Audio-video signal communication management method and system

Abstract

The application provides an audio and video signal communication management method and system, and belongs to the computer field, and solves the problem that a conference system cannot intelligently perform dynamic adjustment and control on audio and video input and output devices according to specific conditions of a conference, and usually relies on manual operation. The method comprises the following steps: allocating communication addresses to audio and video input devices and audio and video output devices in a local area network, so that audio and video signals from multiple audio and video input devices are routed to designated multiple audio and video output devices in a matrix manner, the audio and video input devices comprising multiple microphones fixedly distributed in a conference area; in the case of sound collection and amplification by using a target microphone, keeping the activation state of multiple other microphones distributed in the conference area and outside the target microphone, so that the other microphones receive audio data around themselves; and controlling the amplification volume of the target microphone based on the sound pressure level of the audio data received by the other microphones distributed at different positions in the conference area.

Description

Technical Field

[0001] The present application relates to the field of computer technology, and in particular to a method and system for managing audio and video signal communication. Background Art

[0002] IP matrix systems are widely used in various scenarios requiring efficient audio and video processing and control, as well as intelligent environmental management, particularly in modern conference rooms, multi-purpose halls, classrooms, lecture halls, and command centers. In conference settings, the system can easily handle a variety of meeting needs, such as video conferencing, remote collaboration, and product demonstrations. Users can quickly configure audio and video signal routing to ensure clear presentation and communication of meeting content. However, existing conference systems lack the ability to intelligently and dynamically adjust and control audio and video input and output devices based on the specific circumstances of the meeting, often relying on manual intervention. Summary of the Invention

[0003] The embodiments of the present application provide an audio and video signal communication management method and system, which can solve the problem in related technologies that the conference system cannot intelligently perform dynamic adjustment and control of audio and video input and output devices according to the specific circumstances of the meeting and usually relies on manual labor.

[0004] A first aspect of an embodiment of the present application provides a method for managing audio and video signal communication, including:

[0005] Assigning communication addresses to audio and video input devices and audio and video output devices in the local area network so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area.

[0006] When the target microphone is used for sound collection and amplification, maintaining activation of multiple microphones other than the target microphone distributed in the conference area so that the other microphones receive audio data around them;

[0007] The amplified volume of the target microphone is controlled based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area, so as to increase the amplified volume of the target microphone when the sound pressure level is too low, and reduce the amplified volume of the target microphone when the sound pressure level is too high.

[0008] Optionally, it also includes:

[0009] When the target audio and video output device is playing audio and video, keeping microphones distributed at different locations in the conference area activated so that the microphones receive audio data around them;

[0010] The audio and video playback volume of the target audio and video output device is controlled based on the sound pressure level of the audio data received by the microphones distributed at different locations in the conference area, so as to increase the audio and video playback volume of the target audio and video output device when the sound pressure level is too low, and reduce the audio and video playback volume of the target audio and video output device when the sound pressure level is too high.

[0011] Optionally, it also includes:

[0012] parsing the audio data, and extracting target audio data that matches the speech audio obtained by the sound recording from the audio data;

[0013] The amplification volume of the target microphone is controlled based on the sound pressure level of the target audio data received by the other microphones distributed at different positions in the conference area.

[0014] Optionally, it also includes:

[0015] parsing the audio data, and extracting target audio data that matches the played audio and video data from the audio data;

[0016] The audio and video playback volume of the target audio and video output device is controlled based on the sound pressure level of the target audio data received by the microphones distributed at different positions in the conference area.

[0017] Optionally, controlling the amplification volume of the target microphone based on the sound pressure levels of the audio data received by the other microphones distributed at different locations in the conference area includes:

[0018] If the sound pressure level does not meet the preset sound pressure range, adjust the amplification volume of the target microphone until the sound pressure levels of the audio data received by the other microphones distributed at different locations in the conference area meet the preset sound pressure range; or

[0019] Check the placement of other microphones in the conference area to determine the closest microphone to the sound reinforcement equipment and the farthest microphone farthest away;

[0020] The amplification volume of the target microphone is adjusted until the sound pressure levels of the audio data received by the nearest microphone and the farthest microphone both meet the preset sound pressure range.

[0021] Optionally, also include:

[0022] Allocating temporary addresses to multiple user smart mobile terminals that join the local area network, establishing communication links with the multiple user smart mobile terminals based on the temporary addresses, and respectively obtaining location information of the multiple user smart mobile terminals based on the communication links;

[0023] Determine the actual distribution area of ​​users in the conference area based on the acquired location information of multiple user smart mobile terminals;

[0024] If it is determined that the actual distribution area of ​​users in the conference area is larger than the distribution area of ​​the microphone, then when the target microphone is used for sound collection and amplification, if it is determined based on the acquired location information of the multiple user smart mobile terminals that the actual distribution area of ​​users in the conference area is larger than the distribution area of ​​the microphone, when the target microphone is used for sound collection and amplification, the audio collection state of the multiple user smart mobile terminals is maintained so that the multiple user smart mobile terminals receive the audio data around themselves;

[0025] The amplification volume of the target microphone is controlled based on the sound pressure levels of the audio data collected by the multiple user smart mobile terminals distributed at different locations in the conference area.

[0026] Optionally, also include:

[0027] respectively acquiring user information of the plurality of user smart mobile terminals based on the communication link;

[0028] In the case where the microphone position does not match the position of the selected user to speak, determining the target user smart mobile terminal based on the user information of the selected user to speak;

[0029] Communication addresses are allocated to the audio and video input device user smart mobile terminal and audio and video output device in the local area network, and a matrix signal routing network is constructed to route the audio and video signals of the target user smart mobile terminal to the designated audio and video output device in a matrix manner.

[0030] A second aspect of the embodiments of the present application provides an audio and video signal communication management system, including:

[0031] A matrix unit is used to assign communication addresses to audio and video input devices and audio and video output devices in the local area network, so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area.

[0032] a monitoring unit configured to maintain activation of a plurality of other microphones distributed in the conference area other than the target microphone when the target microphone is used for sound reception and amplification, so that the other microphones receive audio data around them;

[0033] A control unit is used to control the amplification volume of the target microphone based on the sound pressure level of the audio data received by the other microphones distributed at different positions in the conference area, so as to increase the amplification volume of the target microphone when the sound pressure level is too low, and reduce the amplification volume of the target microphone when the sound pressure level is too high.

[0034] The third aspect of the embodiments of the present application provides an electronic device, comprising a memory and a processor, wherein the processor is configured to execute the computer program stored in the memory to implement the steps of the audio and video signal communication management method.

[0035] The fourth aspect of the embodiments of the present application provides a computer readable storage medium, which stores a computer program, wherein the computer program is executed by a processor to implement the steps of the audio and video signal communication management method.

[0036] In summary, the audio and video signal communication management method provided by the embodiments of the present application allocates communication addresses to audio and video input devices and audio and video output devices in a local area network, so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in a conference area. In the case of using a target microphone for sound collection and sound amplification, the activation states of multiple other microphones distributed in the conference area except the target microphone are maintained, so that the other microphones receive audio data around themselves. The sound amplification volume of the target microphone is controlled based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area. The conventional microphone is only used for single sound amplification function in the conference, and the scheme extends the collection function of the other non-target microphones to an environmental sound pressure monitoring and feedback mechanism through the cooperative work of multiple microphones. Both the waste of idle equipment is avoided and the maximization of equipment value is realized. The system can intelligently adjust the volume according to real-time sound pressure level data during the speech process, and does not rely on manual intervention. This intelligent control based on existing equipment improves the technical level of the conference sound system. The scheme performs matrix management on the existing audio input / output devices in the conference room, does not affect the functions and use habits of the existing system, can be deployed through software configuration, and avoids the complexity of large-scale hardware modification of the conference environment. The scheme does not require additional hardware, but fully utilizes the fixedly distributed microphones in the existing conference room, realizes sound pressure monitoring and dynamic sound amplification adjustment through software logic and signal processing, and completes the intelligent upgrading of the system through the introduction of advanced sound pressure monitoring algorithms and matrix signal management technologies. It has significant economic efficiency, and has the advantages of low investment, quick effect and strong compatibility.

[0037] Correspondingly, the system, the electronic device and the computer readable storage medium provided by the embodiments of the present application also have the above technical effects. BRIEF DESCRIPTION OF DRAWINGS

[0038] Figure 1 A possible audio and video signal communication management method provided by the embodiments of the present application is shown in the flowchart;

[0039] Figure 2A schematic structural block diagram of a possible audio and video signal communication management system provided in an embodiment of the present application;

[0040] Figure 3 A schematic diagram of the hardware structure of a possible audio and video signal communication management system provided in an embodiment of the present application;

[0041] Figure 4 A schematic structural block diagram of a possible electronic device provided in an embodiment of the present application;

[0042] Figure 5 A schematic structural block diagram of a possible computer-readable storage medium provided in an embodiment of the present application. DETAILED DESCRIPTION

[0043] The embodiments of the present application provide an audio and video signal communication management method and related equipment, which can solve the problem in related technologies that the conference system cannot intelligently perform dynamic adjustment and control of audio and video input and output devices according to the specific circumstances of the meeting, and usually relies on manual labor.

[0044] The terms "first", "second", "third", "fourth", etc. (if any) in the specification and claims of this application and the above-mentioned drawings are used to distinguish similar objects and are not necessarily used to describe a specific order or sequence. It should be understood that the data used in this way can be interchangeable where appropriate, so that the embodiments described herein can be implemented in a sequence other than that illustrated or described herein. In addition, the terms "including" and "having" and any variations thereof are intended to cover non-exclusive inclusions. For example, a process, method, system, product or device that includes a series of steps or units is not necessarily limited to those steps or units that are clearly listed, but may include other steps or units that are not clearly listed or that are inherent to these processes, methods, products or devices. The technical solutions in the embodiments of the present application will be clearly and completely described below in conjunction with the drawings in the embodiments of the present application. Obviously, the embodiments described are only part of the embodiments of the present application, not all of the embodiments.

[0045] See also Figure 1 , which is a flowchart of an audio and video signal communication management method provided in an embodiment of the present application, may specifically include: S110-S130.

[0046] S110, assigning communication addresses to audio and video input devices and audio and video output devices in the local area network, so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner, wherein the audio and video input devices include multiple microphones fixedly distributed in the conference area;

[0047] S120, while the target microphone is used for sound reception and amplification, maintaining activation of multiple other microphones distributed in the conference area other than the target microphone, so that the other microphones receive surrounding audio data;

[0048] S130: Control the amplification volume of the target microphone based on the sound pressure levels of the audio data received by the other microphones distributed at different locations in the conference area.

[0049] As you can understand, the core of this audio and video signal communication management method lies in utilizing multiple microphones distributed throughout the conference area to collect sound in real time. Based on changes in sound pressure level (i.e., the physical intensity of sound), the amplification volume of the target microphone is dynamically adjusted. This solution automatically adjusts speaking volume, eliminating manual intervention and improving the conference experience and the intelligent level of audio and video signal transmission.

[0050] For example, by assigning communication addresses to audio and video input / output devices in a local area network, a matrix-style signal routing network is constructed to enable dynamic switching of multiple signal inputs and outputs. This ensures that each input signal can be transmitted to the corresponding output device as needed, forming a flexible audio and video signal distribution system. Multiple fixed microphones are deployed in the conference area, not only for sound amplification but also for collecting real-time sound pressure level data in the environment. Non-target microphones (those not selected for speaking) sense changes in sound pressure in the environment and provide a basis for volume adjustment for the target microphone. The amplification volume of the target microphone is dynamically adjusted based on the sound signal (especially the sound pressure level) captured by the non-target microphone. If the sound pressure level is too low, the amplification volume is increased; if the sound pressure level is too high, the amplification volume is reduced. By having multiple microphones work together, the delays or errors that may occur when adjusting a single microphone are reduced, ensuring uniform sound coverage. Through automated adjustment, the inconvenience or errors caused by manual volume adjustment are avoided.

[0051] For example, multiple microphones can be evenly distributed throughout the conference area to capture sound signals from different areas. High-sensitivity microphones are prioritized to accurately capture subtle changes in sound pressure. Speakers are distributed throughout key locations to ensure sound coverage. An audio signal processor and matrix routing device can be included for audio signal routing and dynamic adjustment. Communication modules can also be used to enable high-speed data transmission and synchronization between devices.

[0052] For example, the system assigns a unique communication address to each input / output device via the network and constructs a matrix routing table. The target microphone captures the speaker's voice, while other microphones simultaneously collect surrounding sound pressure data. Based on the matrix routing table, the target microphone's audio signal is transmitted to the designated speaker. The sound pressure level captured by non-target microphones is analyzed in real time, calculating the difference between the target microphone's and the target microphone's. The target microphone's amplification volume is dynamically adjusted based on this difference. For example, if the difference is too large, the amplification volume is increased; if the difference is too small, the amplification volume is decreased. The system continuously monitors the effectiveness of these adjustments and optimizes the amplitude and frequency of these adjustments using intelligent algorithms. For example, consider a speech scenario in a large conference room with 10 microphones distributed throughout the room, with the speaker located in the center. During a speech, the target microphone captures the speaker's voice, while the other nine microphones monitor the sound distribution in different areas. If the target microphone's sound pressure level is low, while the other microphones' levels differ significantly, the system automatically increases the target microphone's amplification volume to ensure that the sound reaches every area.

[0053] In summary, the audio and video signal communication management method provided by the above-mentioned embodiments of the present application assigns communication addresses to audio and video input devices and audio and video output devices in a local area network, thereby routing audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in a conference area. While the target microphone is used for sound collection and amplification, multiple other microphones distributed in the conference area, other than the target microphone, remain activated so that the other microphones receive surrounding audio data. The amplification volume of the target microphone is controlled based on the sound pressure level of the audio data received by the other microphones at different locations in the conference area. Traditional microphones in conferences are only used for a single sound amplification function. However, this solution, through the coordinated operation of multiple microphones, expands the collection function of other non-target microphones into an ambient sound pressure monitoring and feedback mechanism. This avoids the waste of idle equipment and maximizes its value. During a speech, the system can intelligently adjust the volume based on real-time sound pressure level data, without relying on manual intervention. This intelligent control based on existing equipment has advanced the technical level of conference audio systems. This solution implements matrix management of the existing conference room's audio input / output devices, preserving existing system functionality and user habits. Deployment is simple through software configuration, avoiding the complexity of large-scale hardware modifications to the conference environment. Rather than requiring additional hardware, this solution fully utilizes the existing fixed microphones in the conference room, implementing sound pressure monitoring and dynamic amplification adjustments through software logic and signal processing. By incorporating advanced sound pressure monitoring algorithms and matrix signal management technology, this system upgrade is intelligent. It offers significant cost-effectiveness, low investment, rapid results, and strong compatibility.

[0054] In some examples, this also includes:

[0055] When the target audio and video output device is playing audio and video, keeping microphones distributed at different locations in the conference area activated so that the microphones receive audio data around them;

[0056] The audio and video playback volume of the target audio and video output device is controlled based on the sound pressure level of the audio data received by the microphones distributed at different locations in the conference area, so as to increase the audio and video playback volume of the target audio and video output device when the sound pressure level is too low, and reduce the audio and video playback volume of the target audio and video output device when the sound pressure level is too high.

[0057] As you can understand, when audio is playing on the audio and video output devices, all microphones in the conference room remain active, collecting ambient sound pressure data. These microphones cover different areas of the conference room, and the sound pressure level data collected reflects the current sound field uniformity and volume. Based on multi-point sound pressure level monitoring data, the system calculates the adjustment range for the target speaker playback volume: if the sound pressure level in a certain area is low, the audio and video playback volume is increased; if the sound pressure level is high, the playback volume is reduced to avoid harshness or echo. Based on real-time monitoring data, the system intelligently adjusts the audio and video playback volume, ensuring a uniform and comfortable listening experience for all participants regardless of their seating position in the conference room. In addition to dynamically adjusting the sound amplification of the target microphones, dynamic control of the speaker playback volume further improves the uniformity of the conference sound field, ensuring that sound is transmitted seamlessly throughout the conference area. Through real-time monitoring and intelligent feedback, the system eliminates the need for manual volume adjustment, significantly reducing the workload of meeting management. This solution is suitable for a variety of scenarios, including single-site meetings, remote collaboration, and video presentations, providing a consistent and high-quality audio and video experience. The system relies on existing conference microphones and speakers, eliminating the need for additional equipment investment, making it both economical and efficient. By dynamically adjusting the playback volume, it optimizes sound field coverage, eliminates volume differences between areas, and enhances meeting interactivity and participation.

[0058] For example, in a video presentation combined with discussion, during a product demonstration, speakers play audio commentary accompanying the video content, while attendees may discuss the topic in low voices. The non-target microphone monitors the sound pressure data in different areas in real time. If the discussion drowns out the audio commentary, the system automatically increases the playback volume. If the volume is too high and disrupts the discussion, the system automatically reduces it.

[0059] In some examples, this also includes:

[0060] parsing the audio data, and extracting target audio data that matches the speech audio obtained by the sound recording from the audio data;

[0061] The amplification volume of the target microphone is controlled based on the sound pressure level of the target audio data received by the other microphones distributed at different positions in the conference area.

[0062] It's understandable that, based on the existing solution, the addition of audio data parsing and matching capabilities enables the system to distinguish between ambient sound and speech audio, further improving the accuracy and reliability of dynamic sound amplification adjustments. By parsing the audio data received by the microphone and matching it with the speech audio captured by the target microphone, the system can identify the target audio data related to the speech and adjust the amplification volume of the target microphone based on the sound pressure level of that data.

[0063] It is understandable that the system parses the audio data received by the non-target microphone and extracts characteristic parameters (such as spectral characteristics, time series, sound texture, etc.). These features are matched in real time with the speech audio obtained by the target microphone, and the target audio data consistent with the speech audio is identified. Based on the sound pressure level of the target audio data received by the non-target microphone, the system calculates the sound pressure distribution at different positions and judges the uniformity of the sound field. Ignoring the interference of environmental noise and irrelevant sound sources, the amplification volume is adjusted only based on the target audio data. According to the difference in sound pressure levels of the target audio data received in different areas, the amplification volume of the target microphone is dynamically adjusted: in areas with lower sound pressure, the amplification volume is increased; in areas with higher sound pressure, the amplification volume is reduced to ensure comfortable listening.

[0064] For example, the target microphone collects the speaker's audio data and extracts its feature parameters. Other non-target microphones simultaneously collect ambient sound, analyze the audio, and extract its features. The system matches the audio features collected by the non-target microphones with the target microphone's speech audio features, filtering out ambient noise or irrelevant sound sources. Only the target audio data matching the speech audio is retained. Based on the target audio data received by the non-target microphones at different locations, the sound pressure level distribution for each area is calculated. If the target audio sound pressure level in a certain area is low, the system automatically increases the amplification volume; if the sound pressure level is too high, the system appropriately reduces the amplification volume to ensure uniform sound coverage. The system continuously monitors and adjusts the amplification effect, optimizing the volume control strategy in real time based on changes in the sound field. Thus, through feature analysis and matching, the system only controls the amplification of the speech audio, filtering out interference from ambient noise or irrelevant sound sources, and improving the accuracy of the amplification effect. Dynamically adjusting the amplification volume based on the sound pressure level of the target audio data achieves intelligent adaptation to complex sound fields. Even in noisy environments or with interference from multiple sound sources, the system can maintain a clear and balanced speech volume, enhancing the auditory experience for participants. No additional hardware is required, fully tapping into the potential of existing microphone and audio / video systems, enabling intelligent functionality expansion through software upgrades. This solution is ideal for complex meeting scenarios, such as those with multiple speakers taking turns, high background noise, and mixed audio and video playback.

[0065] In some examples, this also includes:

[0066] parsing the audio data, and extracting target audio data that matches the played audio and video data from the audio data;

[0067] The audio and video playback volume of the target audio and video output device is controlled based on the sound pressure level of the target audio data received by the microphones distributed at different positions in the conference area.

[0068] Understandably, in more complex conference scenarios, audio data may need to match not only the spoken audio but also the target audio and video data (such as the audio portion of the video being played), allowing for more precise control of the playback volume of the audio and video output devices. By parsing the audio data and extracting the target audio data that matches the played audio and video data, the system can dynamically adjust the playback volume to ensure that the sound coverage of the video content is uniform and not interfered with by other environmental factors.

[0069] As you can understand, the system analyzes the audio data received by non-target microphones, extracting characteristic information such as its spectral characteristics and time domain properties. This extracted data is then compared with the audio portion of the playing audio and video for feature matching, selecting the target audio data that aligns with the audio and video content. Based on the target audio data, the system calculates the sound pressure level distribution in different areas and assesses the uniformity of sound coverage. Other ambient sounds or speech sounds are ignored, and the sound field is optimized only for the playing audio and video. Based on the sound pressure level distribution, the system adjusts the playback volume of the target audio and video output devices in real time: if the sound pressure level is insufficient, the playback volume is increased; if the sound pressure level is excessive, the playback volume is appropriately reduced to ensure comfortable listening. The system can simultaneously process both the speech audio and the playing audio and video, analyzing and matching them separately, and dynamically adjusting their sound pressure levels to ensure clear sound sources and uniform coverage. This analysis and matching allows the system to monitor the sound pressure level and adjust the volume of the target audio playing only, avoiding interference from ambient noise or irrelevant sound sources. Dynamic adjustment of the playback volume of the target audio and video devices ensures uniform coverage of content such as video commentary and background sound effects across different areas, enhancing the attendee experience. The system can independently process speech audio and video in multi-source scenarios, enabling parallel optimization and enhancing system intelligence. It fully utilizes existing conferencing equipment without requiring additional hardware investment, optimizing complex audio scenarios through software algorithm upgrades. The system dynamically senses changes in sound pressure levels and adjusts playback volume in real time to ensure optimal sound quality.

[0070] In some examples, controlling the amplification volume of the target microphone based on the sound pressure levels of the audio data received by the other microphones distributed at different locations in the conference area includes:

[0071] If the sound pressure level does not meet the preset sound pressure range, the amplification volume of the target microphone is adjusted until the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area meet the preset sound pressure range.

[0072] It's understandable that in certain conference scenarios, the amplification volume of a target microphone can be dynamically adjusted to meet a preset sound pressure range based on the sound pressure levels received by other microphones distributed throughout the conference area. This mechanism monitors sound pressure level changes in real time and, combined with dynamic volume adjustment strategies, ensures uniform and clear sound transmission within the conference room, preventing excessive lows or highs. Based on the real-time sound pressure levels collected by multiple microphones, the system determines whether each area in the conference room meets the set sound pressure range (e.g., 50-70dB SPL). If the sound pressure level in a particular area falls below the lower limit of the preset range, the sound is insufficient, and the amplification volume of the target microphone needs to be increased. Conversely, if it exceeds the upper limit, the amplification volume needs to be reduced. The system continuously monitors the adjusted sound pressure levels through a real-time feedback mechanism until all areas meet the preset range. This closed-loop control process ensures timely and accurate adjustments. By adjusting the amplification volume of the target microphone, sound coverage in the conference area is optimized, preventing participants from hearing unclear or excessively loud sounds.

[0073] For example, the system collects audio data in real time through distributed microphones and calculates the sound pressure level at each microphone position. Compared with the preset range (such as 50-70dB SPL), it is determined whether the current sound field meets the requirements. If the sound pressure level in some areas is lower than the lower limit of the preset range, the system increases the volume of the target microphone until the sound pressure level meets the range requirements. If the sound pressure level in some areas is higher than the upper limit of the preset range, the system reduces the volume of the target microphone until the sound pressure level enters the range. While adjusting the volume, the system continuously monitors the change of the sound pressure level, and further optimizes the volume adjustment amplitude and speed according to the latest data to ensure uniform sound propagation. When the sound pressure level of all monitoring points meets the preset range, the system enters a stable state and maintains the volume unchanged. Taking the sound field of a large conference room as an example, 10 monitoring microphones are distributed in the conference room to collect sound pressure data. The speaker's volume is too small, and the sound pressure level in some areas is only 45dB SPL, which is lower than the preset range (50-70dB SPL). The system automatically increases the volume of the target microphone to make the sound pressure level monitored by the distributed microphone reach the range. Finally, the sound pressure in all areas reaches 55-65dB SPL, and the participants feel comfortable. Or in some cases, for example, during heated discussions, the speaker's volume is high, and the sound pressure level in some areas reaches 80dB SPL, exceeding the preset range (50-70dB SPL). The system reduces the volume of the target microphone to gradually reduce the sound pressure level and avoid harsh or echo-inducing sounds. After adjustment, the sound pressure level in all areas returns to the range of 65-70dB SPL, making the conference environment more comfortable. Thus, whether in a large space scenario or with multiple speakers switching, the system can adjust the volume in real time to keep the sound pressure level within the preset range. Based on the real-time feedback of the distributed microphones, the system adjusts the volume of the target microphone to achieve sound coverage optimization throughout the conference room. Through continuous sound pressure level monitoring and feedback, the system avoids the lag and uncertainty of traditional manual adjustment. The speaking sound is always clear and moderate, and participants do not need to deliberately increase or decrease the sound, and there is no situation of unclear or harsh sound. The system realizes intelligent adjustment based on existing microphones, speakers and signal processing equipment, without additional hardware investment.

[0074] In some examples, the sound pressure level of the audio data received by the other microphones distributed in different positions in the conference area controls the volume of the target microphone, comprising:

[0075] Query the arrangement position of the other microphones in the conference area, determine the nearest microphone closest to the sound amplification device and the farthest microphone farthest from the sound amplification device;

[0076] Adjust the volume of the target microphone until the sound pressure level of the audio data received by the nearest microphone and the farthest microphone meets the preset sound pressure range.

[0077] As can be understood, by querying the positions of other microphones in the conference area, the system dynamically determines the microphone closest and farthest from the target sound amplification device. Based on the sound pressure levels received by the microphones at these two extreme locations, the amplification volume of the target microphone is dynamically adjusted until their sound pressure levels both fall within the preset sound pressure range. This mechanism ensures uniform sound coverage while avoiding issues where the sound is too loud or too quiet in certain areas. Based on the fixed microphone layout in the conference room, the system queries the physical position of each microphone and records its distance from the target sound amplification device. The two microphones closest and farthest from the sound amplification device are determined as key reference points for evaluating sound coverage. The sound pressure levels received by the closest and farthest microphones are monitored in real time to determine whether they are within a preset range (e.g., 50-70dB SPL). If the sound pressure levels do not fall within the range, the amplification volume of the target microphone is adjusted. The system gradually increases or decreases the amplification volume of the target microphone until the sound pressure levels of the closest and farthest microphones both fall within the preset range, ensuring uniform and clear sound from the closest to the farthest point in the coverage area. This system dynamically adjusts the amplification volume of the target microphone to ensure uniform coverage from the closest point to the farthest point. A dual-point monitoring mechanism for the closest and farthest microphones prevents excessive loudness in close proximity and insufficient coverage at greater distances. The system features closed-loop control, monitoring and optimizing the amplification volume in real time during presentations. The solution is compatible with conference rooms of varying sizes, adapting to various meeting environments through positional calibration and dynamic adjustment. This ensures uniform sound distribution without manual intervention, enhancing attendees' auditory comfort and sense of engagement.

[0078] In some examples, this also includes:

[0079] Allocating temporary addresses to multiple user smart mobile terminals that join the local area network, establishing communication links with the multiple user smart mobile terminals based on the temporary addresses, and respectively obtaining location information of the multiple user smart mobile terminals based on the communication links;

[0080] Determine the actual distribution area of ​​users in the conference area based on the acquired location information of multiple user smart mobile terminals;

[0081] If it is determined that the actual distribution area of ​​users in the conference area is larger than the distribution area of ​​the microphones, then when using the target microphones for sound collection and amplification, the audio collection state of the multiple user smart mobile terminals is maintained so that the multiple user smart mobile terminals receive the audio data around them;

[0082] The amplification volume of the target microphone is controlled based on the sound pressure levels of the audio data collected by the multiple user smart mobile terminals distributed at different locations in the conference area.

[0083] It can be understood that by assigning temporary addresses to multiple user intelligent mobile terminals (such as mobile phones, tablets, etc.) joining the local area network and establishing communication links, the location information and audio acquisition data of the user terminals are dynamically acquired. Especially in the case where the user distribution area exceeds the coverage range of the fixed microphone, the intelligent mobile terminal is used as an auxiliary audio acquisition device to further optimize the conference sound coverage effect and the volume control of the target microphone.

[0084] It can be understood that the system dynamically allocates temporary addresses to multiple intelligent mobile terminals within the local area network, establishes communication links with the terminals through these addresses, and realizes real-time acquisition of location and audio data. The system acquires the location information of the intelligent mobile terminal through the communication link and determines the actual distribution range of the user. When the actual distribution range of the user is greater than the coverage area of the fixed microphone, the audio acquisition function of the intelligent mobile terminal is activated as a supplementary sound source. The audio data collected by the intelligent mobile terminal supplements the monitoring range of the original fixed microphone, especially the sound information at the edge or blind area of the conference area. The system dynamically adjusts the volume of the target microphone according to the sound pressure level of the audio data collected by the intelligent mobile terminal to ensure the uniformity and clarity of the sound coverage range. Based on the comprehensive analysis of the sound field distribution based on the data collected by the fixed microphone and the intelligent mobile terminal, the volume of the target microphone is dynamically adjusted to optimize the conference sound effect.

[0085] For example, a user's smart mobile terminal connects to the system via a local area network. The system assigns a temporary address to each terminal and establishes a communication link. The system uses this communication link to obtain the location information of the user's smart mobile terminal and create a user distribution map. If the user distribution range exceeds the coverage area of ​​the fixed microphone, the system activates the smart terminal's audio collection function. The smart terminal collects ambient audio data, which, combined with the data collected by the fixed microphone, forms a complete sound field distribution map. The system determines the uniformity of sound coverage based on the sound pressure level differences at different locations. If the sound pressure level in a certain area is insufficient, the amplification volume of the target microphone is increased; if the sound pressure level is too high, the volume is reduced. Combining data from the fixed microphone and smart terminal ensures uniform and clear sound coverage throughout the conference area. The system dynamically adjusts the amplification volume based on real-time user location and sound pressure level data to optimize sound quality. This allows the smart terminal to supplement the fixed microphone's monitoring blind spots, achieving more comprehensive sound field coverage. Based on the actual user distribution location and the integrated sound pressure data, the amplification volume is dynamically adjusted to ensure uniform sound distribution within the conference area. The system dynamically adapts to changes in user distribution, enabling real-time optimization for scenarios beyond the coverage of fixed microphones and group discussions. This eliminates the need for additional hardware investment and fully leverages the audio acquisition capabilities of user smart terminals, reducing system deployment costs. Whether users are located in the center of the conference or in peripheral areas, clear and balanced audio is delivered, enhancing participation and user experience. For example, consider a meeting where users are located beyond the range of fixed microphones. Six fixed microphones cover the center of the conference room, but some participants are located in peripheral or remote areas. Users connect to the system through their smartphones via the local area network. The system acquires user distribution information and identifies some users outside the coverage of the fixed microphones. The system activates the smartphone's audio acquisition function, collects sound pressure level data from peripheral areas, and dynamically adjusts the amplification volume of the targeted microphones to ensure adequate coverage in these areas. By dynamically assigning temporary addresses to smart mobile terminals, establishing communication links, and acquiring location information, this solution integrates the audio acquisition capabilities of smart terminals into the conference audio management system. When user distribution exceeds the coverage range of fixed microphones, smart terminals are used to supplement audio collection functions and dynamically adjust the target microphone's amplification volume based on its sound pressure level, significantly improving the conference audio coverage and sound quality, providing a flexible and intelligent audio solution for modern conference scenarios.

[0086] In some examples, this also includes:

[0087] respectively acquiring user information of the plurality of user smart mobile terminals based on the communication link;

[0088] In the case where the microphone position does not match the position of the selected user to speak, determining the target user smart mobile terminal based on the user information of the selected user to speak;

[0089] Communication addresses are allocated to the audio and video input device user smart mobile terminal and audio and video output device in the local area network, and a matrix signal routing network is constructed to route the audio and video signals of the target user smart mobile terminal to the designated audio and video output device in a matrix manner.

[0090] As can be understood, this audio and video signal communication management method aims to address issues in traditional conference systems such as manual volume adjustment, uneven sound coverage, and mismatches between speaker and microphone positions. By utilizing fixed microphones within the conference room, attendees' smart mobile devices, and matrix-based audio and video signal routing, it achieves intelligent audio signal acquisition, dynamic adjustment, and efficient transmission. Communication addresses are assigned to audio and video input devices (fixed microphones, user smart mobile devices) and audio and video output devices (speakers, display devices) within the local area network, creating a matrix signal routing network. Through matrix routing, audio and video signals are flexibly transmitted from input devices to designated output devices. Fixed microphone acquisition: Multiple fixed microphones distributed throughout the conference area are used for sound collection, amplification, and ambient sound pressure level monitoring. Smart mobile terminal acquisition: When attendees' distribution exceeds the coverage range of fixed microphones or when the speaker's position mismatches with the microphone's, attendees' smart mobile devices (such as mobile phones or tablets) are used for audio acquisition and location information. The acquired audio data is parsed to extract target audio data that matches the speech audio or played audio and video data. Filter out ambient noise and irrelevant sound sources to ensure accurate audio processing. Adjust the amplification volume of the target microphone in real time based on the sound pressure level of the target audio data collected by fixed microphones and smart mobile terminals in different locations. This ensures uniform sound coverage within the conference area and prevents excessively loud or quiet sounds. If the speaker's position does not match the fixed microphone's position, the target user's smart mobile terminal is identified as the audio and video input device based on user information. Matrix routing transmits the audio and video signals from the target user's smart mobile terminal to the designated audio and video output device, enabling flexible audio capture and transmission.

[0091] Exemplarily, after the system is started, communication addresses are assigned to all fixed microphones, speakers, and user smart mobile terminals. A device matrix is ​​constructed to prepare for routing control of audio and video signals. The user smart mobile terminal accesses the local area network, and the system assigns it a temporary address. The location information and user information of the smart mobile terminal are obtained through the communication link. The fixed microphone continuously collects environmental audio data for sound reception, amplification, and sound pressure level monitoring. The smart mobile terminal activates the audio collection function when necessary (such as when the user distribution exceeds the coverage range of the fixed microphone or the speaker position does not match). The collected audio data is parsed to extract the target audio data that matches the speech audio or the played audio and video data. The sound pressure level of the target audio data received at each location (fixed microphone and smart mobile terminal) is calculated. Determine whether the sound pressure level is within the preset range and evaluate the uniformity of the sound coverage. Based on the sound pressure level data, adjust the amplification volume of the target microphone to ensure that the sound pressure level in each area meets the preset range. When playing audio and video content, adjust the playback volume of the audio and video output device to ensure uniform sound coverage. When the speaker's position doesn't match the fixed microphone's position, the system identifies the speaker's smart mobile terminal as the target audio and video input device based on user information. Matrix routing transmits the terminal's audio and video signals to the designated audio and video output device, achieving efficient audio transmission. The system continuously monitors changes in sound pressure levels and user positions, dynamically adjusting volume and signal routing strategies to ensure optimal conference audio quality.

[0092] By acquiring the location information of users' smart mobile devices, the system can understand the actual distribution of participants in real time. When participants exceed the coverage range of fixed microphones, the audio collection function of smart mobile devices can be used to fill in audio monitoring blind spots. This expands the sound coverage area, ensuring that all participants can clearly hear the meeting content. This improves the system's adaptability and meets the needs of meetings of varying sizes and layouts. If the speaker's position does not match the fixed microphone's position, the system uses user information to determine the speaker's smart mobile device as the audio and video input device. Matrix routing efficiently transmits the speaker's audio and video signals to the designated output device. This eliminates the need for speakers to deliberately move to the fixed microphone position, improving meeting flexibility and efficiency. This ensures clear speech quality regardless of location. Combining audio data from fixed microphones and smart mobile devices, the system comprehensively monitors the sound field in the meeting area. Based on the integrated sound pressure level data, it dynamically adjusts the amplification volume to ensure even sound distribution. This avoids uneven sound coverage and areas with excessively loud or quiet sound. This improves participant auditory comfort and enhances the meeting experience. Analyze audio data and extract target audio data that matches the speech audio or playback audio and video data. Filter out ambient noise and irrelevant sound sources to ensure accurate volume adjustment and signal transmission. Improve the accuracy of audio processing and avoid noise interference. Enhance the quality of the audio signal and ensure the smooth progress of the meeting. 5. Make full use of existing equipment and reduce implementation costs. Utilize the existing fixed microphones in the conference room and the participants' smart mobile terminals, eliminating the need for additional hardware equipment. Through software function expansion, achieve system intelligence and functional enhancements. Reduce system deployment and maintenance costs, with significant economic advantages. Improve equipment utilization and avoid resource waste.

[0093] For example, consider a large conference where attendees are located outside the coverage area of ​​a fixed microphone. This involves numerous participants scattered throughout the conference room, with some areas outside the fixed microphone's coverage. The attendee's smartphone connects to the conference LAN, and the system assigns a temporary address and obtains their location information. The system detects that the attendee's distribution area is larger than the fixed microphone's coverage area and activates the smartphone's audio collection function. Combining the audio data from the fixed microphone and the smartphone, the system dynamically adjusts the target microphone's amplification volume to ensure uniform sound coverage throughout the room. This ensures that all attendees can clearly hear the conference content, and the sound is evenly distributed. The system fully utilizes the attendee's smartphone device, eliminating the need for additional equipment.

[0094] According to some embodiments, when a participant is far away and the sound received is small, the voice signal is routed to the participant's own smart mobile terminal through a matrix method, and the participant can directly play the audio content through his or her smart device, thereby ensuring a clear listening experience.

[0095] For example, when the conference room is large or the participants are in the blind spot of the sound amplification equipment (such as the corner of the room, branch venue, etc.), the sound attenuation causes the sound pressure level in these areas to be low, and the participants cannot hear the speech clearly. The system can route the audio signal of the target microphone directly to the participant's smart mobile terminal, so that it can receive and play the sound through its own device. The system uses audio and video signal matrix routing technology to route the speaker's audio signal to the smart terminal of the designated participant. Each participant joins the local area network through his or her smart terminal as a receiving device to form a distributed audio coverage system. After receiving the routing signal, the participant's smart mobile terminal plays it through its own speaker or headphones. The system can dynamically adjust the terminal playback volume to ensure coordination with the overall conference sound environment. Participants can control the volume of their smart terminals to adapt to their personal preferences without affecting other participants.

[0096] For example, a participant's smart mobile terminal is connected to the local area network, and the system assigns a unique communication address to each terminal. The system uses matrix routing to establish an audio transmission path from the target microphone to the smart terminal. Using sound pressure level data collected by fixed microphones and smart mobile terminals, the system determines which areas of the meeting are experiencing difficulty hearing the audio. The system then sets the participant terminals in these areas as target receivers for the audio signal. The speech audio signal collected by the target microphone is transmitted via matrix routing to the designated participant's smart terminal. After receiving the signal, the smart terminal plays the audio through its built-in speaker or headphones. Based on real-time changes in the sound field, the system dynamically adjusts the routing path and terminal playback volume to ensure optimal conference audio quality. This allows smart mobile terminals to supplement the coverage gap of fixed microphones and speakers, resolving the issue of unclear hearing for participants at a distance. Participants can adjust the terminal volume to meet their individual listening needs without disturbing other participants. This eliminates the need for additional hardware and utilizes participants' existing smart terminals to optimize audio coverage, reducing system costs. Participants can listen directly through their terminals, enhancing their engagement and attention to the meeting content. It is suitable for various scenarios such as large-scale meetings, small-scale discussions, and multi-venue collaboration, meeting the needs of different scales.

[0097] For example, the user's smart mobile terminal can receive and play audio signals by using the terminal's own browser or common network protocols without installing a special application (App). This method avoids the burden of users installing additional applications while ensuring the ease of use and compatibility of the system. For example, participants access the URL provided by the conference system (such as a web page in the local area network) through the terminal's browser. The conference system pushes audio signals to the terminal through WebRTC technology or real-time audio streaming (such as HTTP Live Streaming, HLS). Users do not need to download any software, just open the web page to receive and play audio through the browser.

[0098] According to some embodiments, smart mobile terminals distributed in the conference area can be combined with fixed microphones to form a "virtual cluster microphone". These devices not only collect audio at a single point, but also dynamically optimize the signal quality through a cluster collaboration mode. The system uses the multi-point audio collection function of distributed terminals and audio data fusion technology to achieve environmental noise reduction and speech signal enhancement. For example, when a participant speaks, multiple terminals closest to the speaker collaboratively collect speech audio data, and the system automatically performs weighted fusion to extract the highest quality audio signal. The system monitors the distribution of participants in real time and dynamically adjusts the composition of the "virtual cluster microphone" based on their location information. When the speaker moves, the system automatically selects the latest optimal microphone cluster to ensure high quality and stability of audio collection. In this way, the limitation of single terminal signal collection is broken through, and the dynamic collaboration of multiple distributed terminals is combined to significantly improve the quality of the audio signal. It is suitable for conference environments with dynamically changing participant distribution and complex noise.

[0099] According to some embodiments, a "pre-listening synchronization" function may also be included, which provides a real-time audio preloading mechanism on the user terminal through distributed caching technology, so that the terminal can obtain synchronized audio playback effects under different network conditions. The system assigns a local audio cache node to each user, and caches audio data near the user terminal through edge computing technology. The audio data is pushed in segments according to microsecond timestamps to ensure that the terminal can achieve seamless playback even in the case of network fluctuations. Before the signal is pushed, the system loads a small amount of audio preview data for the user terminal to maintain playback synchronization when the network delay fluctuates. For example, in a remote conference scenario, participants in different locations can hear the speech at the same time without delay or freeze. As a result, playback synchronization is improved: even under complex network conditions, each participant can still get a real-time, synchronized audio experience. Enhanced reliability: The caching mechanism reduces the dependence of real-time audio transmission on network fluctuations.

[0100] The above describes the audio and video signal communication management method in the embodiment of the present application. The following describes the audio and video signal communication management system in the embodiment of the present application.

[0101] See also Figure 2 In the embodiments of the present application, an embodiment of an audio and video signal communication management system is described, which may include:

[0102] The matrix unit 201 is used to assign communication addresses to audio and video input devices and audio and video output devices in the local area network, so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area.

[0103] The monitoring unit 202 is configured to maintain activation of a plurality of other microphones distributed in the conference area other than the target microphone when the target microphone is used for sound reception and amplification, so that the other microphones receive audio data around them;

[0104] The control unit 203 is configured to control the amplification volume of the target microphone based on the sound pressure levels of the audio data received by the other microphones distributed at different locations in the conference area.

[0105] In summary, the audio and video signal communication management system provided in the above embodiments assigns communication addresses to audio and video input devices and output devices in a local area network, routing audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in a conference area. While the target microphone is used for sound collection and amplification, multiple other microphones distributed in the conference area, other than the target microphone, remain activated to allow them to receive surrounding audio data. The amplification volume of the target microphone is controlled based on the sound pressure level of the audio data received by the other microphones at different locations in the conference area. Traditional microphones in conferences are used solely for amplification. However, this solution, through the coordinated operation of multiple microphones, expands the collection function of other non-target microphones into an ambient sound pressure monitoring and feedback mechanism. This avoids the waste of idle equipment while maximizing its value. During speeches, the system can intelligently adjust the volume based on real-time sound pressure level data, without relying on manual intervention. This intelligent control based on existing equipment advances the technical level of conference audio systems. This solution implements matrix management of the existing conference room's audio input / output devices, preserving existing system functionality and user habits. Deployment is simple through software configuration, avoiding the complexity of large-scale hardware modifications to the conference environment. Rather than requiring additional hardware, this solution fully utilizes the existing fixed microphones in the conference room, implementing sound pressure monitoring and dynamic amplification adjustments through software logic and signal processing. By incorporating advanced sound pressure monitoring algorithms and matrix signal management technology, this system upgrade is intelligent. It offers significant cost-effectiveness, low investment, rapid results, and strong compatibility.

[0106] above Figure 2 The audio and video signal communication management system in the embodiment of the present application is described from the perspective of modular functional entities. The audio and video signal communication management system in the embodiment of the present application is described in detail from the perspective of hardware processing. Figure 3 An embodiment of the audio and video signal communication management system 300 in the embodiment of the present application includes:

[0107] Input device 301, output device 302, processor 303 and memory 304, wherein the number of processor 303 can be one or more, Figure 3 In some embodiments of the present application, the input device 301, the output device 302, the processor 303 and the memory 304 may be connected via a bus or other means, wherein: Figure 3 The bus connection is taken as an example.

[0108] By calling the operation instructions stored in the memory 304, the processor 303 is configured to perform the following steps:

[0109] Assigning communication addresses to audio and video input devices and audio and video output devices in the local area network so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area.

[0110] When the target microphone is used for sound collection and amplification, maintaining activation of multiple microphones other than the target microphone distributed in the conference area so that the other microphones receive audio data around them;

[0111] The amplification volume of the target microphone is controlled based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area.

[0112] By calling the operation instructions stored in the memory 304, the processor 303 is also used to execute Figure 1 Any method in the corresponding embodiment.

[0113] See also Figure 4 , Figure 4 Schematic diagram of an electronic device according to an embodiment of the present application.

[0114] like Figure 4 As shown, an embodiment of the present application provides an electronic device 400, including an electronic device memory 410, an electronic device processor 420, and a computer program 411 stored in the electronic device memory 410 and executable on the electronic device processor 420. When the electronic device processor 420 executes the computer program 411, the following steps are implemented:

[0115] Assigning communication addresses to audio and video input devices and audio and video output devices in the local area network so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area.

[0116] When the target microphone is used for sound collection and amplification, maintaining activation of multiple microphones other than the target microphone distributed in the conference area so that the other microphones receive audio data around them;

[0117] The amplification volume of the target microphone is controlled based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area.

[0118] In the specific implementation process, when the electronic device processor 420 executes the computer program 411, it can achieve Figure 1 Any implementation manner in the corresponding embodiments.

[0119] Since the electronic device introduced in this embodiment is a device used to implement an audio and video signal communication management system in the embodiment of this application, based on the method introduced in the embodiment of this application, technical personnel in this field can understand the specific implementation method of the electronic device of this embodiment and its various variations. Therefore, how the electronic device implements the method in the embodiment of this application will not be introduced in detail here. As long as the equipment used by technical personnel in this field to implement the method in the embodiment of this application falls within the scope of protection to be protected by this application.

[0120] See also Figure 5 , Figure 5 A schematic diagram of an embodiment of a computer-readable storage medium provided in an embodiment of the present application.

[0121] like Figure 5 As shown, this embodiment provides a computer-readable storage medium 500 on which a target computer program 511 is stored. When the target computer program 511 is executed by a processor, the following steps are implemented:

[0122] Assigning communication addresses to audio and video input devices and audio and video output devices in the local area network so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area.

[0123] When the target microphone is used for sound collection and amplification, maintaining activation of multiple microphones other than the target microphone distributed in the conference area so that the other microphones receive audio data around them;

[0124] The amplification volume of the target microphone is controlled based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area.

[0125] The computer program product includes one or more computer instructions. When the computer program instructions are loaded and executed on a computer, the process or function described in the embodiment of the present application is generated in whole or in part. The computer may be a general-purpose computer, a special-purpose computer, a computer network, or other programmable device. The computer instructions may be stored in a computer-readable storage medium or transmitted from one computer-readable storage medium to another computer-readable storage medium. For example, the computer instructions may be transmitted from one website, computer, server, or data center to another website, computer, server, or data center via a wired (e.g., coaxial cable, optical fiber, digital subscriber line (DSL)) or wireless (e.g., infrared, wireless, microwave, etc.) method. The computer-readable storage medium may be any available medium that a computer can store or a data storage device such as a server or data center that includes one or more available media integrated therein. The available medium may be a magnetic medium (e.g., a floppy disk, a hard disk, a tape), an optical medium (e.g., a DVD), or a semiconductor medium (e.g., a solid-state drive (SSD)).

[0126] Those skilled in the art will clearly understand that, for the convenience and brevity of description, the specific working processes of the systems, devices and units described above can refer to the corresponding processes in the aforementioned method embodiments and will not be repeated here.

[0127] In the several embodiments provided in this application, it should be understood that the disclosed systems, devices and methods can be implemented in other ways. For example, the device embodiments described above are merely schematic. For example, the division of the units is merely a logical function division. In actual implementation, there may be other division methods, such as multiple units or components can be combined or integrated into another system, or some features can be ignored or not executed. Another point is that the mutual coupling or direct coupling or communication connection shown or discussed can be an indirect coupling or communication connection through some interfaces, devices or units, which can be electrical, mechanical or other forms.

[0128] The units described as separate components may or may not be physically separate, and the components shown as units may or may not be physical units, that is, they may be located in one place or distributed across multiple network units. Some or all of these units may be selected to achieve the purpose of this embodiment according to actual needs.

[0129] In addition, each of the functional units in the various embodiments of the present application can be integrated in one processing unit, or each unit can be physically present separately, or two or more units can be integrated in one unit. The integrated unit can be realized in the form of hardware or in the form of a software functional unit.

[0130] When the integrated unit is realized in the form of a software functional unit and sold or used as an independent product, it can be stored in a computer readable storage medium. Based on such an understanding, the technical solutions of the present application, essentially or in part, or all or part of the technical solutions can be embodied in the form of a software product. The computer software product is stored in a storage medium, and includes several instructions for causing a computer device (which can be a personal computer, a server, or a network device, etc.) to execute all or part of the steps of the methods according to the various embodiments of the present application. The aforementioned storage medium includes: U disk, mobile hard disk, read-only memory (ROM), random access memory (RAM), magnetic disk or optical disk, and various other media that can store program codes.

[0131] The above-described embodiments are merely used to illustrate the technical solutions of the present application, rather than limit the technical solutions thereof; even though the present application has been described in detail with reference to the foregoing embodiments, those of ordinary skill in the art should understand that: they can still make modifications to the technical solutions recorded in the foregoing embodiments, or make equivalent replacements to some of the technical features thereof; and these modifications or replacements do not cause the essence of the corresponding technical solutions to deviate from the spirit and scope of the technical solutions of the embodiments of the present application.

Claims

1. A method for managing audio and video signal communication, characterized in that: include: Assigning communication addresses to audio and video input devices and audio and video output devices in the local area network so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area. When the target microphone is used for sound reception and amplification, multiple other microphones distributed in the conference area other than the target microphone are kept activated so that the other microphones receive audio data surrounding them, wherein multiple speakers are distributed at key locations in the conference area to ensure sound coverage, and the audio data surrounding them received by the other microphones includes the audio data emitted by the speakers; controlling the amplified volume of the target microphone based on the sound pressure levels of the audio data received by the other microphones distributed at different locations in the conference area, so as to increase the amplified volume of the target microphone when the sound pressure level is too low and decrease the amplified volume of the target microphone when the sound pressure level is too high; The controlling the amplification volume of the target microphone based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area includes: parsing the audio data received by the other microphones distributed at different locations in the conference area, and extracting target audio data that matches the speech audio obtained by the sound recording from the audio data; Based on the sound pressure levels of the target audio data received by the other microphones distributed at different positions in the conference area, the sound pressure level distribution of each area is calculated to increase the amplification volume of the target microphone when the sound pressure level of the target audio data received by the other microphones in a certain area is too low, and to reduce the amplification volume of the target microphone when the sound pressure level of the target audio data received by the other microphones in a certain area is too high.

2. The method according to claim 1, characterized in that Also includes: When the target audio and video output device is playing audio and video, keeping microphones distributed at different locations in the conference area activated so that the microphones receive audio data around them; The audio and video playback volume of the target audio and video output device is controlled based on the sound pressure level of the audio data received by the microphones distributed at different locations in the conference area, so as to increase the audio and video playback volume of the target audio and video output device when the sound pressure level is too low, and reduce the audio and video playback volume of the target audio and video output device when the sound pressure level is too high.

3. The method according to claim 2, characterized in that Also includes: parsing the audio data, and extracting target audio data that matches the played audio and video data from the audio data; The audio and video playback volume of the target audio and video output device is controlled based on the sound pressure level of the target audio data received by the microphones distributed at different positions in the conference area, so as to increase the audio and video playback volume of the target audio and video output device when the sound pressure level is too low, and reduce the audio and video playback volume of the target audio and video output device when the sound pressure level is too high.

4. The method according to claim 1, wherein The controlling the amplification volume of the target microphone based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the conference area includes: If the sound pressure level does not meet the preset sound pressure range, adjust the amplification volume of the target microphone until the sound pressure levels of the audio data received by the other microphones distributed at different locations in the conference area meet the preset sound pressure range; or Check the placement of other microphones in the conference area to determine the closest microphone to the sound reinforcement equipment and the farthest microphone farthest away; The amplification volume of the target microphone is adjusted until the sound pressure levels of the audio data received by the nearest microphone and the farthest microphone both meet the preset sound pressure range.

5. The method according to any one of claims 1 to 4, characterized in that Also includes: Allocating temporary addresses to multiple user smart mobile terminals that join the local area network, establishing communication links with the multiple user smart mobile terminals based on the temporary addresses, and respectively obtaining location information of the multiple user smart mobile terminals based on the communication links; Determine the actual distribution area of ​​users in the conference area based on the acquired location information of multiple user smart mobile terminals; If it is determined that the actual distribution area of ​​users in the conference area is larger than the distribution area of ​​the microphones, then when using the target microphones for sound collection and amplification, the audio collection state of the multiple user smart mobile terminals is maintained so that the multiple user smart mobile terminals receive the audio data around them; The amplified volume of the target microphone is controlled based on the sound pressure level of the audio data collected by the multiple user smart mobile terminals distributed at different locations in the conference area, so as to increase the amplified volume of the target microphone when the sound pressure level is too low, and reduce the amplified volume of the target microphone when the sound pressure level is too high.

6. The method according to claim 5, characterized in that Also includes: respectively acquiring user information of the plurality of user smart mobile terminals based on the communication link; In the case where the microphone position does not match the position of the selected user to speak, determining the target user smart mobile terminal based on the user information of the selected user to speak; Communication addresses are allocated to the audio and video input device user smart mobile terminal and audio and video output device in the local area network, and a matrix signal routing network is constructed to route the audio and video signals of the target user smart mobile terminal to the designated audio and video output device in a matrix manner.

7. An audio and video signal communication management system, characterized in that: According to any one of claims 1 to 6, the system comprises: A matrix unit is used to assign communication addresses to audio and video input devices and audio and video output devices in the local area network, so as to route audio and video signals from multiple audio and video input devices to designated multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in the conference area. a monitoring unit configured to maintain activation of a plurality of other microphones distributed in the conference area other than the target microphone when the target microphone is used for sound reception and amplification, so that the other microphones receive audio data around them; A control unit is used to control the amplification volume of the target microphone based on the sound pressure level of the audio data received by the other microphones distributed at different positions in the conference area, so as to increase the amplification volume of the target microphone when the sound pressure level is too low, and reduce the amplification volume of the target microphone when the sound pressure level is too high.

8. An electronic device, characterized in that: The electronic device includes at least one processor and at least one memory connected to the processor, wherein the processor is used to call program instructions in the memory to execute the audio and video signal communication management method according to any one of claims 1 to 6.

9. A storage medium, characterized in that: The storage medium includes a stored program, wherein when the program is executed, the device where the storage medium is located is controlled to execute the audio and video signal communication management method according to any one of claims 1 to 6.

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