A combined coding system for rescue soldiers based on audio, video and satellite communications

Abstract

The present invention discloses a combined coding system for a single-soldier rescuer based on audio, video and satellite communications, and belongs to the field of emergency communication technology. The combined coding system for a single-soldier rescuer based on audio, video and satellite communications includes an information acquisition module, a signal coding module and a combined coding module. The present invention solves the problem that the prior art uses multi-network communication equipment for signal transmission, which easily leads to unstable or interrupted communication signals in areas with complex terrain, thus affecting the effect of emergency rescue. The present invention ensures the accuracy and real-time nature of rescue information, enables rapid transmission of rescue information, and can provide a richer and more intuitive communication means for the rescue site, which helps the command center to understand the on-site situation more accurately and make timely and effective decisions. It can overcome the limitations of the ground communication network and ensure that communication can be kept unimpeded under harsh conditions, greatly expanding the coverage of rescue operations and improving rescue efficiency.

Description

Technical Field

[0001] The invention relates to the technical field of emergency communication, in particular to a rescue single-soldier combined-channel coding system based on audio, video and satellite communications. Background Art

[0002] Emergency communication refers to the comprehensive use of various communication resources to ensure the communication means and methods required for rescue, emergency assistance and necessary communications when natural or man-made sudden emergencies occur, including when communication needs increase suddenly. It is a temporary special communication mechanism provided to respond to natural or man-made emergencies.

[0003] The Chinese patent application with publication number CN119110253A discloses a multi-network integrated emergency communication command and dispatch system, including terminal acquisition equipment, multi-network communication equipment, on-site command center, remote communication link and remote dispatch center connected in sequence; the terminal acquisition equipment collects video signals, voice data, sensor data and positioning data information on site; the multi-network communication equipment includes LTE private network communication equipment, LoRA private network equipment and wireless ad hoc network communication equipment; the LTE private network communication equipment provides broadband wireless communication services and supports the transmission of video, voice and data; the LoRA private network communication equipment connects to low-power devices and collects data, and sends the collected data through LoRA communication; the wireless ad hoc network communication equipment establishes a temporary communication network for the terminal acquisition equipment in the absence of fixed network facilities; the on-site command center is used to perform multi-link aggregation, perform on-site dispatch according to the collected data information, and send it to the remote dispatch center through a remote communication link.

[0004] In actual use, the above patent application utilizes multi-network communication equipment for signal transmission, which can easily lead to unstable or interrupted communication signals in areas with complex terrain, thus affecting the effectiveness of emergency rescue. Therefore, it does not meet existing needs. In response to this, we have proposed a single-soldier rescue combined coding system based on audio, video and satellite communications. Summary of the invention

[0005] The purpose of the present invention is to provide a single-soldier rescue combined coding system based on audio, video and satellite communications, which ensures the accuracy and real-time nature of rescue information, enables rescue information to be transmitted quickly, and provides a more abundant and intuitive communication means for the rescue site, which helps the command center to understand the on-site situation more accurately and make timely and effective decisions. Audio and video transmission through satellite communication equipment can overcome the limitations of ground communication networks and ensure that communication can be maintained unimpeded under adverse conditions, greatly expanding the coverage of rescue operations, improving rescue efficiency, and solving the problems raised in the above-mentioned background technology.

[0006] To achieve the above object, the present invention provides the following technical solution: a rescue single-soldier combined coding system based on audio, video and satellite communications, comprising:

[0007] The information acquisition module is used to collect multi-channel dynamic video using the camera equipment carried by the rescue soldier, and collect audio signals through the communication equipment;

[0008] A signal encoding module, used to encode audio and video signals and convert them into audio and video digital signals;

[0009] The combining and encoding module is used to combine the converted audio and video digital signals using asynchronous time division multiplexing technology to restore them to the original digital audio and video information, specifically including:

[0010] Divide the transmission channel time into multiple time slots, each time slot corresponds to a signal transmission, and set the time slot allocation principle;

[0011] The encoded audio and video digital signals are transmitted and multiplexed in the allocated time slots according to the set time slot allocation principle to generate multiplexed signals, and the multiplexed signals are combined into one transmission line for transmission through a combiner;

[0012] At the receiving end, the multiplexed signal is separated in time sequence by a demultiplexer to restore the original audio and video signal;

[0013] The restored original audio and video signals are converted into analog signals using a satellite modem, and the converted analog signals are transmitted to the ground command center via satellite communication equipment;

[0014] At the ground command center, the satellite modem converts the received analog signal back into digital audio and video signals, decodes the converted digital audio and video signals, and restores the decoded digital audio and video signals to the original digital audio and video information.

[0015] Preferably, the signal encoding module comprises:

[0016] The audio encoding module is used to process the audio signal obtained by the rescue soldier using the audio processor and encode the processed audio signal, specifically including:

[0017] The audio processor is used to amplify, filter and reduce noise of the acquired audio signal, thereby filtering out background noise in the audio signal;

[0018] Perform dynamic compression and limiting processing on the audio signal to reduce the dynamic range of the audio signal, increase the density of the sound, and make the peak amplitude of the audio signal uniform and consistent;

[0019] Encoding the processed audio signal, and converting the processed audio signal into an audio digital signal by using a pulse code modulation method;

[0020] The video encoding module is used to process the video signal obtained by the rescue soldier using a video processor and encode the processed video signal.

[0021] Preferably, the audio signal is dynamically compressed, specifically comprising:

[0022] Extract the volume intensity corresponding to each frame of the audio signal in the audio signal;

[0023] Extract the expected output volume level;

[0024] Setting a volume threshold according to the original intensity of the audio signal and the preset segmentation point intensity combined with the expected output volume intensity;

[0025] Setting a first ratio of volume compression after exceeding the volume threshold according to the volume threshold;

[0026] The first ratio is obtained by the following formula:

[0027] ;

[0028] Among them, R 1 Represents the first ratio; R 0 Indicates the preset initial ratio; Y g Indicates the volume threshold; Y 0 Indicates the preset initial volume threshold reference value; L min and L max Indicates the minimum and maximum volume levels of the audio signal; F p Indicates the average frequency corresponding to the audio signal; F max Indicates the maximum frequency value reached by the audio signal;

[0029] Comparing the audio signal with a preset volume threshold, and intercepting the audio signal whose volume intensity exceeds the preset volume threshold as the observed audio signal;

[0030] Whether to perform audio signal compression according to a first ratio is determined according to the ratio of the observed audio signal.

[0031] Preferably, the volume threshold is set according to the original intensity of the audio signal and the preset segmentation point intensity in combination with the expected output volume intensity, specifically including:

[0032] Extracting the original intensity of the audio signal and the expected output volume intensity of the audio signal;

[0033] Extracting the audio intensity corresponding to the preset segmentation points; wherein the audio intensity corresponding to the segmentation points are respectively the first segmentation point intensity and the second segmentation point intensity, and the first segmentation point intensity is lower than the second segmentation point intensity;

[0034] Obtaining a volume threshold by using the original intensity of the audio signal and the expected output volume intensity of the audio signal in combination with the intensity of the first segmentation point and the intensity of the second segmentation point;

[0035] The volume threshold is obtained by the following formula:

[0036] ;

[0037] Among them, Y g Indicates the volume threshold; Y 0 Indicates the preset initial volume threshold reference value; L in and L out Respectively represent the original intensity of the audio signal and the expected output volume intensity of the audio signal; L 01 and L 02 Represent the first segment point intensity and the second segment point intensity respectively.

[0038] Preferably, determining whether to perform audio signal compression according to a first ratio according to the ratio of the observed audio signal specifically includes:

[0039] extracting a ratio of the observed audio signal and comparing it with a preset ratio threshold;

[0040] When the ratio of the observed audio signal is lower than a preset ratio threshold, it is determined to compress the audio signal according to a first ratio;

[0041] When the ratio of the observed audio signal is not lower than a preset ratio threshold, adjusting the first ratio by using the audio intensity of each frame of the audio signal included in the audio signal to obtain a second ratio;

[0042] The second ratio is obtained by the following formula:

[0043] ;

[0044] Among them, R 2 Represents the second ratio; R 1 represents the first ratio; n represents the number of frame audio signals contained in the audio signal; L i represents the audio intensity of the i-th frame audio signal; Y g represents the volume threshold; P represents the ratio of the observed audio signal; P y Indicates the preset ratio threshold; L b Indicates the standard deviation of the audio intensity of n frames of audio signal; Y 0Indicates the preset initial volume threshold reference amount.

[0045] Preferably, the video encoding module specifically includes:

[0046] The original uncompressed video signal transmitted by the rescue soldier is obtained. The video signal consists of continuous image frames, and each frame of the video image is divided into macro blocks of 16x16 pixels;

[0047] Motion estimation technology is used to identify the displacement of similar regions between adjacent frames and only encode the motion information and residuals;

[0048] In the same frame, the content of the current macroblock is predicted based on the surrounding pixels and the residual is encoded again. This method is particularly suitable for static or less changing areas.

[0049] The macroblock in pixel domain is converted to frequency domain by using discrete cosine transform method, and the transformed coefficients are quantized;

[0050] The continuous values ​​are mapped to finite discrete values ​​and then encoded to convert the quantized discrete values ​​into binary codes.

[0051] Preferably, the combined encoding module includes:

[0052] A time slot allocation module is used to allocate time slots to transmission channels according to the transmission requirements of audio and video digital signals;

[0053] The multiplexing and combining module is used to multiplex the audio and video signals in the allocated time slots after completing the signal and time slot allocation, and combine the multiplexed signals into one transmission line through a combiner;

[0054] A signal conversion module is used to convert the acquired audio and video digital signals into analog signals for transmission, and demodulate the received analog signals back into audio and video digital signals;

[0055] Network interface, used for communication between rescue soldiers and satellite communication equipment, and between satellite communication equipment and ground command center.

[0056] Preferably, the multiplexing and combining module includes:

[0057] The encoded audio and video digital signals are transmitted and multiplexed in the allocated time slots according to the set time slot allocation principle, and the multiplexed signals are combined into one transmission line for transmission through a combiner;

[0058] When the multiplexed signal is transmitted, the multiplexer is used to scan the input line of the multiplexed signal in turn. If there is a signal in the line, the signal is placed in the asynchronous time division multiplexing frame. If there is no signal in the line, the line is skipped.

[0059] When the asynchronous time division multiplexing frame is full of signals, the multiplexed signals are separated in time sequence by a demultiplexer at the receiving end to restore the original audio and video signals.

[0060] Preferably, the signal conversion module specifically includes:

[0061] The restored original audio and video signals are converted into analog signals using a satellite modem, and the converted analog signals are transmitted to the ground command center via satellite communication equipment;

[0062] At the ground control center, the satellite modem restores the characteristics of the analog signal, extracts the original audio and video digital signal, and converts the analog signal back into digital information based on the extracted original audio and video digital signal;

[0063] Decoding the converted audio and video digital signals, and restoring the decoded audio and video digital signals to the original digital audio and video information;

[0064] The restored digital audio and video information is played in real time by the ground command center, and is restored to audio signals through communication equipment for conversations.

[0065] Preferably, the time slot allocation module specifically includes:

[0066] Detect audio and video signals to determine the transmission requirements of audio and video signals, including data transmission volume and transmission rate;

[0067] Allocate time slots for audio and video signals according to the transmission requirements of audio and video signals. When the audio and video signals need to transmit signals, time slots are allocated. If the audio and video signals do not need to transmit signals, no time slots are allocated.

[0068] Adjust the allocated time slot width according to the amount of audio and video signal transmission, and adjust the time slot allocation strategy according to the priority and real-time requirements of the audio and video signals;

[0069] When the transmission of the audio and video signals is completed, the time slots occupied by the audio and video signals are released, and the released time slots continue to be occupied by other audio and video signals that need to be transmitted.

[0070] Compared with the prior art, the present invention has the following beneficial effects:

[0071] The present invention utilizes audio processor and video processor to process the signal, thus ensuring the accuracy and real-time nature of the rescue information. The satellite modem builds a bridge for the communication between the rescue soldier and the ground command center, so that the rescue information can be transmitted quickly. The combined coding technology can integrate the audio and video coding and decoding technology, so that the rescue soldier can transmit the video, voice and data in real time through the satellite communication equipment, thus providing the rescue site with richer and more intuitive communication means, helping the command center to understand the on-site situation more accurately and make timely and effective decisions. The audio and video transmission through the satellite communication equipment can overcome the limitations of the ground communication network, thus ensuring the smooth communication under adverse conditions, thus greatly expanding the coverage of the rescue operation and improving the rescue efficiency. BRIEF DESCRIPTION OF THE DRAWINGS

[0072] Figure 1 It is a schematic diagram of the module of the rescue single-soldier combined coding system based on audio, video and satellite communication of the present invention;

[0073] Figure 2 The present invention is a schematic diagram of a method of a rescue single-soldier combined coding system based on audio, video and satellite communications. DETAILED DESCRIPTION

[0074] The following will be combined with the drawings in the embodiments of the present invention to clearly and completely describe the technical solutions in the embodiments of the present invention. Obviously, the described embodiments are only part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the present invention.

[0075] In order to solve the problem that the existing technology uses multi-network communication equipment for signal transmission, which is easy to cause unstable or interrupted communication signals in areas with complex terrain, thus affecting the effectiveness of emergency rescue, please refer to Figure 1-Figure 2 , this embodiment provides the following technical solutions:

[0076] The rescue single-soldier combined coding system based on audio, video and satellite communications includes:

[0077] The information acquisition module is used to collect multi-channel dynamic video using the camera equipment carried by the rescue soldier, and collect audio signals through the communication equipment;

[0078] A signal encoding module, used to encode audio and video signals and convert them into audio and video digital signals;

[0079] The combining and encoding module is used to combine the converted audio and video digital signals using asynchronous time division multiplexing technology to restore them to the original digital audio and video information, specifically including:

[0080] Divide the transmission channel time into multiple time slots, each time slot corresponds to a signal transmission, the length and number of time slots are determined according to the signal transmission rate and bandwidth requirements, and set the time slot allocation principle;

[0081] The encoded audio and video digital signals are transmitted in the allocated time slots according to the set time slot allocation principle. The audio and video signals are multiplexed during transmission, and the multiplexed signals are combined into one transmission line for transmission through a combiner to ensure that each signal does not overlap on the time axis to avoid interference between signals. Each signal only occupies the channel for data transmission in its own time slot, which ensures that multiple signals share the same channel but do not conflict with each other.

[0082] When the multiplexed signal is transmitted, the multiplexer is used to scan the input line of the multiplexed signal in turn. If there is a signal in the line, the signal is placed in the asynchronous time division multiplexing frame. If there is no signal in the line, the line is skipped.

[0083] When the asynchronous time-division multiplexing frame is full of signals, the multiplexed signals are separated in time sequence by a demultiplexer at the receiving end to restore the original audio and video signals;

[0084] The restored original audio and video signals are converted into analog signals using a satellite modem, and the converted analog signals are transmitted to the ground command center via satellite communication equipment;

[0085] At the ground command center, the satellite modem converts the received analog signal back into digital audio and video signals, decodes the converted digital audio and video signals, and restores the decoded digital audio and video signals to the original digital audio and video information for subsequent use.

[0086] By using asynchronous time division multiplexing technology, time slots can be dynamically allocated according to the actual data transmission requirements of the signal source, thereby improving channel utilization and reducing resource waste.

[0087] Signal encoding module, including:

[0088] An audio encoding module is used to process the audio signal obtained by the rescue soldier using an audio processor and encode the processed audio signal;

[0089] The video encoding module is used to process the video signal obtained by the rescue soldier using a video processor and encode the processed video signal.

[0090] Audio encoding module, including:

[0091] The audio processor is used to amplify, filter and reduce noise of the acquired audio signal, and filter the background noise in the audio signal. Through these processes, the quality of the audio signal can be enhanced, and a clearer and more pleasant sound effect can be achieved. The clarity of the audio can be improved, and key information can be more prominent, which is crucial for rescue communications and can ensure the accurate transmission of instructions and information.

[0092] Dynamically compress and limit the audio signal to reduce the dynamic range of the audio signal, increase the density of the sound, and make the peak amplitude of the audio signal uniform, which helps to maintain the clarity and intelligibility of the sound in a noisy environment;

[0093] The processed audio signal is encoded and converted into an audio digital signal using pulse code modulation.

[0094] The audio processor can effectively manage and control the balance between each channel, provide better stereo effect and sense of space. By processing the audio signals obtained by the rescue soldiers, it can improve the audio quality, suppress noise, enhance the voice signal, and ensure the clarity and intelligibility of the audio in complex environments.

[0095] Specifically, the audio signal is dynamically compressed, including:

[0096] Extract the volume intensity corresponding to each frame of the audio signal in the audio signal;

[0097] Extract the expected output volume level;

[0098] Setting a volume threshold according to the original intensity of the audio signal and the preset segmentation point intensity combined with the expected output volume intensity;

[0099] Setting a first ratio of volume compression after exceeding the volume threshold according to the volume threshold;

[0100] The first ratio is obtained by the following formula:

[0101] ;

[0102] Among them, R 1 Represents the first ratio; R 0 Indicates the preset initial ratio; Y g Indicates the volume threshold; Y 0 Indicates the preset initial volume threshold reference value; L min and L max Indicates the minimum and maximum volume levels of the audio signal; F p Indicates the average frequency corresponding to the audio signal; F max Indicates the maximum frequency value reached by the audio signal;

[0103] Comparing the audio signal with a preset volume threshold, and intercepting the audio signal whose volume intensity exceeds the preset volume threshold as the observed audio signal;

[0104] Whether to perform audio signal compression according to a first ratio is determined according to the ratio of the observed audio signal.

[0105] The technical effect of the above technical solution is: by extracting the volume intensity of each frame in the audio signal and comparing it with the expected output volume intensity, the solution can identify the volume fluctuation in the audio. Dynamic compression can reduce the difference between the volume peak and the average value, so that the volume of the entire audio is more balanced, avoiding some parts being too loud and other parts being too quiet. The preset segmentation point intensity is used to set the volume threshold, which allows the solution to retain important audio details during the compression process. By setting a reasonable threshold, it can be ensured that these important parts are not over-compressed, thereby maintaining the naturalness and dynamic range of the audio. The first ratio is calculated by a complex formula, which is the core parameter in the dynamic compression process. This formula takes into account multiple factors, including the volume threshold, the initial volume threshold reference amount, the volume intensity range of the audio signal, and the frequency characteristics of the audio signal (average frequency and maximum frequency). This adaptive method makes the compression process more flexible and precise, and can be adjusted according to the actual situation of the audio content. By intercepting the audio signal whose volume intensity exceeds the preset threshold as the observation object, and deciding whether to compress based on the ratio of the observed audio signal, the scheme can selectively apply compression to avoid unnecessary processing. This processing helps reduce the risk of audio distortion and sound quality degradation, and improves the clarity and audibility of the final audio. For the end user, the audio signal after dynamic compression is more uniform in volume, reducing auditory fatigue. At the same time, because important audio details and dynamic range are retained, the audio content is more vivid and engaging.

[0106] In summary, this technical solution achieves the optimization of volume balance, detail preservation and protection, application of adaptive compression ratio, improvement of audio quality and enhancement of user experience by dynamically compressing audio signals. These technical effects together improve the overall quality and audibility of audio signals.

[0107] Specifically, the volume threshold is set according to the original intensity of the audio signal and the preset segmentation point intensity combined with the expected output volume intensity, specifically including:

[0108] Extracting the original intensity of the audio signal and the expected output volume intensity of the audio signal;

[0109] Extracting the audio intensity corresponding to the preset segmentation points; wherein the audio intensity corresponding to the segmentation points are respectively the first segmentation point intensity and the second segmentation point intensity, and the first segmentation point intensity is lower than the second segmentation point intensity;

[0110] Obtaining a volume threshold by using the original intensity of the audio signal and the expected output volume intensity of the audio signal in combination with the intensity of the first segmentation point and the intensity of the second segmentation point;

[0111] The volume threshold is obtained by the following formula:

[0112] ;

[0113] Among them, Y g Indicates the volume threshold; Y 0 Indicates the preset initial volume threshold reference value; L in and L out Respectively represent the original intensity of the audio signal and the expected output volume intensity of the audio signal; L 01 and L 02 Represent the first segment point intensity and the second segment point intensity respectively.

[0114] The technical effect of the above technical solution is: by extracting the original intensity and expected output volume intensity of the audio signal, and combining the preset segmentation point intensity (first segmentation point intensity and second segmentation point intensity), the solution can more finely control the setting of the volume threshold. This refined control helps to more accurately adjust the volume of the audio signal in the subsequent audio processing process to meet specific auditory needs. The setting of the volume threshold has an important impact on the dynamic range of the audio signal. By comprehensively considering the original intensity of the audio signal, the expected output volume intensity and the segmentation point intensity, the solution can optimize the dynamic range of the audio signal, so that the audio signal is smoother and more balanced while maintaining the original details. Reasonable volume threshold setting helps to reduce the distortion and noise of the audio signal. Through accurate volume threshold calculation, the solution can avoid the degradation of sound quality caused by excessive processing when compressing or limiting the audio signal. Therefore, the solution helps to improve the overall quality of the audio signal. By presetting the segmentation point intensity and using the formula to calculate the volume threshold, the solution can automatically complete the volume threshold setting process. This greatly improves the efficiency of audio processing and reduces the time and cost of manual adjustment. The volume threshold calculation formula in the solution has certain flexibility and scalability. By adjusting the parameters in the formula (such as the preset initial volume threshold reference, segmentation point strength, etc.), it can adapt to different types of audio signals and processing requirements. This makes the solution more widely applicable in practical applications. In this technical solution, the volume threshold calculation formula plays a key role. It comprehensively calculates the appropriate volume threshold based on multiple factors such as the original strength of the audio signal, the expected output volume strength, and the segmentation point strength. This threshold will serve as an important reference in the subsequent audio processing process to control the volume of the audio signal and adjust its dynamic range.

[0115] In summary, the above technical solution sets the volume threshold by comprehensively considering multiple factors of the audio signal, thereby achieving refined volume control, dynamic range optimization, audio quality improvement, processing efficiency improvement, and enhanced flexibility and scalability. These technical effects together improve the processing quality and efficiency of audio signals.

[0116] Specifically, determining whether to perform audio signal compression according to a first ratio according to the ratio of the observed audio signal specifically includes:

[0117] extracting a ratio of the observed audio signal and comparing it with a preset ratio threshold;

[0118] When the ratio of the observed audio signal is lower than a preset ratio threshold, it is determined to compress the audio signal according to a first ratio;

[0119] When the ratio of the observed audio signal is not lower than a preset ratio threshold, adjusting the first ratio by using the audio intensity of each frame of the audio signal included in the audio signal to obtain a second ratio;

[0120] The second ratio is obtained by the following formula:

[0121] ;

[0122] Among them, R 2 Represents the second ratio; R 1 represents the first ratio; n represents the number of frame audio signals contained in the audio signal; L i represents the audio intensity of the i-th frame audio signal; Y g represents the volume threshold; P represents the ratio of the observed audio signal; P y Indicates the preset ratio threshold; L b Indicates the standard deviation of the audio intensity of n frames of audio signal; Y 0 Indicates the preset initial volume threshold reference amount.

[0123] The technical effect of the above technical solution is: the solution determines whether to use the first ratio for compression by comparing the ratio of the observed audio signal with the preset ratio threshold. This strategy makes the compression process more flexible and can be adjusted according to the actual situation of the audio signal. When the ratio of the observed audio signal is low, it means that the volume fluctuation of the audio signal is small, and it is appropriate to use the first ratio for compression. When the ratio is not lower than the preset threshold, the first ratio is considered to be adjusted to adapt to the larger volume fluctuation. When the ratio of the observed audio signal is not lower than the preset ratio threshold, the solution uses the audio intensity of each frame in the audio signal to adjust the first ratio to obtain the second ratio. This adjustment method takes into account the detailed characteristics of the audio signal and makes the compression process more refined. By introducing parameters such as the standard deviation of audio intensity, the solution can more accurately reflect the fluctuation of the audio signal, so as to make more reasonable compression adjustments. Through fine compression adjustment, the solution can reduce unnecessary volume fluctuations while maintaining the overall sound quality of the audio signal. This processing method helps to avoid the loss of sound quality caused by excessive compression, so that the compressed audio signal still has high audibility and clarity. The scheme realizes the automatic processing of audio signal compression through a preset ratio threshold and a formulaic adjustment method. This processing method greatly improves processing efficiency and reduces the time and cost of manual intervention. At the same time, due to the use of standardized processing procedures, the solution is also easier to implement batch processing and integrate into existing audio processing systems. The formulas and adjustment strategies in this solution have a certain degree of adaptability and can be adjusted according to different types of audio signals and processing requirements. By changing the parameters in the formula (such as the preset initial volume threshold reference amount, ratio threshold, etc.), the solution can adapt to different audio processing scenarios and compression requirements.

[0124] In summary, the above technical solution achieves effective compression and processing of audio signals through flexible compression strategies, fine compression adjustment, maintaining audio quality, improving processing efficiency, and strong adaptability. These technical effects jointly improve the processing quality and efficiency of audio signals and meet the diverse needs in practical applications.

[0125] Video encoding module, including:

[0126] The original uncompressed video signal transmitted by the rescue soldier is obtained. The video signal consists of continuous image frames, and each frame of the video image is divided into 16x16 pixel macroblocks for subsequent independent processing and encoding, which is helpful for subsequent prediction and transform coding and improves coding efficiency;

[0127] Motion estimation technology is used to identify the displacement of similar areas between adjacent frames and only encode motion information and residuals, thereby significantly reducing data redundancy;

[0128] In the same frame, the content of the current macroblock is predicted based on the surrounding pixels, and the residual is encoded again. This method is particularly suitable for static or small-change areas;

[0129] The discrete cosine transform method is used to transform the macroblock in the pixel domain into the frequency domain. The transformed coefficients concentrate on the energy distribution of the image. Most of the coefficients are close to zero, which is easier to compress. The transformed coefficients are then quantized.

[0130] The continuous values ​​are mapped to finite discrete values ​​and then encoded to convert the quantized discrete values ​​into binary codes, i.e., audio and video digital signals, further reducing the amount of data while maintaining sufficient image quality for subsequent decoding and reconstruction.

[0131] Combiner encoding module, including:

[0132] A time slot allocation module is used to allocate time slots to transmission channels according to the transmission requirements of audio and video digital signals;

[0133] The multiplexing and combining module is used to multiplex the audio and video signals in the allocated time slots after completing the signal and time slot allocation, and combine the multiplexed signals into one transmission line through a combiner;

[0134] A signal conversion module is used to convert the acquired audio and video digital signals into analog signals for transmission, and demodulate the received analog signals back into audio and video digital signals;

[0135] Network interface, used for communication between rescue soldiers and satellite communication equipment, and between satellite communication equipment and ground command center.

[0136] Multiplexing and combining modules, including:

[0137] The encoded audio and video digital signals are transmitted and multiplexed in the allocated time slots according to the set time slot allocation principle, and the multiplexed signals are combined into one transmission line for transmission through a combiner to ensure that each signal does not overlap on the time axis to avoid interference between signals. Each signal only occupies the channel for data transmission in its own time slot to ensure that multiple signals share the same channel but do not conflict with each other.

[0138] When the multiplexed signal is transmitted, the multiplexer is used to scan the input line of the multiplexed signal in turn. If there is a signal in the line, the signal is placed in the asynchronous time division multiplexing frame. If there is no signal in the line, the line is skipped.

[0139] When the asynchronous time division multiplexing frame is full of signals, at the receiving end, the multiplexed signals are separated in time sequence through a demultiplexer to restore the original audio and video signals.

[0140] The signal conversion module specifically includes:

[0141] The restored original audio and video signals are converted into analog signals using a satellite modem, and the converted analog signals are transmitted to the ground command center via satellite communication equipment;

[0142] At the ground control center, the satellite modem restores the characteristics of the analog signal, extracts the original audio and video digital signal, and converts the analog signal back into digital information based on the extracted original audio and video digital signal;

[0143] Decoding the converted audio and video digital signals, and restoring the decoded audio and video digital signals to the original digital audio and video information;

[0144] The restored digital audio and video information is played in real time by the ground command center, and is restored to audio signals through communication equipment for conversations.

[0145] The time slot allocation module specifically includes:

[0146] Detect audio and video signals to determine the transmission requirements of audio and video signals, including data transmission volume and transmission rate;

[0147] Allocate time slots for audio and video signals according to the transmission requirements of audio and video signals. When the audio and video signals need to transmit signals, time slots are allocated. If the audio and video signals do not need to transmit signals, no time slots are allocated.

[0148] The allocated time slot width is adjusted according to the amount of audio and video signal transmission. For example, audio and video signals may require more time slots when transmitting high-definition video, but may require fewer time slots when transmitting audio or low-resolution video. Since time slots are dynamically allocated, asynchronous time division multiplexing technology can more effectively utilize channel resources and avoid the waste of resources caused by fixed time slot allocation. This flexibility enables the system to respond to the transmission needs of different users and data types, improve overall communication efficiency, and adjust the time slot allocation strategy according to the priority and real-time requirements of audio and video signals. For example, for audio and video signals with higher real-time requirements, time slots are allocated first to ensure timely transmission of data;

[0149] When the transmission of audio and video signals is completed, the time slot occupied by the audio and video signals is released, and the released time slot continues to be occupied by other audio and video signals that need to be transmitted, realizing the redistribution of channel resources. The time slice allocation principle in asynchronous time division multiplexing technology is to dynamically and flexibly allocate based on the actual needs of the signal to improve the utilization rate of channel resources and communication efficiency.

[0150] Working principle: When using the rescue single-soldier combined coding system based on audio, video and satellite communication of the present invention, according to Figure 1 and Figure 2 , including the following steps:

[0151] Step 1: Use the camera equipment carried by the rescue soldier to collect multi-channel dynamic video, and collect audio signals through communication equipment;

[0152] Step 2: Process and encode the audio and video signals collected by the rescue soldier, and convert the audio and video signals into audio and video digital signals;

[0153] Step 3: Transmit and multiplex the encoded audio and video digital signals in the allocated time slots to generate a multiplexed signal, and combine the multiplexed signal into a transmission line for transmission through a combiner. At the receiving end, the multiplexed signal is separated in time sequence through a demultiplexer to restore the original audio and video signal;

[0154] Step 4: Use a satellite modem to convert the restored original audio and video signals into analog signals, and transmit the converted analog signals to the ground command center via satellite communication equipment;

[0155] Step 5: At the ground control center, the satellite modem restores the characteristics of the analog signal, extracts the original audio and video digital signal, and converts the analog signal back into digital information based on the extracted original audio and video digital signal;

[0156] Step 6: Decode the converted audio and video digital signals, and restore the decoded audio and video digital signals to the original digital audio and video information;

[0157] Step 7: The restored digital audio and video information is played in real time by the ground command center, and is restored to audio signals through communication equipment for conversation.

[0158] In summary, the rescue soldier of the present invention can make audio and video calls and data transmission at the same time, thereby improving the rescue efficiency. The audio processor and the video processor process the audio and video signals respectively, thereby reducing the delay and distortion in the audio and video transmission process. The satellite modem converts the processing results of the rescue soldier into signals receivable by the satellite, thereby expanding the communication range. The satellite communication equipment quickly transmits the information to the ground command center, thereby providing strong support for the rescue operation. It combines the audio and video communication technology with the satellite communication technology, and has the characteristics of high efficiency, stability and reliability. The processing of the signal by the audio processor and the video processor ensures the accuracy and real-time nature of the rescue information. The satellite modem builds a bridge for the communication between the rescue soldier and the ground command center, thereby enabling the rescue information to be transmitted quickly. The combined coding technology can integrate the audio and video coding and decoding technology, thereby enabling the rescue soldier to transmit in real time through the satellite communication equipment. The transmission of video, voice and data can provide a richer and more intuitive means of communication for the rescue site, helping the command center to understand the on-site situation more accurately and make timely and effective decisions. Satellite communication technology itself has the advantages of stable communication and strong anti-interference ability. In the rescue single-soldier combined coding, audio and video transmission through satellite communication equipment can overcome the limitations of the ground communication network and ensure smooth communication under harsh conditions such as earthquakes, floods and other natural disasters. Satellite communication has a wide coverage range and can achieve global coverage, so that rescue personnel can keep in touch with the command center no matter where they are, greatly expanding the coverage of the rescue operation, so that the audio and video information of the rescue site can be transmitted to the command center in real time. The ground command center can remotely control and monitor the rescue site based on this information, which can help the command center to more accurately guide the rescue operation and improve the rescue efficiency.

[0159] It should be noted that, in this article, relational terms such as first and second, etc. are only used to distinguish one entity or operation from another entity or operation, and do not necessarily require or imply any such actual relationship or order between these entities or operations. Moreover, the terms "include", "comprise" or any other variants thereof are intended to cover non-exclusive inclusion, so that a process, method, article or device including a series of elements includes not only those elements, but also other elements not explicitly listed, or also includes elements inherent to such process, method, article or device.

[0160] While the embodiments of the present invention have been shown and described, it will be appreciated by those skilled in the art that many changes, modifications, substitutions and variations can be made to the embodiments without departing from the principles and spirit of the invention.

Claims

1. A single-soldier rescue combined coding system based on audio, video and satellite communications, characterized in that: include: The information acquisition module is used to collect multi-channel dynamic video using the camera equipment carried by the rescue soldier, and collect audio signals through the communication equipment; A signal encoding module is used to encode the audio and video signals and convert them into audio and video digital signals; The combining and encoding module is used to combine the converted audio and video digital signals using asynchronous time division multiplexing technology to restore them to the original digital audio and video information, specifically including: Divide the transmission channel time into multiple time slots, each time slot corresponds to a signal transmission, and set the time slot allocation principle; The encoded audio and video digital signals are transmitted and multiplexed in the allocated time slots according to the set time slot allocation principle to generate multiplexed signals, and the multiplexed signals are combined into one transmission line for transmission through a combiner; At the receiving end, the multiplexed signal is separated in time sequence by a demultiplexer to restore the original audio and video signal; The restored original audio and video signals are converted into analog signals using a satellite modem, and the converted analog signals are transmitted to the ground command center via satellite communication equipment; At the ground command center, the satellite modem converts the received analog signal back into digital audio and video signals, decodes the converted digital audio and video signals, and restores the decoded digital audio and video signals to the original digital audio and video information.

2. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 1 is characterized by: The signal encoding module comprises: The audio encoding module is used to process the audio signal obtained by the rescue soldier using the audio processor and encode the processed audio signal, specifically including: The audio processor is used to amplify, filter and reduce noise of the acquired audio signal, thereby filtering out background noise in the audio signal; Perform dynamic compression and limiting processing on the audio signal to reduce the dynamic range of the audio signal, increase the density of the sound, and make the peak amplitude of the audio signal uniform and consistent; Encoding the processed audio signal, and converting the processed audio signal into an audio digital signal by using a pulse code modulation method; The video encoding module is used to process the video signal obtained by the rescue soldier using a video processor and encode the processed video signal.

3. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 2 is characterized by: Dynamically compress the audio signal, including: Extract the volume intensity corresponding to each frame of the audio signal in the audio signal; Extract the expected output volume level; Setting a volume threshold according to the original intensity of the audio signal and the preset segmentation point intensity combined with the expected output volume intensity; Setting a first ratio of volume compression after exceeding the volume threshold according to the volume threshold; The first ratio is obtained by the following formula: ； Wherein, R1 represents the first ratio; R0 represents the preset initial ratio; Y g represents the volume threshold; Y0 represents the preset initial volume threshold reference value; L min and L max Indicates the minimum and maximum volume levels of the audio signal; F p Indicates the average frequency corresponding to the audio signal; F max Indicates the maximum frequency value reached by the audio signal; Comparing the audio signal with a preset volume threshold, and intercepting the audio signal whose volume intensity exceeds the preset volume threshold as the observed audio signal; Whether to perform audio signal compression according to a first ratio is determined according to the ratio of the observed audio signal.

4. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 3 is characterized by: The volume threshold is set according to the original intensity of the audio signal and the preset segmentation point intensity combined with the expected output volume intensity, specifically including: Extracting the original intensity of the audio signal and the expected output volume intensity of the audio signal; Extracting the audio intensity corresponding to the preset segmentation points; wherein the audio intensity corresponding to the segmentation points are respectively the first segmentation point intensity and the second segmentation point intensity, and the first segmentation point intensity is lower than the second segmentation point intensity; Obtaining a volume threshold by using the original intensity of the audio signal and the expected output volume intensity of the audio signal in combination with the intensity of the first segmentation point and the intensity of the second segmentation point; The volume threshold is obtained by the following formula: ； Among them, Y g represents the volume threshold; Y0 represents the preset initial volume threshold reference value; L in and L out Respectively represent the original intensity of the audio signal and the expected output volume intensity of the audio signal; L 01 and L 02 Represent the first segment point intensity and the second segment point intensity respectively.

5. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 4 is characterized by: Determining whether to perform audio signal compression according to a first ratio according to the ratio of the observed audio signal specifically includes: extracting a ratio of the observed audio signal and comparing it with a preset ratio threshold; When the ratio of the observed audio signal is lower than a preset ratio threshold, it is determined to compress the audio signal according to a first ratio; When the ratio of the observed audio signal is not lower than a preset ratio threshold, adjusting the first ratio by using the audio intensity of each frame of the audio signal included in the audio signal to obtain a second ratio; The second ratio is obtained by the following formula: ； Wherein, R2 represents the second ratio; R1 represents the first ratio; n represents the number of frame audio signals contained in the audio signal; L i represents the audio intensity of the i-th frame audio signal; Y g represents the volume threshold; P represents the ratio of the observed audio signal; P y Indicates the preset ratio threshold; L b represents the audio intensity standard deviation of n frames of audio signal; Y0 represents the preset initial volume threshold reference value.

6. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 2 is characterized by: The video encoding module specifically includes: The original uncompressed video signal transmitted by the rescue soldier is obtained. The video signal consists of continuous image frames, and each frame of the video image is divided into macro blocks of 16x16 pixels; Motion estimation technology is used to identify the displacement of similar regions between adjacent frames and only encode the motion information and residuals; In the same frame, the content of the current macroblock is predicted based on the surrounding pixels, and the residual is encoded again. This method is particularly suitable for static or small-change areas; The macroblock in pixel domain is converted to frequency domain by using discrete cosine transform method, and the transformed coefficients are quantized; The continuous values ​​are mapped to finite discrete values ​​and then encoded to convert the quantized discrete values ​​into binary codes.

7. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 1 is characterized by: The combining encoding module comprises: A time slot allocation module is used to allocate time slots to transmission channels according to the transmission requirements of audio and video digital signals; The multiplexing and combining module is used to multiplex the audio and video signals in the allocated time slots after completing the signal and time slot allocation, generate a multiplexed signal, and combine the multiplexed signal into one transmission line through a combiner; A signal conversion module is used to convert the acquired audio and video digital signals into analog signals for transmission, and demodulate the received analog signals back into audio and video digital signals; Network interface, used for communication between rescue soldiers and satellite communication equipment, and between satellite communication equipment and ground command center.

8. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 7 is characterized by: The multiplexing and combining module comprises: The encoded audio and video digital signals are transmitted and multiplexed in the allocated time slots according to the set time slot allocation principle to generate multiplexed signals, and the multiplexed signals are combined into one transmission line for transmission through a combiner; When the multiplexed signal is transmitted, the multiplexer is used to scan the input line of the multiplexed signal in turn. If there is a signal in the line, the signal is placed in the asynchronous time division multiplexing frame. If there is no signal in the line, the line is skipped. When the asynchronous time division multiplexing frame is full of signals, the multiplexed signals are separated in time sequence by a demultiplexer at the receiving end to restore the original audio and video signals.

9. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 7 is characterized by: The signal conversion module specifically includes: The restored original audio and video signals are converted into analog signals using a satellite modem, and the converted analog signals are transmitted to the ground command center via satellite communication equipment; At the ground control center, the satellite modem restores the characteristics of the analog signal, extracts the original audio and video digital signal, and converts the analog signal back into digital information based on the extracted original audio and video digital signal; Decoding the converted audio and video digital signals, and restoring the decoded audio and video digital signals to the original digital audio and video information; The restored digital audio and video information is played in real time by the ground command center, and is restored to audio signals through communication equipment for conversations.

10. The rescue single-soldier combined coding system based on audio, video and satellite communications according to claim 7, characterized in that: The time slot allocation module specifically includes: Detect audio and video signals to determine the transmission requirements of audio and video signals, including data transmission volume and transmission rate; Allocate time slots for audio and video signals according to the transmission requirements of audio and video signals. When the audio and video signals need to transmit signals, time slots are allocated. If the audio and video signals do not need to transmit signals, no time slots are allocated. Adjust the allocated time slot width according to the amount of audio and video signal transmission, and adjust the time slot allocation strategy according to the priority and real-time requirements of the audio and video signals; When the transmission of the audio and video signals is completed, the time slots occupied by the audio and video signals are released, and the released time slots continue to be occupied by other audio and video signals that need to be transmitted.

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