration are averaged to output stable estimates of the Bach parameters. Thus providing information for polar codes The construction provides a reliable basis.

[0147] 3) Sort the Parvian parameters of each complex polarization subchannel, and use the complex polarization subchannel indices corresponding to the top NK Parvian parameters with larger values ​​as frozen bit indices, and use the complex polarization subchannel indices corresponding to the remaining Parvian parameters as information bit indices; K is the preset number of information bits.

[0148] right Sort in descending order, information set Determined by the following formula

[0149]

[0150] in, Indicates will Arranged in descending order Larger values.

[0151] 4) Set the transmit bits under the freeze bit index to 0 and the transmit bits under the information bit index to 1 in order to encode the digital speech signal and obtain the encoding vector. Multiply the encoding vector with the polar code generator matrix GN of length N to obtain the polar coding result of the digital speech signal.

[0152] The polar code construction of the underground magnetic induction channel is completed by the following formula:

[0153]

[0154]

[0155] in, , , yes of Kronecker product of order, This indicates a bit flipping operation. It is a generating matrix. They represent Information Collection Information subvectors and generating matrices composed of elements at corresponding indices The middle belongs to the information set The corresponding rows form a submatrix. Similarly, They represent The supplement to .

[0156] Using digital voice signals as Substituting into the above formula, we obtain the corresponding polarization coding result.

[0157] Experimental verification revealed that, , , At that time, the above-mentioned polar code construction method based on underground magnetic induction channel improved the signal-to-noise ratio by about 2dB compared with the traditional polar code construction method.

[0158] The following is the adaptive modulation and demodulation method based on underground magnetic induction channel state information in the above workflow, specifically as follows: Figure 4 As shown.

[0159] To address the time-varying characteristics of underground magnetic induction channels, using fixed modulation methods fails to fully utilize the channel, resulting in low channel spectral efficiency. By introducing an adaptive modulation and demodulation method based on Channel State Information (CSI), the method can track channel changes in real time. Under favorable channel conditions, higher-order modulation is used to improve data transmission rates; under poor channel conditions, lower-order modulation is used to ensure communication reliability. The specific implementation process is as follows:

[0160] This embodiment restricts the modulation set to BPSK, 4-QAM, 16-QAM, and 64-QAM, corresponding to the first to fourth modulation schemes, respectively. A mapping table is set in the main control unit; the mapping table includes: the signal-to-noise ratio of the system channel. Correspondence with modulation method; when When, the corresponding modulation method is BPSK modulation; when When, the corresponding modulation method is 4-QAM modulation; when When, the corresponding modulation method is 16-QAM modulation; when At this time, the corresponding modulation method is 64-QAM modulation; All of these are switching thresholds.

[0161] The above It is calculated in the following way:

[0162] The system is constructed such that its average bit error rate is less than or equal to a preset average bit error rate. (In this embodiment, the value is taken as) The objective function is defined by the constraints , which aims to maximize the spectral efficiency of the system.

[0163] The objective function mentioned above is:

[0164]

[0165] The constraints are:

[0166]

[0167] in, For the first i The rate of each modulation scheme; It is denoted as infinity; For system channel signal-to-noise ratio

[0168] The probability distribution function of the ratio; The bit error rate function of the system;

[0169] The Lagrange algorithm is used to solve the above objective function, and the above results are obtained. The constructed Lagrange function is as follows:

[0170]

[0171] in, For Lagrange operators.

[0172] Specifically, taking the AWGN channel as an example, the received signal-to-noise ratio (SNR) is:

[0173]

[0174] in, This refers to the transmission power. For the time-varying channel gain of the underground channel; Let V be the variance of the AWGN channel.

[0175] Under BPSK modulation, the system's bit error rate function is:

[0176]

[0177] In M-QAM modulation, the system's bit error rate function is:

[0178]

[0179] M is the modulation order, which in this embodiment is 4, 16 or 64.

[0180] Switching threshold This is obtained by solving the following equation.

[0181]

[0182] This implementation uses a numerical search method to solve the problem, for example, it obtains: , , , That is, when the real-time signal-to-noise ratio of the system channel... In Within the range, the modulation method is BPSK; in Within the range, the modulation method is 4-QAM; in Within the range, the modulation scheme is 16-QAM; in Within the range, the modulation method is 64-QAM.

[0183] This embodiment employs a pilot-assisted channel estimation method to estimate the CSI of the underground magnetic induction channel in real time at the receiver. An L=16-bit preamble sequence is inserted into each transmission frame, and channel estimation is performed using least-squares estimation, as shown in the following formula:

[0184]

[0185] in, This is the pilot vector for transmission. To receive the pilot vector, This indicates the conjugate transpose.

[0186] Furthermore, through estimation With noise variance The instantaneous SNR is calculated as follows:

[0187]

[0188] The above estimates With switching threshold By comparing the modulation methods, the receiver determines the modulation scheme and maps it into short bits to instruct the transmitter to select the appropriate modulation scheme and the receiver to select the subsequent demodulation scheme. Furthermore, the mapped bits are encapsulated into a transmission frame and fed back via the downlink. Upon receiving this feedback, the transmitter switches the modulation scheme, achieving dynamic adaptation.

[0189] The following is a detailed explanation of the continuous bandwidth extension and full-duplex communication method for magnetic antennas based on parallel parasitic structures in the above workflow:

[0190] Specifically, the circuit parameters of the coil antenna are designed as follows: resistance = 100 Ω, inductance = 634 μH, capacitance = 150 pF, its natural resonant frequency Designed at 450 kHz. This is achieved by placing the parasitic coil at a distance (axial spacing) from the center of the coil antenna. At a position of 0.37 m, frequency splitting is artificially induced by utilizing the mutual inductance coupling effect between the two. By performing system analysis on the equivalent circuit model of the magnetic antenna with parasitic structure, and applying a sequential quadratic programming algorithm to jointly optimize the parasitic distance and impedance parameters, the optimal parameter combination for system performance can be obtained.

[0191] Specifically, the parasitic coil and the coil antenna cross-section are parallel to each other and placed coaxially, together forming a magnetic antenna with a parallel parasitic structure. The equivalent resistance of the magnetic antenna... and equivalent resistance They are respectively:

[0192]

[0193]

[0194] Where M is the mutual inductance between the coil antenna and the parasitic coil; ; The number of turns of the coil antenna or parasitic coil; Permeability of free space; Where is the coil radius of the coil antenna; The radius of the parasitic coil; This refers to the axial distance between the coil antenna and the parasitic coil; This is the operating angular frequency of the coil antenna.

[0195] Resistance of coil antenna and the axial spacing between the coil antenna and the parasitic coil. These are the resistance and spacing values ​​that maximize the magnetic antenna bandwidth, selected from their respective preset candidate sets using a sequential quadratic programming algorithm; where the expression for the magnetic antenna bandwidth is:

[0196]

[0197] and For the equation The two real roots of ; The operating angular frequency of the coil antenna The transmit power at the given time is expressed as:

[0198]

[0199] The impedance of the parasitic coil, ; The resistance of the parasitic coil; j The imaginary unit; The inductance of the parasitic coil; The capacitance of the parasitic coil; This represents the voltage amplitude of the coil antenna; The impedance of the coil antenna, ; The inductance is for the coil antenna.

[0200] Under optimal configuration, the antenna's frequency response splits from a single sharp resonant peak into two resonant peaks with equal amplitude and flat transitions, the effect of which is as follows: Figure 5 As shown, this forms a continuous, flat wideband extending from 437.5 kHz to 463.5 kHz, with a total extended bandwidth of 26 kHz, providing the necessary physical spectrum resources for full-duplex communication.

[0201] As a result, the half-power bandwidth of the magnetic antenna has been significantly improved, providing sufficient spectrum resources for the subsequent division of continuous and flat independent transmit and receive subbands in the frequency domain, fundamentally overcoming the limitation of the narrow-band characteristics of traditional magnetic induction antennas on the realization of full-duplex communication.

[0202] Frequency domain partitioning and duplexer construction of extended bandwidth. The continuous total bandwidth obtained above is divided into two independent sub-bands, one high and one low. One sub-band is dedicated to the uplink transmission channel, and the other sub-band is dedicated to the downlink reception channel. A necessary guard interval is reserved between the two sub-bands to suppress adjacent-channel interference. The duplexer is implemented by a transmit bandpass filter in the first conditioning circuit and a receive bandpass filter in the second conditioning circuit. The transmit bandpass filter ensures that the spectral energy of the modulated signal is strictly limited to the uplink frequency band, while the receive bandpass filter is responsible for accurately extracting the effective signal in the downlink frequency band from the received mixed signal and suppressing strong interference from the local transmission link. This implementation method has the advantages of high precision, high stability, and reconfigurability. Specifically, after obtaining the above-mentioned 26 kHz continuous bandwidth, it is scientifically partitioned in the frequency domain to construct a full-duplex channel. In this embodiment, the continuous bandwidth is based on the original resonant frequency. The frequency bands are divided into two sections. The uplink transmit band is 438 kHz – 448 kHz, with a bandwidth of 10 kHz; the downlink receive band is 452 kHz – 463 kHz, also with a bandwidth of 10 kHz. A guard interval is provided between the two sub-bands at 450 kHz to effectively suppress adjacent-channel interference. The frequency domain duplexer is implemented using parallel analog bandpass filters, where the center frequency of the transmit passband filter is designed at [missing information]. Covering the entire uplink transmit frequency band, the aim is to ensure that the spectral energy of the modulated transmit signal is strictly limited within the uplink frequency band, with out-of-band rejection better than 40 dB; the center frequency of the receive bandpass filter is designed at... It covers the entire downlink receiving frequency band. Its core function is to accurately extract the effective signal in the downlink frequency band from the mixed signal received by the antenna, and to strongly suppress the strong interference in the transmission frequency band, thereby solving the full-duplex self-interference problem.

[0203] Synchronous establishment and operation of a full-duplex communication link. Based on automatic coil alignment and frequency band allocation, the system establishes a complete full-duplex communication link through coordinated invocation of polar code encoding, adaptive modulation and demodulation, and frequency domain duplex mechanisms. After automatic coil alignment ensures optimal coupling, both communicating devices synchronously activate the transmit and receive links. In the transmit link, the voice signal sequentially undergoes polar code encoding specifically designed for underground channels and adaptive modulation based on real-time channel conditions, resulting in a symbol stream constrained for transmission within the uplink subband. The receive link synchronously extracts weak useful signals from the downlink subband, recovers the voice signal after adaptive demodulation and polar code decoding. Specifically, based on the aforementioned physical layer and data link layer preparations, the system successfully constructs a complete full-duplex communication link. Its core lies in the fact that both communicating devices employ the aforementioned magnetic antenna with extended bandwidth and utilize the allocated independent sub-frequency bands to synchronously transmit and receive signals within the same time period, thereby achieving efficient and natural two-way real-time voice interaction. A full-duplex communication link is established between device A and device B, allowing both to simultaneously transmit and receive voice signals, ensuring the real-time nature and continuity of the call.

[0204] It should be noted that the specific chip models mentioned in this embodiment are merely examples, intended to clearly illustrate the technical details. Any other chip model with the same or similar functions can be used as an alternative; for example, the XCZU7EV chip in the Zynq UltraScale+ MPSoC series can also be used as the core controller. The above-described workflow and signal processing principles also apply to half-duplex mode, and will not be elaborated upon here.

[0205] In summary, this invention provides an underground magnetic induction voice communication system based on the Zynq architecture, achieving a leap from "0" to "1" in underground magnetic induction voice communication systems and constructing a feature-rich integrated system. Leveraging the advantages of the Zynq chip's heterogeneous platform, complex system control, protocol stack management, and high-performance signal processing tasks are efficiently executed at the PS and PL ends, respectively. Furthermore, a direct sampling RF architecture is adopted, eliminating the need for multiple components such as local oscillators, mixers, and intermediate frequency filters required in superheterodyne architectures. This design not only effectively improves data processing efficiency and ensures real-time voice communication but also significantly reduces system complexity, cost, and size, enabling a highly integrated system to meet the requirements of portable devices and filling the gap in underground magnetic induction voice communication systems.

[0206] Based on this, by integrating an automatic coil alignment mechanism based on received signal strength, a polar code construction method for independent and unequal distribution of sub-channels in underground magnetic induction channels, and an adaptive modulation and demodulation strategy based on underground magnetic induction channel state information, the synergistic effect of multiple anti-interference technologies and adaptive strategies, and a highly integrated design, reliable voice communication in underground environments can be effectively achieved. This solves the portability problem of underground magnetic induction communication equipment, provides communication capabilities for actual underground emergency rescue, and is suitable for underground emergency rescue scenarios such as earthquakes, mine collapses, and tunnel collapses, meeting the requirements for highly reliable voice communication.

[0207] Those skilled in the art will readily understand that the above description is merely a preferred embodiment of the present invention and is not intended to limit the present invention. Any modifications, equivalent substitutions, and improvements made within the spirit and principles of the present invention should be included within the scope of protection of the present invention.

Claims

1. An underground magnetic induction voice communication system based on the Zynq architecture, characterized in that, include: The system comprises a Zynq chip, a voice module, an RF module, and an automatic alignment module. The Zynq chip integrates a main control unit at its PS terminal and a polar encoder, polar decoder, modulator, and demodulator at its PL terminal. The voice module includes a voice acquisition unit, a voice output unit, and an audio compiler. The automatic alignment module includes a coil antenna and a signal strength detector. The RF module includes a DAC unit, an ADC unit, a first conditioning circuit, a second conditioning circuit, and a power amplifier. Before the system performs voice signal transmission or reception tasks, the main control unit controls the coil antenna to rotate at a constant speed until it covers all preset postures, and records the posture of the coil antenna in real time. At the same time, it acquires the signal strength received by the coil antenna as detected by the signal strength detector; and fixes the posture of the coil antenna at the coil antenna posture corresponding to the maximum signal strength. When the system performs the voice signal transmission task, the audio compiler converts the voice signal collected by the voice acquisition unit into a digital voice signal; the digital voice signal is then polarized and modulated by a polar encoder and a modulator to obtain a baseband signal; the baseband signal is then converted from digital to analog by a DAC unit, and then conditioned and amplified by a first conditioning circuit and a power amplifier; the amplified signal is then transmitted through a coil antenna. When the system performs the voice signal reception task, the voice signal received by the coil antenna is conditioned and converted from analog to digital by the second conditioning circuit and the ADC unit in sequence, and then converted into a baseband signal. The baseband signal is demodulated and polarized decoded by the demodulator and polarization decoder in sequence to recover the digital voice signal. The digital voice signal is converted into an analog voice signal by the audio compiler and then output by the voice output unit. The polarization encoder performs polarization encoding on the digital speech signal in the following manner: Will Channel merging is performed on 1 independent but differently distributed sub-channels to obtain a channel of length 1. The vector channel; where the transfer function of the vector channel is... ; and They represent The input and output sequences of this transmission; It is the first i The transfer function of each sub-channel; The code length is the polar code. Channel decomposition is performed on the vector channel to obtain There are 3 complex polarization sub-channels; among which, the 1st... i The transfer function of the complex polarization sub-channel is: in, This represents the input bit sequence of the polarization encoder; for The first in i 1 bit; for The subsequence is middle Previous i A sequence consisting of -1 bits; for The subsequence of represents middle After A sequence consisting of bits; For length is The set of all binary sequences; Calculate the Parsons parameters for each complex polarization subchannel; The Bartholomew's parameters of each complex polarization subchannel are sorted, and the indices of the complex polarization subchannels corresponding to the top NK Bartholomew's parameters with the largest values ​​are used as frozen bit indices, while the indices of the complex polarization subchannels corresponding to the remaining Bartholomew's parameters are used as information bit indices; K is the preset number of information bits. The transmit bits under the freeze bit index are set to 0, and the transmit bits under the information bit index are set to 1 to encode the digital speech signal, resulting in a coded vector. This coded vector is then combined with the generator matrix G of a polar code of length N. N Multiplying these results yields the polarization coding of the digital speech signal.

2. The underground magnetic induction voice communication system according to claim 1, characterized in that, The magnetic induction voice communication system operates in full-duplex mode; the center frequencies of the filters in the first conditioning circuit and the second conditioning circuit are different, but the bandwidths are the same; The automatic alignment module further includes: a parasitic coil with the same number of turns as the coil antenna; the parasitic coil and the coil antenna have parallel cross-sections and are placed coaxially to form a magnetic antenna with a parallel parasitic structure; Resistance of coil antenna and the axial spacing between the coil antenna and the parasitic coil. These are the resistance and spacing values ​​that maximize the bandwidth of the magnetic antenna, selected from their respective preset candidate sets through optimization algorithms. The expression for the magnetic antenna bandwidth is: and For the equation The two real roots of ; The operating angular frequency of the coil antenna The transmit power at the given time is expressed as: The impedance of the parasitic coil, ; The resistance of the parasitic coil; j The imaginary unit; The inductance of the parasitic coil; The capacitance of the parasitic coil; This represents the voltage amplitude of the coil antenna; The impedance of the coil antenna, ; The inductance of the coil antenna; The capacitance of the coil antenna; ; The vacuum permeability; The number of turns of the coil antenna or parasitic coil; Where is the coil radius of the coil antenna; The radius of the parasitic coil is denoted as .

3. The underground magnetic induction voice communication system according to claim 1, characterized in that, The magnetic induction voice communication system operates in half-duplex mode; the center frequencies and bandwidths of the filters in the first conditioning circuit and the second conditioning circuit are the same. The radio frequency module further includes a T / R switch, used to connect the power amplifier and the coil antenna when the system performs a voice signal transmission task, at which time the second conditioning circuit is disconnected from the coil antenna; and to connect the second conditioning circuit and the coil antenna when the system performs a voice signal reception task, at which time the power amplifier and the coil antenna are disconnected.

4. The underground magnetic induction voice communication system according to claim 1, characterized in that, The main control unit is equipped with a mapping table; the mapping table includes: the signal-to-noise ratio of the system channel. The correspondence between modulation methods; modulation methods include: the first to the fourth modulation methods; the first modulation method is BPSK modulation; the second modulation method is 4-QAM modulation; the third modulation method is 16-QAM modulation; the fourth modulation method is 64-QAM modulation; when When, the corresponding modulation method is BPSK modulation; when When, the corresponding modulation method is 4-QAM modulation; when When, the corresponding modulation method is 16-QAM modulation; when At this time, the corresponding modulation method is 64-QAM modulation; All are switching thresholds; The It is calculated in the following way: The system is constructed such that its average bit error rate is less than or equal to a preset average bit error rate. The objective function is defined by constraints to maximize the spectral efficiency of the system. The objective function is: The constraints are: in, For the first i The rate of each modulation scheme; It is denoted as infinity; Let be the probability distribution function of the system channel signal-to-noise ratio; The bit error rate function of the system; The objective function is solved using the Lagrange algorithm to obtain the... ; The main control unit is also used to perform channel estimation on the baseband signal input to the demodulator when the system is performing a voice signal receiving task, and then calculate the signal-to-noise ratio of the current system channel; determine the modulation method corresponding to the signal-to-noise ratio of the current system channel based on the mapping table, and make the demodulator adopt the corresponding demodulation method, and at the same time feed back to the transmitting end so that the modulator in the transmitting end adopts the modulation method.

5. The underground magnetic induction voice communication system according to any one of claims 1-4, characterized in that, The main control unit is connected to the audio compiler via an I2C interface; the main control unit is connected to the polar encoder, polar decoder, modulator and demodulator via an AXI bus interface; the audio compiler is connected to the polar encoder and polar decoder via an I2S interface; the coil antenna is connected to the RF module via a coaxial cable.

6. The underground magnetic induction voice communication system according to any one of claims 1-4, characterized in that, The magnetic induction voice communication system is mounted on the gimbal; the main control unit is used to control the gimbal to rotate the coil antenna at a constant speed before the system performs voice signal transmission or reception tasks.

7. The underground magnetic induction voice communication system according to any one of claims 1-4, characterized in that, The main control unit uses the Linux operating system.

8. The underground magnetic induction voice communication system according to any one of claims 1-4, characterized in that, The voice acquisition unit is a microphone; the voice output unit is a speaker.

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