A multi-terminal networked audio broadcast system

Abstract

This invention discloses a multi-terminal network audio broadcasting system, belonging to the field of audio broadcasting technology. It includes a central server and multiple distributed audio terminals. The central server comprises: a geographic information module for storing digital elevation models and the deployment locations of each audio terminal; an adaptive coding module for dynamically adjusting the redundancy coding rate of the audio stream based on link quality feedback reported by each audio terminal; and a routing calculation module for generating a multi-hop broadcast tree based on a terrain-compensated propagation loss model. Each audio terminal includes a positioning module for collecting its real-time geographic coordinates and reporting them to the central server. This invention achieves accurate path loss calculation, adaptive redundancy coding, dynamic power adjustment, intelligent broadcast tree construction, efficient packet loss recovery, and dynamic relay coverage, improving the coverage, stability, and energy efficiency of audio broadcasting.

Description

Technical Field

[0001] This invention relates to the field of audio broadcasting technology, specifically a multi-terminal network audio broadcasting system. Background Technology

[0002] Audio broadcasting systems are core infrastructure for public information dissemination, emergency notifications, and policy announcements. Traditional audio broadcasting systems, primarily analog FM broadcasting and wired constant voltage broadcasting, rely on a single transmitter or a small number of terminals to achieve signal coverage, making them only suitable for densely populated, flat urban environments. These systems suffer from limitations such as limited coverage, high cabling costs, weak signal resistance to terrain interference, and high maintenance difficulty, making them unsuitable for rural environments with vast areas, sparse populations, and complex terrain (mountains, hills, and ravines).

[0003] With the development of wireless communication and network technologies, multi-terminal network audio broadcasting systems are gradually replacing traditional broadcasting. These systems achieve wide-area wireless coverage by linking distributed audio terminals through a central server, becoming the mainstream solution for rural broadcasting. However, when existing multi-terminal network audio broadcasting systems are implemented in rural scenarios, the accuracy of path loss calculation is insufficient. They rely solely on the free space loss model, leading to large deviations in the estimation of link loss between terminals and unreasonable broadcast routing.

[0004] Therefore, those skilled in the art have provided a multi-terminal network audio broadcasting system to solve the problems mentioned in the background art. Summary of the Invention

[0005] This invention overcomes the shortcomings of existing multi-terminal network audio broadcasting systems in rural scenarios, and provides a multi-terminal network audio broadcasting system adapted to sparsely populated rural areas. It achieves accurate path loss calculation, adaptive redundancy coding, dynamic power adjustment, intelligent broadcast tree construction, efficient packet loss recovery, and dynamic relay blind spot filling, thereby improving the coverage, stability, and energy utilization of audio broadcasting.

[0006] To achieve the above objectives, the present invention provides the following technical solution: A multi-terminal network audio broadcasting system includes a central server and multiple distributed audio terminals; The central server includes: a geographic information module for storing digital elevation models and the deployment locations of each audio terminal; an adaptive coding module for dynamically adjusting the redundancy coding rate of the audio stream based on the link quality feedback reported by each audio terminal; and a routing calculation module for generating a multi-hop broadcast tree based on a terrain-compensated propagation loss model. The audio terminal includes: a positioning module for collecting the real-time geographical coordinates of the audio terminal and reporting them to the central server; a signal measurement module for periodically measuring the signal strength, signal-to-noise ratio, and packet loss rate received by the audio terminal from neighboring audio terminals or the central server, forming link quality data; an audio decoding and playback module for receiving audio data packets, decoding them, and converting them into analog audio signals for output to the speaker; a data forwarding module for receiving audio data packets from the parent node in the broadcast tree structure and forwarding them to the child node audio terminals in the downlink direction of the broadcast tree; and a status reporting module for packaging the location information, link quality data, remaining battery power, and current operating temperature of the audio terminal and sending them uplink to the central server according to a preset time slot. The central server periodically receives location and link quality data reported by each audio terminal, calculates the predicted path loss value between each pair of audio terminals, constructs the minimum spanning tree broadcast path based on the predicted loss value, and assigns a redundancy coefficient to each audio terminal.

[0007] As a further aspect of the present invention: the predicted path loss value Obtain it through the following methods: Acquire the terminal and terminal Based on the geographic coordinates and altitude, digital elevation model profile data is extracted from the line connecting the two terminals. All topographic peaks on this profile, along with their respective convex heights and Fresnel radii, are determined. The diffraction loss factor caused by each peak is calculated, and then the basic free-space loss is logarithmically summed with all diffraction loss factors. Specifically, The calculation expression is: ; in, For the terminal and The straight-line distance between them The radio frequency center frequency, At the speed of light, This represents the total number of terrain sharp peaks extracted from the digital elevation model profile. For the first The diffraction loss factor caused by the individual blade peaks, this factor By measuring the height of the blade peak protrusion First Fresnel radius at the edge peak The ratio is obtained by performing piecewise logarithmic operations, and when this ratio is less than a preset threshold, The value is zero.

[0008] As a further aspect of the present invention: the adaptive encoding module of the central server, based on the terminal... The terminal is calculated based on the depth of the broadcast tree and the average path loss of its neighboring terminals. Corresponding audio data packet redundancy Specifically, redundancy By using the terminal parent node to Predicted path loss value With terminal Average path loss of neighboring nodes within a preset range around itself Perform a weighted summation and divide by the reference loss value. After normalization, the normalized result is input into a sigmoid function and mapped to the (0, 1) interval to obtain the result. The calculation expression is as follows: ; in, For standard logistic S-type functions, and The weights are preset and their sum is 1. This is a reference loss value, which is equal to the path loss calculation value when the distance in free space is the preset reference distance.

[0009] As a further aspect of the present invention: the audio terminal further includes a dynamic power adjustment module, which calculates a transmission power adjustment coefficient based on the terminal's remaining battery power and the number of downstream terminals in the broadcast tree. (where subscript) (This refers to any terminal); specifically, for any terminal Its power adjustment coefficient By normalizing the remaining energy ratio of the terminal Number of downstream terminals Sum the reciprocals of the terms, add 1, take the natural logarithm, and multiply by the sensitivity factor. Finally, subtract the product from 1 and multiply it by the reference transmit power to obtain the result. The calculation expression is as follows: ; in, Rated transmission power, This is an adjustment sensitivity factor with a value between 0 and 1. To normalize the remaining energy and set its value to [0, 1], This represents the number of downstream terminals that the current terminal is responsible for forwarding in the broadcast tree, and when... When it is zero, Process as zero.

[0010] As a further aspect of the present invention: when constructing the minimum spanning tree broadcast path, the routing calculation module predicts the path loss value. As a benchmark for edge weights, and in conjunction with the terminal and terminal Each terminal's dynamic power adjustment coefficient corrects the edge weights, where the terminal... Power adjustment coefficient and terminal Power adjustment coefficient According to respectively The corrected edge weights are calculated in the following manner. The Prim algorithm is used to generate a broadcast tree with the central server as the root node.

[0011] As a further aspect of the present invention: after receiving an audio data packet, the audio terminal first uses a signal measurement module to check the sequence number of the data packet. If a missing packet is found, a packet retransmission request is initiated to the parent node or neighboring node through the status reporting module. If the terminal density in the area where the current terminal is located is lower than a preset threshold, a redundant decoding mode is activated, utilizing... The corresponding forward error correction coding directly recovers the missing data packets without initiating a retransmission request.

[0012] As a further aspect of the present invention: the central server also includes a broadcast content segmentation module. This module dynamically segments the audio stream into data blocks of unequal length based on the average terminal response time of each sub-region in the sparsely populated rural area. The data block length corresponding to terminals far from the central server is less than a preset value to reduce single-packet retransmission overhead. Specifically, for broadcast trees with a depth of... The terminal, its data block length By measuring the length of the base block With depth The result is obtained by multiplying the exponential decay function, and the base of the exponential decay function is determined by the ratio between the average moving speed of the terminal in the region and the audio playback buffer duration.

[0013] As a further aspect of the present invention: when the status reporting module of the audio terminal reports signal measurement data, it adopts a time-division multiple access mechanism based on geographic grouping: with the central server as the origin, the rural area is divided into several concentric rings. Terminals in the same ring use the same uplink time slot. Different rings are allocated different time slot lengths according to their radius. The larger the radius of the ring, the longer the time slot is allocated, in order to adapt to the large difference in propagation delay between terminals far from the center and the central server under the conditions of vast areas and sparse population.

[0014] As a further aspect of the present invention: the system also includes a mobile terminal relay mode. When a fixed audio terminal goes offline due to a malfunction or depletion of power, the central server detects that there are still uncovered terminals downstream of the terminal. It then temporarily authorizes mobile terminals installed on agricultural machinery or vehicles entering the broadcast area as dynamic relay nodes. The weight of the dynamic relay node is calculated by the matching degree between its movement trajectory and the direction of the missing link. The matching degree is obtained by weighted summation of the cosine of the vector angle between the current position of the mobile terminal and the position of the offline terminal, and the projection component of the movement speed in the direction of the connection.

[0015] As a further aspect of the present invention: the central server also includes an audio broadcast quality prediction module. This audio broadcast quality prediction module uses a Kalman filter algorithm to predict the link status of each terminal in the next broadcast cycle based on the packet loss rate, retransmission count, and signal attenuation trend of each terminal in historical broadcast data. It also adjusts the audio coding bitrate, forward error correction redundancy, and broadcast tree topology in advance based on the prediction results. The state transition matrix of the Kalman filter is dynamically updated according to the time-varying laws of temperature, humidity, and electromagnetic wave attenuation coefficient caused by vegetation growth in typical daily variations in sparsely populated rural areas.

[0016] Compared with the prior art, the beneficial effects of the present invention are: This invention accurately calculates path loss between terminals through terrain compensation, enabling the construction of reasonable broadcast routes tailored to complex rural terrain. It also adaptively adjusts the audio redundancy coding rate based on link quality and broadcast tree depth, balancing packet loss recovery with bandwidth utilization efficiency. Furthermore, it dynamically adjusts the transmission power based on the terminal's remaining battery power and the number of downstream terminals, effectively extending terminal battery life and reducing energy consumption. The invention also optimizes the packet loss handling mechanism, allowing for direct recovery of lost packets through redundant decoding in low-terminal-density areas to ensure broadcast continuity. The use of a geographic group time-division multiple access (GTA) mechanism to allocate uplink time slots adapts to long-distance propagation delays, improving data reporting success rates. In case of fixed terminal failure, mobile terminals can be authorized as dynamic relays to fill coverage gaps. Combined with Kalman filtering algorithms to predict link status and pre-optimize coding and routing parameters, the invention achieves wide coverage, low energy consumption, high stability, and easy maintenance for audio broadcasting in sparsely populated rural areas. Attached Figure Description

[0017] Figure 1 This is a schematic diagram of the structure of a multi-terminal network audio broadcasting system. Detailed Implementation

[0018] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0019] As mentioned in the background section of this application, research has found that existing multi-terminal network audio broadcasting systems, when implemented in rural scenarios, suffer from insufficient accuracy in path loss calculation. They rely solely on the free space loss model, resulting in large deviations in the estimation of link loss between terminals and unreasonable broadcast routing, thus exhibiting certain defects.

[0020] To address the aforementioned shortcomings, this application discloses a multi-terminal network audio broadcasting system that achieves accurate path loss calculation, adaptive redundancy coding, dynamic power adjustment, intelligent broadcast tree construction, efficient packet loss recovery, and dynamic relay blind spot filling, thereby improving the coverage, stability, and energy efficiency of audio broadcasting.

[0021] The following will describe in detail, with reference to the accompanying drawings, how the solution of this application solves the above-mentioned technical problems.

[0022] Please see Figure 1In this embodiment of the invention, a multi-terminal network audio broadcasting system includes a central server and multiple distributed audio terminals. The central server includes: a geographic information module for storing digital elevation models and the deployment locations of each audio terminal; an adaptive coding module for dynamically adjusting the redundancy coding rate of the audio stream based on link quality feedback reported by each audio terminal; and a routing calculation module for generating a multi-hop broadcast tree based on a terrain-compensated propagation loss model. Each audio terminal includes: a positioning module for collecting the real-time geographic coordinates of the audio terminal and reporting them to the central server; and a signal measurement module for periodically measuring the signal strength, signal-to-noise ratio, and packet loss rate received by the audio terminal from neighboring audio terminals or the central server, forming... The system includes: a link quality data transmission module; an audio decoding and playback module for receiving audio data packets, decoding them, and converting them into analog audio signals for output to the speaker; a data forwarding module for receiving audio data packets from parent nodes in the broadcast tree structure and forwarding them to child audio terminals in the downlink direction of the broadcast tree; and a status reporting module for packaging the location information, link quality data, remaining battery power, and current operating temperature of the audio terminal and sending them uplink to the central server according to preset time slots. The central server periodically receives the location and link quality data reported by each audio terminal, calculates the predicted path loss value between each pair of audio terminals, constructs a minimum spanning tree broadcast path based on the predicted loss value, and assigns a redundancy coefficient to each audio terminal. Through a collaborative architecture of a central server and distributed audio terminals, relying on a geographic information module to store terminal location and terrain data, an adaptive coding module to dynamically adapt the audio redundancy coding rate, and a routing calculation module to generate a multi-hop broadcast tree, combined with terminal positioning, signal measurement, data forwarding, and status reporting functions, the system can accurately obtain the location and link quality of each terminal in sparsely populated rural areas. Based on predicted path loss, it constructs a scientific minimum spanning tree broadcast path to achieve wide-area multi-hop audio broadcast coverage. At the same time, it rationally allocates redundancy coefficients to terminals, effectively improving the stability and coverage efficiency of audio transmission. This adapts to the widely distributed deployment scenarios in rural areas, enabling reliable transmission and unified scheduling and control of broadcast signals.

[0023] In this embodiment, the predicted path loss value Obtain it through the following methods: Acquire the terminal and terminal Based on the geographic coordinates and altitude, digital elevation model profile data is extracted from the line connecting the two terminals. All topographic peaks on this profile, along with their respective convex heights and Fresnel radii, are determined. The diffraction loss factor caused by each peak is calculated, and then the basic free-space loss is logarithmically summed with all diffraction loss factors. Specifically, The calculation expression is: ; in, For the terminal and The straight-line distance between them The radio frequency center frequency, At the speed of light, This represents the total number of terrain sharp peaks extracted from the digital elevation model profile. For the first The diffraction loss factor caused by the individual blade peaks, this factor By measuring the height of the blade peak protrusion First Fresnel radius at the edge peak The ratio is obtained by performing piecewise logarithmic operations, and when this ratio is less than a preset threshold, The value is zero.

[0024] Based on the terminal's geographic coordinates, altitude, and digital elevation model profile data, terrain peaks are extracted and diffraction loss factors are calculated. The predicted path loss value is obtained by logarithmically summing the free space basic loss and the diffraction loss of each peak. Compared with the traditional calculation method that only uses the free space loss model, this method significantly improves the accuracy of path loss estimation in complex terrains such as rural mountains and hills. It provides a real and reliable link loss basis for broadcast route construction, effectively avoids routing planning deviations caused by terrain occlusion, and ensures the transmission stability and coverage effectiveness of multi-hop broadcast links.

[0025] Specific examples: Two audio terminals in rural hilly areas , For example, let's set the calculation parameters: terminal spacing. Radio frequency center frequency speed of light Only one terrain edge peak exists on the cross section connecting the two terminals. The ratio of the height of the blade peak protrusion to the first Fresnel radius is less than a preset threshold, resulting in a diffraction loss factor. .

[0026] Substitute the parameters into the path loss calculation formula: ; Step-by-step calculation: Free space basics: , ; Diffraction loss term: ; Final path loss: .

[0027] The results incorporate the influence of terrain edge peaks, which is more in line with the actual transmission patterns of complex rural terrain compared to pure free space loss calculations, and provides a more accurate basis for route planning.

[0028] In this embodiment, the adaptive encoding module of the central server is based on the terminal The terminal is calculated based on the depth of the broadcast tree and the average path loss of its neighboring terminals. Corresponding audio data packet redundancy Specifically, redundancy By using the terminal parent node to Predicted path loss value With terminal Average path loss of neighboring nodes within a preset range around itself Perform a weighted summation and divide by the reference loss value. After normalization, the normalized result is input into a sigmoid function and mapped to the (0, 1) interval to obtain the result. The calculation expression is as follows: ; in, For standard logistic S-type functions, and The weights are preset and their sum is 1. This is a reference loss value, which is equal to the path loss calculation value when the distance in free space is the preset reference distance.

[0029] Based on the terminal's depth in the broadcast tree, the predicted path loss from the parent node to the terminal, and the average path loss of neighboring terminals, the redundancy of audio data packets is accurately calculated through weighted normalization and combined with a sigmoid function. This allows the redundancy coding rate to be dynamically and adaptively adjusted according to the actual link quality. When the link quality is poor, the redundancy is automatically increased to ensure reliable audio data transmission, while when the link quality is good, the redundancy is automatically reduced to save bandwidth resources. This completely solves the problem of bandwidth waste or severe packet loss caused by fixed redundancy coding, allowing audio transmission efficiency and anti-interference capability to achieve the optimal balance in complex rural terrain scenarios.

[0030] Specific examples: The aforementioned audio terminals for rural and hilly areas will continue to be used. In this scenario, set uniform calculation parameters: Weighting coefficients: (satisfy ); Reference Loss : Take reference distance Calculated according to the free space model ; From parent node to terminal Predicted path loss ; terminal Average path loss of neighboring nodes .

[0031] Substitute the parameters into the redundancy calculation formula: ; where the standard logistic S-type function is: ; Calculate the weighted sum of normalized losses: ; Substituting into the sigmoid function yields the redundancy: .

[0032] This redundancy can be precisely matched to the terminal. The actual link quality ensures both the ability to withstand packet loss in complex terrain and avoids excessive redundancy that would waste bandwidth.

[0033] In this embodiment, the audio terminal also includes a dynamic power adjustment module, which calculates a transmit power adjustment coefficient based on the terminal's remaining battery power and the number of downstream terminals in the broadcast tree. (where subscript) (This refers to any terminal); specifically, for any terminal Its power adjustment coefficient By normalizing the remaining energy ratio of the terminal Number of downstream terminals Sum the reciprocals of the terms, add 1, take the natural logarithm, and multiply by the sensitivity factor. Finally, subtract the product from 1 and multiply it by the reference transmit power to obtain the result. The calculation expression is as follows: ; in, Rated transmission power, This is an adjustment sensitivity factor with a value between 0 and 1. To normalize the remaining energy and set its value to [0, 1], This represents the number of downstream terminals that the current terminal is responsible for forwarding in the broadcast tree, and when... When it is zero, Process as zero.

[0034] The transmit power adjustment coefficient is dynamically calculated based on the normalized remaining power of the audio terminal and the number of downstream terminals in the broadcast tree. Under the premise of meeting the data forwarding coverage requirements of downstream terminals, the transmit power can be intelligently adjusted to avoid the energy waste caused by constant transmit power, effectively extend the terminal's battery life, adapt to application scenarios in rural areas with inconvenient power supply and high maintenance difficulty, and improve the stability and energy saving of the system in the long term.

[0035] Specific examples: Using the aforementioned audio terminal scenario in rural hilly areas, for any terminal Perform power adjustment calculations and set parameters uniformly: Rated reference transmit power ; Power adjustment sensitivity factor ; Terminal x Normalized Remaining Energy ; Terminal x Number of downstream terminals .

[0036] Substitute the parameters into the formula for calculating the transmit power adjustment coefficient: ; Calculate the ratio of electricity consumption to the number of downstream terminals: ; Calculate the inner summation term of the logarithm: ; Calculate the natural logarithm: ; Calculate the sensitivity factor product: ; Calculate the power adjustment ratio: ; Final adjustment of transmit power: .

[0037] This calculation, while ensuring normal forwarding by the five downstream terminals, appropriately reduces the transmission power, thus meeting coverage requirements while saving power and achieving a balance between battery life and transmission performance.

[0038] In this embodiment, the routing calculation module predicts the path loss value when constructing the minimum spanning tree broadcast path. As a benchmark for edge weights, and in conjunction with the terminal and terminal Each terminal's dynamic power adjustment coefficient corrects the edge weights, where the terminal... Power adjustment coefficient and terminal Power adjustment coefficient According to respectively The corrected edge weights are calculated in the following manner. The Prim algorithm is used to generate a broadcast tree with the central server as the root node.

[0039] Using the predicted path loss value as the edge weight benchmark, and combining the dynamic power adjustment coefficients of the terminals at both ends to correct the edge weights, the Prim algorithm is used to generate a broadcast tree with the central server as the root node. This can optimize the routing plan by linking the terrain path loss with the real-time transmission power status of the terminals, making the broadcast tree topology more in line with the complex terrain of rural areas and the actual operating conditions of terminal energy consumption and power. This significantly improves the rationality of broadcast routing, transmission stability and coverage effectiveness, and avoids the problems of power mismatch, transmission interruption or coverage blind spots caused by planning routes solely based on path loss.

[0040] Specific examples: Using the same scene parameters for rural hilly areas as mentioned above, select the terminal With audio terminal Perform edge weight calculation: terminal , Inter-path loss value ; terminal Dynamic transmit power ; terminal Dynamic transmit power (Calculated values ​​for terminals of the same specifications in the same scenario).

[0041] Substitute the parameters into the corrected weight calculation formula: ; Calculate the power product term: ; Calculate the corrected weights: .

[0042] Based on the corrected edge weights, the routing calculation module uses the Prim algorithm to traverse all terminals and prioritizes the links with smaller edge weights to build a broadcast tree. This ensures that the broadcast path takes into account both terrain loss and terminal power status, adapting to the broadcast transmission needs of sparsely populated rural areas.

[0043] In this embodiment, after receiving an audio data packet, the audio terminal first uses a signal measurement module to check the sequence number of the data packet. If a missing packet is found, a packet retransmission request is initiated to the parent node or neighboring node through the status reporting module. If the terminal density in the area where the current terminal is located is lower than a preset threshold, a redundant decoding mode is activated, utilizing... The corresponding forward error correction coding directly recovers the missing data packets without initiating a retransmission request.

[0044] A packet loss handling mechanism that adaptively switches between packet loss retransmission and redundant decoding is adopted. When the terminal density is normal, packet loss retransmission is initiated to ensure data accuracy. When the terminal density is lower than a preset threshold due to the vast rural area and sparse population, the redundant decoding mode is automatically activated. Missing data packets are directly recovered by forward error correction coding without the need to initiate retransmission requests. This effectively solves the problems of retransmission failure and broadcast interruption caused by the sparse terminal population in remote areas, and significantly improves the continuity and reliability of audio transmission.

[0045] Specific examples: Using the same scenario as described above for rural hilly areas, with audio terminals Example of packet loss handling demonstration: Preset terminal density threshold: 2 units / square kilometer; terminal Actual terminal density in the area: 1 unit / square kilometer (below the threshold). terminal Corresponding audio redundancy: .

[0046] When the terminal When receiving audio data packets, the signal measurement module detects a missing data packet sequence number. Because the terminal density in the area is lower than a preset threshold, it does not initiate a retransmission request to the parent / neighboring node, but directly starts the redundancy decoding mode based on the redundancy. The corresponding forward error correction coding recovers missing data packets in real time, completes audio data restoration, and then decodes and plays the audio normally, ensuring smooth and uninterrupted broadcasting.

[0047] In this embodiment, the central server also includes a broadcast content segmentation module. This module dynamically segments the audio stream into data blocks of varying lengths based on the average terminal response time of each sub-region in the sparsely populated rural area. The data block length corresponding to terminals far from the central server is less than a preset value to reduce single-packet retransmission overhead. Specifically, for broadcast tree depths of... The terminal, its data block length By measuring the length of the base block With depth The result is obtained by multiplying the exponential decay function, and the base of the exponential decay function is determined by the ratio between the average moving speed of the terminal in the region and the audio playback buffer duration.

[0048] Based on the average terminal response time of each sub-region in sparsely populated rural areas, and combined with the broadcast tree depth, the audio stream is dynamically divided into data blocks of unequal length. This results in shorter data blocks for terminals far from the central server, effectively reducing single-packet retransmission overhead and transmission latency. It adapts to the transmission characteristics of wide-area multi-hop broadcasting in rural areas, avoiding the problems of long data packets being easily lost and having high retransmission costs in weak links and long-distance transmission, and significantly improving the playback smoothness and overall transmission efficiency of audio broadcasts.

[0049] Specific examples: Using the same unified broadcasting scenario for rural hilly areas as described above, the broadcasting tree depth is... The terminal performs audio data block length calculation: Basic data block length ; Average moving speed of terminals in this area Audio playback buffering time The base of the exponential decay function ; Select a location far from the central server and a broadcast tree depth. The terminal.

[0050] Substitute the parameters into the data block length calculation formula: ; Calculate the exponential decay term: ; Calculate the target data block length: .

[0051] The data block length of the long-distance terminal with a depth of 3 is only 25ms, which is much shorter than the basic block length. This can significantly reduce the amount of retransmission data and the time spent after a single packet is lost, making it suitable for the transmission characteristics of long-distance weak links in sparsely populated rural areas.

[0052] In this embodiment, when the audio terminal's status reporting module reports signal measurement data, it adopts a time-division multiple access mechanism based on geographic grouping: with the central server as the origin, the rural area is divided into several concentric rings. Terminals within the same ring use the same uplink time slot, and different rings are allocated different time slot lengths according to their radius. The larger the radius of the ring, the longer the time slot is allocated, in order to adapt to the large difference in propagation delay between terminals far from the center and the central server under the conditions of vast areas and sparse population.

[0053] The system adopts a time-division multiple access (TDMA) reporting mechanism based on geographic grouping. The rural area is divided into concentric rings with the central server as the origin. Terminals within the same ring share the same uplink time slot. Rings with larger radii are allocated longer time slots. This can accurately adapt to the differences in propagation delays of long-distance terminals in sparsely populated rural areas, avoid uplink conflicts and data loss caused by unified time slots, significantly improve the success rate and stability of uplink transmission of terminal status data, and optimize the efficiency of time slot resource utilization.

[0054] Specific examples: Using the same unified broadcasting scenario for rural hilly areas, a three-level concentric ring system is defined with the central server as the origin. The time slot length is set to be proportional to the ring radius, and the calculation formula is as follows: ;in: This is the time slot scaling factor. Where is the radius of the annulus. This refers to the uplink time slot length allocated to the corresponding circular ring.

[0055] Inner ring radius Time slot length: ; Middle Ring Radius Time slot length: ; Outer ring radius Time slot length: ; The outer ring is far from the server and has a greater propagation delay. It is allocated the longest time slot, which can ensure that status data such as terminal positioning, link quality, and power consumption are stably uploaded to the central server, adapting to the transmission characteristics of wide-area coverage in rural areas.

[0056] In this embodiment, the system also includes a mobile terminal relay mode. When a fixed audio terminal goes offline due to a malfunction or power depletion, the central server detects that there are still uncovered terminals downstream of the terminal. It then temporarily authorizes mobile terminals installed on agricultural machinery or vehicles entering the broadcast area as dynamic relay nodes. The weight of the dynamic relay node is calculated by the matching degree between its movement trajectory and the direction of the missing link. The matching degree is obtained by weighted summation of the cosine value of the vector angle between the current position of the mobile terminal and the position of the offline terminal, as well as the projection component of the movement speed in the direction of the connection.

[0057] The addition of a mobile terminal relay mode allows the central server to temporarily authorize mobile terminals mounted on agricultural machinery and vehicles within the broadcast area as dynamic relay nodes when fixed audio terminals go offline due to malfunction or power depletion and there are still uncovered terminals downstream. The relay weight is calculated by matching the movement trajectory with the direction of the missing link, which can quickly fill the coverage blind spots caused by the malfunctioning terminal and prevent downstream terminals from losing contact. This significantly improves the fault self-healing capability and coverage continuity of the broadcast system in sparsely populated rural areas. The link filling can be completed without the need to deploy additional fixed terminals, effectively reducing the system deployment and maintenance costs.

[0058] Specific examples: Using the aforementioned unified broadcasting scenario in rural hilly areas, we assume that fixed audio terminal A goes offline due to battery depletion, and there are uncovered terminals downstream of it. The central server triggers a mobile relay mechanism, selecting mobile terminals M mounted on agricultural machinery within the area as candidate dynamic relays, and calculating relay weights (matching degree). Weighting coefficients: (Cosine weight of the angle between vectors) (Velocity projection weights); The cosine of the angle between the position vectors of mobile terminal M and offline terminal A: ; The projected component of the mobile terminal speed in the direction of the missing link: ; Relay weight (matching degree) calculation formula: ; Calculate the cosine weighted term of the included angle: ; Calculate the velocity projection weighting term: ; Final relay weights: .

[0059] The central server determines that the matching degree of mobile terminal M meets the standard based on the relay weight (which is greater than the preset value), and authorizes it as a dynamic relay node to take over the forwarding task of offline terminal A, so as to ensure that downstream terminals can receive broadcast signals normally and maintain coverage continuity.

[0060] In this embodiment, the central server also includes an audio broadcast quality prediction module. This module uses a Kalman filter algorithm to predict the link status of each terminal in the next broadcast cycle based on the packet loss rate, retransmission count, and signal attenuation trend of each terminal in historical broadcast data. It also adjusts the audio coding bitrate, forward error correction redundancy, and broadcast tree topology in advance based on the prediction results. The state transition matrix of the Kalman filter is dynamically updated according to the time-varying laws of temperature, humidity, and electromagnetic wave attenuation coefficient caused by vegetation growth in typical daily variations in sparsely populated rural areas.

[0061] The audio broadcast quality prediction module uses historical packet loss rate, retransmission count, and signal attenuation trend to predict the terminal link status of the next broadcast cycle using the Kalman filter algorithm. It also dynamically updates the state transition matrix based on the time-varying laws of electromagnetic wave attenuation caused by rural temperature, humidity, and vegetation growth. This allows for precise pre-adjustment of audio coding bitrate, forward error correction redundancy, and broadcast tree topology, achieving proactive optimization of link quality. This effectively offsets transmission fluctuations caused by time-varying rural environments, significantly improves the stability and continuity of audio broadcasting, and ensures long-term reliable operation of the system.

[0062] Specific examples: Using the aforementioned unified broadcasting scenario in rural hilly areas, with audio terminals For example, let's set the basic parameters uniformly: Previous cycle terminal Packet loss rate: ; State transition matrix: (Updated dynamically based on daily temperature, humidity, and vegetation decay); Environmental random noise: ; Previous period prediction error covariance (initial normal value): ; Next cycle packet loss rate prediction: ; Prediction error covariance update: ; Packet loss rate prediction: ; Covariance calculation in steps: Step 1: Calculate the square of the state transition matrix ; Step 2: Multiply by the periodic covariance ; Step 3: Add environmental noise item ; final: ; It is a reliability indicator of the prediction results. The smaller the value, the more accurate the prediction. Updating it means taking into account the impact of environmental interference and terrain attenuation, so that the link prediction in the next cycle is more in line with the actual rural scenario.

[0063] This invention accurately calculates path loss between terminals through terrain compensation, enabling the construction of reasonable broadcast routes tailored to complex rural terrain. It also adaptively adjusts the audio redundancy coding rate based on link quality and broadcast tree depth, balancing packet loss recovery with bandwidth utilization efficiency. Furthermore, it dynamically adjusts the transmission power based on the terminal's remaining battery power and the number of downstream terminals, effectively extending terminal battery life and reducing energy consumption. The invention also optimizes the packet loss handling mechanism, allowing for direct recovery of lost packets through redundant decoding in low-terminal-density areas to ensure broadcast continuity. The use of a geographic group time-division multiple access (GTA) mechanism to allocate uplink time slots adapts to long-distance propagation delays, improving data reporting success rates. In case of fixed terminal failure, mobile terminals can be authorized as dynamic relays to fill coverage gaps. Combined with Kalman filtering algorithms to predict link status and pre-optimize coding and routing parameters, the invention achieves wide coverage, low energy consumption, high stability, and easy maintenance for audio broadcasting in sparsely populated rural areas.

[0064] Obviously, those skilled in the art can make various modifications and variations to this invention without departing from its spirit and scope. Therefore, if these modifications and variations fall within the scope of the claims of this invention and their equivalents, this invention also intends to include these modifications and variations.

[0065] The above description is merely a preferred embodiment of the present invention, but the scope of protection of the present invention is not limited thereto. Any equivalent substitutions or modifications made by those skilled in the art within the scope of the technology disclosed in the present invention, based on the technical solution and inventive concept of the present invention, should be covered within the scope of protection of the present invention.

Claims

1. A multi-terminal network audio broadcasting system, characterized in that, It includes a central server and multiple distributed audio terminals; The central server includes: a geographic information module for storing digital elevation models and the deployment locations of each audio terminal; an adaptive coding module for dynamically adjusting the redundancy coding rate of the audio stream based on the link quality feedback reported by each audio terminal; and a routing calculation module for generating a multi-hop broadcast tree based on a terrain-compensated propagation loss model. The audio terminal includes: a positioning module for collecting the real-time geographical coordinates of the audio terminal and reporting them to the central server; a signal measurement module for periodically measuring the signal strength, signal-to-noise ratio, and packet loss rate received by the audio terminal from neighboring audio terminals or the central server, forming link quality data; an audio decoding and playback module for receiving audio data packets, decoding them, and converting them into analog audio signals for output to the speaker; a data forwarding module for receiving audio data packets from the parent node in the broadcast tree structure and forwarding them to the child node audio terminals in the downlink direction of the broadcast tree; and a status reporting module for packaging the location information, link quality data, remaining battery power, and current operating temperature of the audio terminal and sending them uplink to the central server according to a preset time slot. The central server periodically receives location and link quality data reported by each audio terminal, calculates the predicted path loss value between each pair of audio terminals, constructs the minimum spanning tree broadcast path based on the predicted loss value, and assigns a redundancy coefficient to each audio terminal.

2. The multi-terminal network audio broadcasting system according to claim 1, characterized in that, The predicted path loss value Obtain it through the following methods: Acquire the terminal and terminal Based on the geographic coordinates and altitude, digital elevation model profile data is extracted from the line connecting the two terminals. All topographic peaks on this profile, along with their respective convex heights and Fresnel radii, are determined. The diffraction loss factor caused by each peak is calculated, and then the basic free-space loss is logarithmically summed with all diffraction loss factors. Specifically, The calculation expression is: ； in, For the terminal and The straight-line distance between them The radio frequency center frequency, At the speed of light, This represents the total number of terrain sharp peaks extracted from the digital elevation model profile. For the first The diffraction loss factor caused by the individual blade peaks, this factor By measuring the height of the blade peak protrusion First Fresnel radius at the edge peak The ratio is obtained by performing piecewise logarithmic operations, and when this ratio is less than a preset threshold, The value is zero.

3. The multi-terminal network audio broadcasting system according to claim 2, characterized in that, The adaptive encoding module of the central server is based on the terminal. The terminal is calculated based on the depth of the broadcast tree and the average path loss of its neighboring terminals. Corresponding audio data packet redundancy Specifically, redundancy By using the terminal parent node to Predicted path loss value With terminal Average path loss of neighboring nodes within a preset range around itself Perform a weighted summation and divide by the reference loss value. After normalization, the normalized result is input into a sigmoid function and mapped to the (0, 1) interval to obtain the result. The calculation expression is as follows: ； in, For standard logistic S-type functions, and The weighting coefficients are preset and their sum is 1. This is a reference loss value, which is equal to the path loss calculation value when the distance in free space is the preset reference distance.

4. A multi-terminal network audio broadcasting system according to claim 3, characterized in that, The audio terminal also includes a dynamic power adjustment module, which calculates a transmission power adjustment coefficient based on the terminal's remaining battery power and the number of downstream terminals in the broadcast tree. Specifically, for any terminal Its power adjustment coefficient By normalizing the remaining energy ratio of the terminal Number of downstream terminals Sum the reciprocals of the terms, add 1, take the natural logarithm, and multiply by the sensitivity factor. Finally, subtract the product from 1 and multiply it by the reference transmit power to obtain the result. The calculation expression is as follows: ； in, Rated transmission power, This is an adjustment sensitivity factor with a value between 0 and 1. To normalize the remaining energy and set its value to [0, 1], This represents the number of downstream terminals that the current terminal is responsible for forwarding in the broadcast tree, and when... When it is zero, Process as zero.

5. A multi-terminal network audio broadcasting system according to claim 4, characterized in that, The routing calculation module predicts path loss values ​​when constructing the minimum spanning tree broadcast path. As a benchmark for edge weights, and in conjunction with the terminal and terminal Each terminal's dynamic power adjustment coefficient corrects the edge weights, where the terminal... Power adjustment coefficient and terminal Power adjustment coefficient According to respectively The corrected edge weights are calculated in the following manner. The Prim algorithm is used to generate a broadcast tree with the central server as the root node.

6. A multi-terminal network audio broadcasting system according to claim 5, characterized in that, After receiving an audio data packet, the audio terminal first uses a signal measurement module to check the packet's sequence number. If a missing packet is found, it initiates a packet retransmission request to its parent or neighboring nodes via a status reporting module. If the terminal density in the area where the current terminal is located is lower than a preset threshold, a redundant decoding mode is activated, utilizing... The corresponding forward error correction coding directly recovers the missing data packets without initiating a retransmission request.

7. A multi-terminal network audio broadcasting system according to claim 6, characterized in that, The central server also includes a broadcast content segmentation module. This module dynamically segments the audio stream into data blocks of varying lengths based on the average terminal response time of each sub-region in the sparsely populated rural area. The data blocks corresponding to terminals far from the central server are shorter than a preset value to reduce single-packet retransmission overhead. Specifically, for broadcast trees with a depth of... The terminal, its data block length By measuring the length of the base block With depth The result is obtained by multiplying the exponential decay function, and the base of the exponential decay function is determined by the ratio between the average moving speed of the terminal in the region and the audio playback buffer duration.

8. A multi-terminal network audio broadcasting system according to claim 7, characterized in that, When the audio terminal's status reporting module reports signal measurement data, it adopts a time-division multiple access mechanism based on geographic grouping: with the central server as the origin, the rural area is divided into several concentric rings. Terminals within the same ring use the same uplink time slot, and different rings are allocated different time slot lengths according to their radius. The larger the radius of the ring, the longer the time slot is allocated, in order to adapt to the large difference in propagation delay between terminals far from the center and the central server under the conditions of vast areas and sparse population.

9. A multi-terminal network audio broadcasting system according to claim 8, characterized in that, The system also includes a mobile terminal relay mode. When a fixed audio terminal goes offline due to a malfunction or depletion of power, the central server detects that there are still uncovered terminals downstream of the terminal. It then temporarily authorizes mobile terminals installed on agricultural machinery or vehicles entering the broadcast area as dynamic relay nodes. The weight of the dynamic relay node is calculated by the matching degree between its movement trajectory and the direction of the missing link. The matching degree is obtained by weighted summation of the cosine of the vector angle between the current position of the mobile terminal and the position of the offline terminal, as well as the projection component of the movement speed in the direction of the connection.

10. A multi-terminal network audio broadcasting system according to claim 9, characterized in that, The central server also includes an audio broadcast quality prediction module. This module uses a Kalman filter algorithm to predict the link status of each terminal in the next broadcast cycle based on the packet loss rate, retransmission count, and signal attenuation trend of each terminal in historical broadcast data. It also adjusts the audio coding bitrate, forward error correction redundancy, and broadcast tree topology in advance based on the prediction results. The state transition matrix of the Kalman filter is dynamically updated according to the time-varying laws of temperature, humidity, and electromagnetic wave attenuation coefficient caused by vegetation growth in typical daily variations in sparsely populated rural areas.

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