NAVTEX demodulation system

Abstract

The application discloses a NAVTEX demodulation system. The system comprises a receiver interface for receiving a PCM audio stream sent by a medium / high frequency receiver and a control instruction issued by an upper system, a digital audio processor for performing standardized processing on two checked PCM audio streams, a NAVTEX demodulation module for sequentially and respectively performing dynamic frequency offset estimation compensation, soft detection matching filtering and double-channel time diversity soft decoding on two complex baseband signals to obtain a NAVTEX standard character string row data stream with validity marks, an information encoding module for encoding and packaging the NAVTEX standard character string row data stream with validity marks into a JSON format message, and an application interface module for sending out the JSON format message. The application introduces a soft detection mechanism to replace a traditional hard decision mode, improves decision accuracy through soft information merging, and enhances demodulation capability in a low signal-to-noise ratio environment.

Description

Technical Field

[0001] This invention relates to the field of NAVTEX demodulation system technology, and more specifically to a NAVTEX demodulation system. Background Technology

[0002] Maritime safety communication is an indispensable means of ensuring the safe navigation of ships at sea. Maritime safety communication facilities effectively support daily ship traffic safety management and maritime law enforcement, broadcast maritime safety information, and are extremely important for preventing maritime accidents. my country's maritime safety communication mainly relies on the GMDSS communication system.

[0003] GMDSS is an abbreviation for the Global Maritime Distress and Safety System, which mainly consists of three parts: a satellite communication system, a terrestrial radio communication system, and a maritime safety information broadcasting system. The establishment and development of the Global Maritime Distress and Safety System (GMDSS) is an internationally mandatory requirement. The shore-based GMDSS facilities built by my country and other countries around the world provide reliable safety guarantees for ships equipped with GMDSS facilities.

[0004] A system that broadcasts and automatically receives navigational warnings, weather warnings, weather forecasts, and emergency information (collectively referred to as maritime safety information) to ships via direct-printed telegrams. It is an integral part of the Global Maritime Distress and Safety System and the Worldwide Navigational Warning Service (WWNWS) as defined in International Maritime Organization (IMO) Resolution A.419(XI). Ships use NAVTEX receiving equipment to receive navigational warnings, weather warnings, weather forecasts, and emergency information, while coastal radio stations use the NAVTEX broadcasting system to broadcast NAVTEX messages, with the broadcast format conforming to GB / T 18766-2009 "Technical Requirements for NAVTEX Systems" and the Ministry of Transport's JT / T 1519-2024 "NAVTEX System Broadcast Format Specification".

[0005] Chinese invention patent application CN201110300669.9 discloses a NAVTEX demodulator and demodulation method based on DSTFT. It proposes a NAVTEX demodulator based on Discrete Short-Time Fourier Transform (DSTFT) and uses software-defined radio technology to achieve digital demodulation. The demodulator consists of an A / D sampling module, a signal demodulation module, and an information decoding module. It reduces the performance requirements of the front-end A / D converter through down-sampling technology, and can still clearly distinguish the NAVTEX signal spectrum even at low signal-to-noise ratios. It features software upgradeability and high flexibility. NAVTEX demodulation is implemented using dedicated modem devices, relying on dedicated hardware. Existing technologies rely on dedicated hardware and cannot be implemented on general-purpose commercial x86 servers. Their receiver interfaces are analog audio interfaces, which cannot be networked. The demodulation module uses traditional dedicated signal processing circuit technology, relying on microcontrollers, digital signal processing chips, etc. The interfaces are diverse, the functions are limited, and the transmission relies on dedicated lines, which limits its application. Its disadvantages can be summarized as follows: 1. Strong hardware dependence: Although software radio technology is used, it still relies on specific DSP chips (such as Freescale's 56800E series DSP56853) to implement core functions, and lacks cross-platform capability; 2. Lack of networking capabilities: The system uses traditional hardware interface methods, which cannot be directly connected to IP networks, making it difficult to integrate with modern networked systems and not supporting remote management; 3. Closed architecture: As a closed hardware system, it is difficult to achieve efficient software-level integration with upper-layer applications such as GMDSS business systems, and data exchange relies on physical interfaces. 4. Difficult to upgrade and maintain: Upgrading system functions requires updating firmware or reprogramming, which must be done on-site by professional technicians, resulting in high maintenance costs; 5. Fixed resource allocation: It is impossible to dynamically allocate computing resources according to the actual load, resulting in low resource utilization. 6. Limitations in signal processing: Although DSTFT demodulation and spectrum synchronization algorithms are used, there are shortcomings in frequency offset compensation and soft detection. The demodulation performance needs to be improved in complex channel environments. 7. Insufficient interface standardization: Although the system has a display and control unit to provide user interaction, it lacks standardized network interface protocols and has poor interoperability with IP network devices. 8. Deployment limitations: It must be deployed on a specific hardware platform and cannot run on general-purpose servers or cloud environments; deployment locations are limited. 9. High expansion costs: Adding channels or functions often requires updating the DSP / FPGA chip or adding additional hardware, resulting in high expansion costs and long cycles. Summary of the Invention

[0006] The purpose of this invention is to address the shortcomings of existing technologies by providing a NAVTEX demodulation system.

[0007] To achieve the above objectives, the present invention provides a NAVTEX demodulation system, comprising: The receiver interface is used to receive control commands and verify the control commands. After the verification is successful, the parameters are configured according to the control commands. Then, according to the control commands, the receiver receives PCM audio streams of various sampling frequencies sent by the intermediate frequency receiver or the high frequency receiver, and performs preprocessing and verification on the received PCM audio streams. A digital audio processor is used to standardize the verified PCM audio stream based on verified control instructions to obtain a complex baseband signal. The NAVTEX demodulation module is used to perform dynamic frequency offset estimation compensation, soft detection matched filtering and soft decoding on the complex baseband signal in sequence to obtain a NAVTEX standard string data stream with validity markers; The information encoding module is used to encode the NAVTEX standard string data stream with validity markers to obtain NAVTEX maritime messages, and to encapsulate the NAVTEX maritime messages into JSON format messages; The application interface module is used to send the JSON formatted message.

[0008] Furthermore, the receiver interface receives PCM audio streams sent by the medium / high frequency receiver by creating an RTP session listening port, and its preprocessing methods for the received PCM audio streams include: The audio interruptions caused by network transmission jitter are eliminated by combining adaptive jitter buffering with the WSOLA algorithm. The receiver interface verifies the PCM audio stream in the following way: Verify the RTP packet header and discard packets with abnormal sequence numbers or timestamp jumps exceeding the threshold.

[0009] Furthermore, the receiver interface receives JSON format control commands via a RESTful API, and verifies the validity of the control commands through digital signature verification.

[0010] Furthermore, the digital audio processor performs standardization processing in the following manner: An FIR multiphase filter is used to resample PCM audio streams with multiple sampling frequencies to a preset frequency. A local complex exponential sequence is generated based on the intermediate frequency offset in the control command. The spectrum shift of the PCM audio stream is completed through the local complex exponential sequence to obtain a preliminary complex baseband signal. The FIR low-pass filter is dynamically generated, and the passband bandwidth of the FIR low-pass filter is dynamically adjusted according to the real-time signal-to-noise ratio to perform noise reduction processing on the initial complex baseband signal. It is equipped with a fast and slow dual-loop AGC control strategy. The slow loop is used to stabilize the average power of the noise-reduced complex baseband signal, while the fast loop is used to suppress shortwave channel burst pulse interference in the noise-reduced complex baseband signal in order to obtain the complex baseband signal.

[0011] Furthermore, the NAVTEX demodulation module performs dynamic frequency offset estimation compensation in the following way: Perform a 512-point discrete short-time Fourier transform on the complex baseband signal, traverse the spectrum to search for the signal energy peak, and obtain a rough frequency offset. ; Bit synchronization is performed on the complex baseband signal. During this process, the phase sequence of the synchronization header is extracted, and the phase slope is obtained by fitting the phase sequence of the synchronization header. ; Calculate the residual frequency offset for: ; in, For preset frequency, This represents the total number of sampling points in the synchronization head. According to the rough frequency offset and residual frequency offset Generate compensation factor ,in, It is a natural constant. The imaginary unit, For discrete sampling point numbers, Preset frequency The reciprocal of, based on the compensation factor The complex baseband signal is phase-rotated in real time to cancel out all frequency shifts in one go.

[0012] Furthermore, the phase slope The least squares method was used for fitting, as detailed below: ; in, This represents the discrete sequence number of each sampling point within the synchronization header. This is the average of the discrete sequence numbers of all sampling points in the synchronization head. For the first The phase of the complex baseband signal corresponding to each sampling point This is the average phase of the complex baseband signal at all sampling points of the synchronization head.

[0013] Furthermore, the coarse frequency offset is recalculated after processing each frame. and residual frequency offset And it is updated exponentially, as follows: ; Among them, the updated coarse frequency offset With residual frequency offset The sum of Coarse frequency offset calculated for the current frame With residual frequency offset The sum of The coarse frequency offset calculated for the previous frame With residual frequency offset sum.

[0014] Furthermore, the soft detection matched filtering method is as follows: A raised cosine roll-off filter is designed to perform matched filtering on the complex baseband signal after dynamic frequency offset estimation and compensation in order to obtain the sampling points of each symbol period. Calculate the log-likelihood ratio (LLR) for each symbol period sampling point, and quantize the LLR into an 8-bit signed integer with a value range of [-127, 127]. Encapsulate an 8-bit signed integer into a binary sequence with soft information.

[0015] Furthermore, the PCM audio stream consists of two signals, which are obtained through soft-detection matched filtering to form two binary sequences with soft information. The soft decoding method is as follows: The two binary sequences with soft information are respectively subjected to sliding weighted correlation with the local pre-stored synchronization word template. The weighting coefficient is the absolute value of the current bit log likelihood ratio (LLR). When the peak value of the sliding weighted correlation output is greater than the dynamic synchronization threshold, it is determined that a synchronization header has been detected and the frame start position is locked. Demodulate the binary sequences with soft information after the two synchronization headers respectively, perform character conversion every 7 bits, and simultaneously determine the constant ratio code according to the 7-bit constant ratio code encoding rule. If the character is determined to be a constant ratio code, set the flag of the character to 1; otherwise, set it to 0. The demodulation results from both channels can be categorized into one of the following four types: 1) The character flag for channel 1 and the character flag for channel 2 are both 1; 2) The character flag for channel 1 is 0, and the character flag for channel 2 is 1; 3) The character flag for channel 1 is 1, and the character flag for channel 2 is 0; 4) The character flag for both channel 1 and channel 2 is 0; If the result is 1, select either the character from channel 1 or the character from channel 2 as the current character for output; If the result is 2 or 3, select the character of the channel with character flag set to 1 as the current character for output; If the result is 4, the log-likelihood ratios (LLRs) corresponding to the current characters in Channel 1 and Channel 2 are added bit by bit and merged, and then demodulated and determined again. If the result is determined to be a constant ratio code, the character obtained by merging and demodulation is used as the output of the current character; otherwise, an error character identifier is output.

[0016] Furthermore, the dynamic synchronization threshold is calculated as follows: ; in, The calculated dynamic synchronization threshold, The mean of the output sequence for the sliding correlation operation. is the standard deviation of the sliding correlation output sequence.

[0017] Beneficial effects: 1. Pure software implementation architecture: This invention is a pure software network service system that does not rely on dedicated hardware and can run on general-purpose X86 / ARM servers, realizing the software-defined NAVTEX demodulation function; 2. Standard network interface: Provides a standard IP network interface, supports RTP protocol for transmitting audio data, JSON protocol for transmitting control information, can seamlessly connect to modern IP networks, and supports remote invocation; 3. Service-oriented design: Adopting the microservice architecture concept, the demodulation function is encapsulated as a network service, which can be called by GMDSS business systems through standard APIs, thereby improving system integration. 4. Soft detection technology: Introducing a soft detection mechanism to replace the traditional hard decision method, improving decision accuracy through soft information merging, and enhancing demodulation capability in low signal-to-noise ratio environments; 5. Dynamic frequency offset compensation: A variable frequency offset estimation and compensation technique is proposed to accurately estimate the frequency offset of the signal that has completed bit synchronization and dynamically compensate the received signal to improve demodulation performance. 6. Deployment flexibility: It can be deployed in environments such as virtual machines and containers, without being limited by physical location, supports cloud deployment, and adapts to dynamic changes in system scale; 7. Remote centralized management and control: Supports remote monitoring, fault diagnosis and software upgrades at any node in the network, reducing operation and maintenance costs and response time; 8. Lossless data transmission: Business data is transmitted entirely via IP, avoiding the lossy process of multiple A / D and D / A conversions in traditional systems, and ensuring high-fidelity audio transmission; 9. Dynamic resource allocation: Computing resources can be dynamically allocated based on real-time load, improving server resource utilization and reducing hardware investment costs; 10. Domestic substitution: Realize the domestic and networked application of NAVTEX demodulation technology, reduce dependence on imported dedicated hardware equipment, and meet the self-controllability requirements of GMDSS system; 11. Intelligent business expansion: The scale and functions of the business can be flexibly adjusted through software configuration as needed, without the need to replace hardware, to adapt to the new needs of maritime safety information services. Attached Figure Description

[0018] Figure 1 This is a schematic diagram of the NAVTEX demodulation system according to an embodiment of the present invention. Detailed Implementation

[0019] The present invention will be further illustrated below with reference to the accompanying drawings and specific embodiments. These embodiments are implemented based on the technical solutions of the present invention, and it should be understood that these embodiments are only used to illustrate the present invention and are not intended to limit the scope of the present invention.

[0020] like Figure 1 As shown, this embodiment of the invention provides a NAVTEX demodulation system, including a receiver interface 1, a digital audio processor 2, a NAVTEX demodulation module 3, an information encoding module 4, and an application interface module 5.

[0021] Receiver interface 1 is used to receive control commands and verify them. After successful verification, it configures parameters according to the control commands. After parameter configuration, receiver interface 1 receives PCM audio streams of various sampling frequencies sent by an intermediate frequency (MF) receiver or a high frequency (HF) receiver, preprocesses the received PCM audio streams, and then verifies the PCM audio streams.

[0022] Specifically, receiver interface 1 receives JSON-formatted control commands via a RESTful API (port 8080), and verifies the validity of these commands using digital signature verification (HMAC-SHA256). The aforementioned control commands specifically include: Startup command: Open the navigation receiving channel numbered NAV01, set the receiving center frequency to 518kHz and the receiving bandwidth to 300Hz, and start radio frequency signal acquisition; Parameter configuration command: Configure the receiver's automatic gain target level to -15dB, and set the intermediate frequency offset to 2.5ppm to correct hardware frequency deviation.

[0023] The above control commands are in JSON format, specifically as follows: {"cmd":"start","freq_khz":518,"bandwidth_hz":300,"channel_id":"NAV01"}; {"cmd":"config","agc_target_db":-15,"freq_comp_ppm":2.5}.

[0024] Receiver interface 1 receives PCM audio streams sent by medium / high frequency receivers by creating an RTP (Real-time Transport Protocol) session listening port (default 7504, configurable). The PCM audio stream is packaged into individual RTP data packets, each containing a segment of PCM audio sample data. RTP packets include sequence numbers and timestamps to mark the audio sequence and acquisition time. By listening to the RTP session listening port, a continuous stream of RTP data packets is received, reconstructing the continuous PCM audio stream. The PCM audio stream sampling frequency is 8kHz, 16kHz, or 48kHz, configurable as needed, with a sampling bit depth of 16-bit signed integers.

[0025] The receiver interface 1 performs preprocessing on the received PCM audio stream in the following ways: The audio interruptions caused by network transmission jitter are eliminated by using an adaptive jitter buffer (with a dynamic buffer depth adjustment range of 50ms-500ms) combined with the WSOLA algorithm.

[0026] The receiver interface 1 verifies the PCM audio stream in the following way: Verify the RTP header and discard packets with abnormal sequence numbers or timestamp jumps exceeding the threshold (10ms). Digital audio processor 2 is used to normalize the verified PCM audio stream based on the verified control commands to obtain a complex baseband signal. The normalization process performed by digital audio processor 2 is as follows: 1) Resampling: FIR multiphase filters are used to resample PCM audio streams of various sampling frequencies to a preset frequency (48kHz) to meet the bandwidth requirements of subsequent processing. The filter stopband attenuation is designed to be no less than 80dB.

[0027] 2) Digital Down-Conversion: A local complex exponential sequence is generated based on the intermediate frequency offset in the control command. This local complex exponential sequence is used to perform spectrum shifting of the PCM audio stream, resulting in a preliminary complex baseband signal. The aforementioned local complex exponential sequence can be represented as follows: ,in, It is a natural constant. The imaginary unit, To control the intermediate frequency offset in the command. For discrete sampling point numbers, It is the reciprocal of the preset frequency.

[0028] 3) Adaptive Low-Pass Filtering: Dynamically generates an FIR low-pass filter (bandwidth of 250Hz, transition band of 50Hz-100Hz) and dynamically adjusts the passband bandwidth of the FIR low-pass filter according to the real-time signal-to-noise ratio to perform noise reduction processing on the initial complex baseband signal. For example, at high signal-to-noise ratios, the passband is widened (250+100Hz) to fully preserve the useful sideband signals; at low signal-to-noise ratios, it is narrowed (250+50Hz) to filter out out-of-band interference noise.

[0029] 4) AGC Control: Equipped with a fast and slow dual-loop AGC control strategy. The slow loop stabilizes the average power of the noise-reduced complex baseband signal, while the fast loop suppresses shortwave channel burst interference in the noise-reduced complex baseband signal to obtain the desired complex baseband signal. The slow loop time constant is set to 1s to stabilize the average output power of the baseband signal at -15dBFS. The fast loop time constant is 10ms to avoid signal saturation distortion.

[0030] The NAVTEX demodulation module 3 is used to sequentially perform dynamic frequency offset estimation compensation, soft detection matched filtering, and soft decoding on the complex baseband signal to obtain a NAVTEX standard string data stream with validity markers. Specifically, the NAVTEX demodulation module 3 performs dynamic frequency offset estimation compensation in the following way: 1) Coarse estimation: Perform a 512-point discrete short-time Fourier transform (DSTFT, window length 10ms, overlap rate 25%) on the complex baseband signal, traverse the spectrum to search for the signal energy peak, and obtain a coarse frequency offset. .

[0031] 2) Precise estimation: Bit synchronization is performed on the complex baseband signal. During this process, the phase sequence of the synchronization header (the bit sequence corresponding to "ZCZC") is extracted, and the phase slope is obtained by fitting the phase sequence of the synchronization header. (The slope of the phase change over time). The above phase slope... The least squares method was used for fitting, as detailed below: ; in, This represents the discrete sequence number of each sampling point within the synchronization header. This is the average of the discrete sequence numbers of all sampling points in the synchronization head. For the first The phase of the complex baseband signal corresponding to each sampling point This is the average phase of the complex baseband signal at all sampling points of the synchronization head.

[0032] Calculate the residual frequency offset for: ; in, This represents the total number of sampling points in the synchronization head.

[0033] 3) Dynamic compensation: based on coarse frequency offset. and residual frequency offset Generate compensation factor Based on compensation factor The complex baseband signal undergoes real-time phase rotation to cancel out all frequency shifts in one go. The phase compensation calculation is performed entirely in the complex domain, effectively avoiding demodulation failure caused by periodic angular winding.

[0034] The coarse frequency offset can also be recalculated after processing each frame (240ms). and residual frequency offset And it is updated exponentially, as follows: ; Among them, the updated coarse frequency offset With residual frequency offset the sum of Coarse frequency offset calculated for the current frame With residual frequency offset the sum of The coarse frequency offset calculated from the previous frame With residual frequency offset sum.

[0035] The NAVTEX demodulation module 3 performs soft detection matched filtering in the following way: 1) Matched filtering: Design a raised cosine roll-off filter (roll-off coefficient α=0.3) to perform matched filtering on the complex baseband signal after dynamic frequency offset estimation compensation in order to obtain the sampling points of each symbol period; 2) Soft Information Generation: Calculate the log-likelihood ratio (LLR) for each symbol period sampling point, and quantize the LLR into an 8-bit signed integer with a value range of [-127, 127]. The calculation method for the LLR is as follows: ; in, To calculate bits The log-likelihood ratio, When the transmitted bit is 0, the received signal The probability, When the transmitted bit is 1, the received signal The probability, It is the natural logarithm function. This is the complex baseband received sample value of the current symbol after matched filtering. , These are the ideal complex baseband signal templates for transmitting bits 0 and 1, respectively. This is the variance estimate of the Gaussian noise in the channel. A lookup table method can be used to accelerate the calculation of the log-likelihood ratio (LLR).

[0036] An 8-bit signed integer is encapsulated into a binary sequence with soft information, replacing the traditional hard decision 0 / 1 bit output, providing redundant error correction basis for subsequent soft decoding.

[0037] NAVTEX messages can be transmitted alternately through channels 1 and 2 using time diversity. The original message content of the two channels is completely identical, thus allowing the reception of two PCM audio streams. Each PCM audio stream needs to be processed independently. After soft-detection matched filtering, two binary sequences with soft information are obtained. Based on this, the NAVTEX demodulation module 3 performs soft decoding as follows: 1) Soft correlation synchronization: Two binary sequences with soft information are respectively subjected to sliding weighted correlation with a locally pre-stored synchronization word template. The weighting coefficient is the absolute value of the current bit-log likelihood ratio (LLR). When the peak value of the sliding weighted correlation output is greater than the dynamic synchronization threshold, a synchronization header is detected, the frame start position is locked, and subsequent soft decoding is initiated. The preferred method for calculating the dynamic synchronization threshold is as follows: ; in, The calculated dynamic synchronization threshold, The mean of the output sequence for the sliding correlation operation. This represents the standard deviation of the sliding correlation output sequence. Using a dynamic threshold can adapt to changes in channel noise intensity. When noise is high, the threshold automatically increases to avoid false synchronization triggers; when noise is low, the threshold automatically decreases to prevent weak synchronization peaks from being missed. Compared to a fixed threshold, this approach offers stronger adaptability.

[0038] 2) Time-based soft decoding: Demodulate the binary sequences with soft information after the two synchronization headers respectively, and perform character conversion every 7 bits (i.e., map them to NAVTEX standard characters in 7-bit groups). At the same time, perform constant ratio code discrimination according to the 7-bit constant ratio code encoding rule. If it is determined to be a constant ratio code, set the flag of the character to 1, otherwise set it to 0.

[0039] The demodulation results from both channels can be categorized into one of the following four types: 1) The character flag for channel 1 and the character flag for channel 2 are both 1; 2) The character flag for channel 1 is 0, and the character flag for channel 2 is 1; 3) The character flag for channel 1 is 1, and the character flag for channel 2 is 0; 4) The character flag for both channel 1 and channel 2 is 0; If the result is 1, select either the character from channel 1 or the character from channel 2 as the current character for output; If the result is 2 or 3, select the character of the channel with character flag set to 1 as the current character for output; If the result is 4, the log-likelihood ratios (LLRs) corresponding to the current characters in Channel 1 and Channel 2 are added bit by bit and merged, and then demodulated and constant ratio code discrimination is performed again. If the discrimination is constant ratio code, that is, the character flag is 1, the character obtained by merging and demodulation is used as the output of the current character, thereby obtaining a NAVTEX standard string line data stream with validity marking; otherwise, that is, the character flag is 0, and an error character identifier is output for upper-layer fault alarm.

[0040] Information encoding module 4 is used to encode the NAVTEX standard string data stream with validity markers to obtain NAVTEX maritime messages, and encapsulates the NAVTEX maritime messages and maintenance metadata into a JSON format message. The encoding strictly follows the ITU-R M.540 maritime communication standard concatenation, including the start character ZCZC, transmitter B1, message number B2B3B4, and end character NNNN. The aforementioned maintenance metadata may include timestamps (UTC), signal-to-noise ratio (dB), frequency offset estimates (Hz), and demodulation confidence, etc. The specific format of the JSON format message is as follows: { "msg_id": "NAV20240520123456", "content": "ZCZC HA01 ... NNNN", "metadata": {"snr": 18.5, "freq_offset": 1.2, "confidence": 0.92}, "timestamp": "2024-05-20T12:34:56Z"}.

[0041] Application interface module 5 is used to send JSON-formatted messages. Specifically, it can push standardized NAVTEX messages that have been demodulated to the GMDSS maritime business system in real time via a WebSocket long connection. It can also periodically collect operational metrics such as CPU, memory, and demodulation success rate every 5 seconds, providing a monitoring data source based on the SNMPv3 protocol. It can also connect to Kubernetes and automatically scale Pod instances based on the number of RTP streams uploaded by the receiver, with a single instance supporting 8 concurrent demodulations. It also supports hot upgrades of container images, with business interruption time controlled within 200ms. In addition, it has performed deep instruction set optimization for domestic CPUs such as Loongson and Zhaoxin, and provides a dedicated deployment image for a domestic operating system.

[0042] Furthermore, this invention is preferably deployed on a standard x86 server (Intel Xeon E-2278GE and above) or a domestically produced ARM architecture server (Phytium / Kunpeng series), running a Linux operating system (CentOS 7.6+ / Ubuntu 20.04+), with ≥8GB of memory and ≥1Gbps network interface. The system is packaged in Docker containers and supports seamless integration with mainstream MF / HF network receivers (compliant with STANAG 5066 or custom RTP protocols). Implementation effect

[0043] Actual measurements were taken in the South China Sea (receiving a 518kHz signal from a Shanghai coastal radio station): Demodulation threshold: When SNR≥6dB, message accuracy >99.5% (3.2dB improvement over traditional hard decision); Frequency offset tolerance: Demodulation success rate >98% after dynamic compensation within ±150Hz range; Network latency: End-to-end latency from RTP reception to JSON output ≤ 1.2s (meets GMDSS real-time requirements); Resource consumption: Single-path demodulation CPU utilization <8% (Intel Xeon 3.2GHz), supports 32-path concurrent deployment on a single server.

[0044] The above description is merely a preferred embodiment of the present invention. It should be noted that for those skilled in the art, other parts not specifically described are existing technology or common knowledge. Several improvements and modifications can be made without departing from the principle of the present invention, and these improvements and modifications should also be considered within the scope of protection of the present invention.

Claims

1. A NAVTEX demodulation system, characterized in that, include: The receiver interface is used to receive control commands and verify the control commands. After the verification is successful, the parameters are configured according to the control commands. Then, according to the control commands, the receiver receives PCM audio streams of various sampling frequencies sent by the intermediate frequency receiver or the high frequency receiver, and performs preprocessing and verification on the received PCM audio streams. A digital audio processor is used to standardize the verified PCM audio stream based on verified control instructions to obtain a complex baseband signal. The NAVTEX demodulation module is used to perform dynamic frequency offset estimation compensation, soft detection matched filtering and soft decoding on the complex baseband signal in sequence to obtain a NAVTEX standard string data stream with validity markers; The information encoding module is used to encode the NAVTEX standard string data stream with validity markers to obtain NAVTEX maritime messages, and to encapsulate the NAVTEX maritime messages into JSON format messages; The application interface module is used to send the JSON formatted message.

2. The NAVTEX demodulation system according to claim 1, characterized in that, The receiver interface receives PCM audio streams sent by the medium / high frequency receiver by creating an RTP session listening port, and its preprocessing methods for the received PCM audio streams include: The audio interruptions caused by network transmission jitter are eliminated by combining adaptive jitter buffering with the WSOLA algorithm. The receiver interface verifies the PCM audio stream in the following way: Verify the RTP packet header and discard packets with abnormal sequence numbers or timestamp jumps exceeding the threshold.

3. The NAVTEX demodulation system according to claim 1, characterized in that, The receiver interface receives JSON-formatted control commands via a RESTful API, and verifies the validity of the control commands through digital signature verification.

4. The NAVTEX demodulation system according to claim 2, characterized in that, The digital audio processor performs standardized processing in the following manner: An FIR multiphase filter is used to resample PCM audio streams with multiple sampling frequencies to a preset frequency. A local complex exponential sequence is generated based on the intermediate frequency offset in the control command. The spectrum shift of the PCM audio stream is completed through the local complex exponential sequence to obtain a preliminary complex baseband signal. The FIR low-pass filter is dynamically generated, and the passband bandwidth of the FIR low-pass filter is dynamically adjusted according to the real-time signal-to-noise ratio to perform noise reduction processing on the initial complex baseband signal. It is equipped with a fast and slow dual-loop AGC control strategy. The slow loop is used to stabilize the average power of the noise-reduced complex baseband signal, while the fast loop is used to suppress shortwave channel burst pulse interference in the noise-reduced complex baseband signal in order to obtain the complex baseband signal.

5. A NAVTEX demodulation system according to claim 4, characterized in that, The NAVTEX demodulation module performs dynamic frequency offset estimation and compensation in the following way: Perform a 512-point discrete short-time Fourier transform on the complex baseband signal, traverse the spectrum to search for the signal energy peak, and obtain a rough frequency offset. ; Bit synchronization is performed on the complex baseband signal. During this process, the phase sequence of the synchronization header is extracted, and the phase slope is obtained by fitting the phase sequence of the synchronization header. ; Calculate the residual frequency offset for: ； in, For preset frequency, This represents the total number of sampling points in the synchronization head. According to the rough frequency offset and residual frequency offset Generate compensation factor ,in, It is a natural constant. The imaginary unit, For discrete sampling point numbers, Preset frequency The reciprocal of, based on the compensation factor The complex baseband signal is phase-rotated in real time to cancel out all frequency shifts in one go.

6. The NAVTEX demodulation system according to claim 5, characterized in that, The phase slope The least squares method was used for fitting, as detailed below: ； in, This represents the discrete sequence number of each sampling point within the synchronization header. This is the average of the discrete sequence numbers of all sampling points in the synchronization head. For the first The phase of the complex baseband signal corresponding to each sampling point This is the average phase of the complex baseband signal at all sampling points of the synchronization head.

7. A NAVTEX demodulation system according to claim 5, characterized in that, For each frame processed, the coarse frequency offset is recalculated once. and residual frequency offset And it is updated exponentially, as follows: ； Among them, the updated coarse frequency offset With residual frequency offset The sum of Coarse frequency offset calculated for the current frame With residual frequency offset The sum of The coarse frequency offset calculated for the previous frame With residual frequency offset sum.

8. A NAVTEX demodulation system according to claim 5, characterized in that, The soft detection matched filtering method is as follows: A raised cosine roll-off filter is designed to perform matched filtering on the complex baseband signal after dynamic frequency offset estimation and compensation in order to obtain the sampling points of each symbol period. Calculate the log-likelihood ratio (LLR) for each symbol period sampling point, and quantize the LLR into an 8-bit signed integer with a value range of [-127, 127]. Encapsulate an 8-bit signed integer into a binary sequence with soft information.

9. A NAVTEX demodulation system according to claim 8, characterized in that, The PCM audio stream consists of two signals, which are obtained by soft-detection matched filtering to produce two binary sequences with soft information. The soft decoding method is as follows: The two binary sequences with soft information are respectively subjected to sliding weighted correlation with the local pre-stored synchronization word template. The weighting coefficient is the absolute value of the current bit log likelihood ratio (LLR). When the peak value of the sliding weighted correlation output is greater than the dynamic synchronization threshold, it is determined that a synchronization header has been detected and the frame start position is locked. Demodulate the binary sequences with soft information after the two synchronization headers respectively, perform character conversion every 7 bits, and simultaneously determine the constant ratio code according to the 7-bit constant ratio code encoding rule. If the character is determined to be a constant ratio code, set the flag of the character to 1; otherwise, set it to 0. The demodulation results from both channels can be categorized into one of the following four types: 1) The character flag for channel 1 and the character flag for channel 2 are both 1; 2) The character flag for channel 1 is 0, and the character flag for channel 2 is 1; 3) The character flag for channel 1 is 1, and the character flag for channel 2 is 0; 4) The character flag for both channel 1 and channel 2 is 0; If the result is 1, select either the character from channel 1 or the character from channel 2 as the current character for output; If the result is 2 or 3, select the character of the channel with character flag set to 1 as the current character for output; If the result is 4, the log-likelihood ratios (LLRs) corresponding to the current characters in channel 1 and channel 2 are added bit by bit and merged, and then demodulated and constant ratio code is determined again. If it is determined to be a constant ratio code, the character obtained by merging and demodulation is used as the output of the current character. Otherwise, output an error character identifier.

10. A NAVTEX demodulation system according to claim 9, characterized in that, The dynamic synchronization threshold is calculated as follows: ； in, The calculated dynamic synchronization threshold, The mean of the output sequence for the sliding correlation operation. is the standard deviation of the sliding correlation output sequence.

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