Audio and video signal communication management method and system

Abstract

The invention provides an audio and video signal communication management method and system, belongs to the field of computers, and solves the problem that a conference system cannot intelligently and dynamically adjust and control audio and video input and output equipment according to the specific condition of a conference in the prior art, which generally depends on manpower. The method comprises the steps that communication addresses are distributed to audio and video input devices and audio and video output devices in a local area network, audio and video signals from the multiple channels of audio and video input devices are routed to the designated multiple channels of audio and video output devices in a matrix mode, and the audio and video input devices comprise a plurality of microphones fixedly distributed in a conference area; under the condition that the target microphone is used for sound reception and sound amplification, the activation states of a plurality of other microphones distributed in the conference area except the target microphone are kept, so that the other microphones receive audio data around the microphones; and controlling the sound amplification volume of the target microphone based on the sound pressure level of the audio data received by other microphones distributed at different positions of the conference area.

Description

Technical Field

[0001] This application relates to the field of computer technology, and particularly to an audio - video signal communication management method and system. Background Art

[0002] IP matrix systems are widely used in various occasions that require efficient audio - video processing and control as well as intelligent environment management, especially in modern meeting rooms, multi - functional halls, classrooms, lecture halls, command centers, etc. Especially in the meeting scenario, the system can easily meet various meeting requirements, such as video conferencing, remote collaboration, product display, etc. Users can quickly configure the audio - video signal routing to ensure the clear display and transmission of meeting content. However, in related technologies, the meeting system cannot intelligently perform dynamic adjustment and control of audio - video input and output devices according to the specific situation of the meeting and usually relies on manual operation. Summary of the Invention

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

[0004] The first aspect of the embodiments of this application provides an audio - video signal communication management method, including:

[0005] Assign communication addresses to audio - video input devices and audio - video output devices in the local area network, so that audio - video signals from multiple audio - video input devices are routed to specified multiple audio - video output devices in a matrix manner. The audio - video input devices include multiple microphones fixedly distributed in the meeting area;

[0006] When using a target microphone for sound collection and amplification, keep the activation states of multiple other microphones in the meeting area except the target microphone, so that the other microphones receive audio data around themselves;

[0007] 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 positions in the meeting area, so as to increase the amplification volume of the target microphone when the sound pressure level is too low and decrease the amplification volume of the target microphone when the sound pressure level is too high.

[0008] Optionally, it further includes:

[0009] When a target audio - video output device is playing audio - video, keep the activation states of microphones distributed at different positions in the meeting area, so that the microphones receive audio data around themselves;

[0010] Control the audio and video playback volume of the target audio and video output device based on the sound pressure levels of the audio data received by the microphones distributed at different positions in the meeting 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 decrease 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 further includes:

[0012] Parse the audio data, and extract target audio data that matches the speech audio obtained by sound collection from the audio data;

[0013] Control the amplification volume of the target microphone based on the sound pressure levels of the target audio data received by the other microphones distributed at different positions in the meeting area.

[0014] Optionally, it further includes:

[0015] Parse the audio data, and extract target audio data that matches the played audio and video data from the audio data;

[0016] Control the audio and video playback volume of the target audio and video output device based on the sound pressure levels of the target audio data received by the microphones distributed at different positions in the meeting area.

[0017] Optionally, 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 meeting area includes:

[0018] In the case where 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 positions in the meeting area meet the preset sound pressure range; or,

[0019] Query the layout positions of the other microphones in the meeting area, and determine the nearest microphone closest to the amplification device and the farthest microphone farthest away;

[0020] Adjust the amplification volume of the target microphone 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, it further includes:

[0022] Assign temporary addresses to multiple user intelligent mobile terminals connected to the local area network, establish communication links with the multiple user intelligent mobile terminals based on the temporary addresses, and respectively obtain the location information of the multiple user intelligent mobile terminals based on the communication links;

[0023] Determine the actual distribution area of users in the meeting area based on the location information of multiple user intelligent mobile terminals obtained;

[0024] If it is determined that the actual distribution area of users in the meeting area is larger than the microphone distribution area, when using the target microphone for sound collection and amplification, in the case where it is determined that the actual distribution area of users in the meeting area is larger than the microphone distribution area based on the location information of multiple user intelligent mobile terminals obtained, when using the target microphone for sound collection and amplification, maintain the audio collection state of the multiple user intelligent mobile terminals, so that the multiple user intelligent mobile terminals receive the audio data around themselves;

[0025] Control the amplification volume of the target microphone based on the sound pressure level of the audio data collected by the multiple user intelligent mobile terminals distributed at different positions in the meeting area.

[0026] Optionally, it further includes:

[0027] Obtain the user information of the multiple user intelligent mobile terminals respectively based on the communication link;

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

[0029] Assign communication addresses to the audio and video input devices (user intelligent mobile terminals) and audio and video output devices in the local area network, and construct a matrix signal routing network, so that the audio and video signals of the target user intelligent mobile terminal are routed to the specified audio and video output device in a matrix manner.

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

[0031] A matrix unit, used to assign communication addresses to the audio and video input devices and audio and video output devices in the local area network, so that the audio and video signals from multiple audio and video input devices are routed to the specified multiple audio and video output devices through a matrix manner, and the audio and video input devices include multiple microphones fixedly distributed in the meeting area;

[0032] A monitoring unit, used to maintain the activation state of multiple other microphones other than the target microphone in the meeting area when using the target microphone for sound collection and amplification, so that the other microphones receive the audio data around themselves;

[0033] A control unit, 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 meeting area, so as to increase the amplification volume of the target microphone when the sound pressure level is too low and decrease the amplification volume of the target microphone when the sound pressure level is too high.

[0034] In a third aspect of the embodiments of the present application, an electronic device is provided, including a memory and a processor. When the processor executes a computer program stored in the memory, the steps of the above-mentioned audio and video signal communication management method are implemented.

[0035] In a fourth aspect of the embodiments of the present application, a computer-readable storage medium is provided, on which a computer program is stored. When the computer program is executed by a processor, the steps of the above-mentioned audio and video signal communication management method are implemented.

[0036] In summary, for the audio and video signal communication management method provided by the embodiments of the present application, communication addresses are assigned 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 specified multiple audio and video output devices in a matrix manner. The audio and video input devices include multiple microphones fixedly distributed in a meeting area; when using a target microphone for sound collection and amplification, the activation states of multiple other microphones other than the target microphone in the meeting area are maintained, so that the other microphones receive audio data around themselves; based on the sound pressure levels of the audio data received by the other microphones distributed at different positions in the meeting area, the amplification volume of the target microphone is controlled. Traditional microphones are only used for a single amplification function in meetings, while this solution enables multiple microphones to work together, expanding the collection function of other non-target microphones into an ambient sound pressure monitoring and feedback mechanism. This not only avoids waste of idle devices but also maximizes the value of the devices. The system can intelligently adjust the volume according to real-time sound pressure level data during the speech process without relying on manual intervention. This intelligent control based on existing devices improves the technical level of the conference sound system. The solution performs matrix management on the audio input / output devices of existing conference rooms without affecting the functions and usage habits of the existing system. It can be deployed through software configuration, avoiding the complexity of large-scale hardware transformation of the conference environment. The solution does not require additional new hardware, but makes full use of the microphones fixedly distributed in the existing conference room, realizes sound pressure monitoring and dynamic amplification adjustment through software logic and signal processing, and completes the intelligent upgrade of the system by introducing advanced sound pressure monitoring algorithms and matrix signal management technologies. It has significant economic benefits and has the advantages of low investment, quick results, and strong compatibility.

[0037] Correspondingly, the systems, electronic devices, and computer-readable storage media provided by the embodiments of the present invention also have the above-mentioned technical effects. BRIEF DESCRIPTION OF THE DRAWINGS

[0038] Figure 1 It is a schematic flowchart of a possible audio and video signal communication management method provided by the embodiments of the present application;

[0039] Figure 2Schematic structural block diagram of a possible audio - video signal communication management system provided by an embodiment of the present application;

[0040] Figure 3 Schematic hardware structure diagram of a possible audio - video signal communication management system provided by an embodiment of the present application;

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

[0042] Figure 5 Schematic structural block diagram of a possible computer - readable storage medium provided by an embodiment of the present application. Detailed implementation manners

[0043] The embodiments of the present application provide an audio - video signal communication management method and related devices, which can solve the problem in the related art that the conference system cannot dynamically adjust and control audio - video input and output devices intelligently according to the specific situation of the conference and usually relies on manual operation.

[0044] Terms such as "first", "second", "third", "fourth", etc. (if any) in the specification, claims and the above - mentioned drawings of the present application are used to distinguish similar objects and do not necessarily describe a specific order or sequence. It should be understood that such used data can be interchanged under appropriate circumstances so that the embodiments described herein can be implemented in an order different from that illustrated or described herein. In addition, the terms "comprising" and "having" and any variations thereof are intended to cover non - exclusive inclusion. For example, a process, method, system, product or device comprising a series of steps or units does not necessarily limit to those clearly listed steps or units, but may include other steps or units not clearly listed or inherent to these process, method, product or device. The technical solutions in the embodiments of the present application will be described clearly and completely below with reference to the accompanying drawings in the embodiments of the present application. Obviously, the described embodiments are only a part of the embodiments of the present application, rather than all of the embodiments.

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

[0046] S110, allocate communication addresses to audio - video input devices and audio - video output devices in the local area network, so that audio - video signals from multiple audio - video input devices are routed to specified multiple audio - video output devices in a matrix manner, and the audio - video input devices include multiple microphones fixedly distributed in the conference area;

[0047] S120, when using the target microphone for sound collection and amplification, keep the activation states of multiple other microphones distributed in the meeting area other than the target microphone, so that the other microphones receive the audio data around themselves;

[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 positions in the meeting area.

[0049] It can be understood that the core of this audio-visual signal communication management method lies in using multiple microphones distributed in the meeting area to collect sounds in real time, and dynamically adjusting the amplification volume of the target microphone based on the change in the sound pressure level (i.e., the physical intensity of the sound). Through this solution, automatic adjustment of the speaking volume can be achieved, avoiding manual intervention, thereby improving the meeting experience and the intelligent level of audio-visual signal transmission.

[0050] Exemplarily, by assigning communication addresses to the audio-visual input / output devices in the local area network, a matrix-style signal routing network is constructed to achieve dynamic switching of multi-channel signal input and output. This ensures that each input signal can be transmitted to the corresponding output device as needed, forming a flexible audio-visual signal distribution system. Multiple fixed microphones are arranged in the meeting area, which are not only used for amplification but also for real-time collection of the sound pressure level data in the environment. Non-target microphones (not selected for speaking) provide a basis for volume adjustment for the target microphone by sensing the sound pressure changes in the environment. Dynamically adjust the amplification volume of the target microphone according to the sound signals (especially the sound pressure level) captured by the non-target microphones. If the sound pressure level is too low, increase the amplification volume; if the sound pressure level is too high, decrease the amplification volume. Through the collaborative work of multiple microphones, the delay or error that may occur in the adjustment of a single microphone is reduced, ensuring uniform sound coverage. Through automatic adjustment, the inconvenience or error caused by manual volume adjustment is avoided.

[0051] Exemplarily, multiple microphones can be evenly distributed in the meeting area to capture the sound signals in different areas. High-sensitivity microphones are preferred to accurately capture subtle sound pressure changes. Speakers are distributed at key positions in the meeting area to ensure sound coverage. It can include an audio signal processor and a matrix routing device for audio signal routing and dynamic adjustment. And a communication module is used to achieve high-speed data transmission and synchronization between devices.

[0052] Exemplarily, the system assigns a unique communication address to each input / output device via a network and constructs a matrix routing table. The target microphone is used to collect the speaker's voice, and other microphones synchronously collect the surrounding sound pressure data. According to the matrix routing table, the audio signal of the target microphone is transmitted to the designated speaker. The sound pressure level captured by the non-target microphones is analyzed in real time, and the difference between its sound pressure level and that of the target microphone is calculated. The amplification volume of the target microphone is dynamically adjusted according to the 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 adjustment effect and optimizes the adjustment amplitude and frequency through intelligent algorithms. Taking the speaking scenario in a large conference room as an example, assume that there are 10 microphones distributed in the conference room and the speaker is located in the center. During the speech, the target microphone captures the speaker's voice, and the other 9 microphones monitor the sound distribution in different areas. If the sound pressure level captured by the target microphone is low and the sound pressure levels of the other microphones are significantly different, the system automatically increases the amplification volume of the target microphone to ensure that the sound reaches each area.

[0053] In summary, the audio and video signal communication management method provided in the embodiments of the present application assigns communication addresses to the audio and video input devices and audio and video output devices in the local area network, so that the audio and video signals from multiple audio and video input devices are routed to the 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; in the case of using the target microphone for sound collection and amplification, the activation states of the multiple other microphones other than the target microphone in the conference area are maintained, so that the other microphones receive the audio data around them; 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. Traditional microphones are only used for a single amplification function in meetings, while this solution expands the collection function of other non-target microphones to an ambient sound pressure monitoring and feedback mechanism through the collaborative work of multiple microphones. It not only avoids the waste of idle devices but also maximizes the value of the devices. The system can intelligently adjust the volume according to the real-time sound pressure level data during the speech without relying on manual intervention. This intelligent control based on existing devices improves the technical level of the conference sound system. The solution performs matrix management on the audio input / output devices of the existing conference room without affecting the functions and usage habits of the existing system. It can be deployed through software configuration, avoiding the complexity of large-scale hardware transformation of the conference environment. The solution does not require additional new hardware, but fully utilizes the microphones fixedly distributed in the existing conference room, realizes sound pressure monitoring and dynamic amplification adjustment through software logic and signal processing, and completes the intelligent upgrade of the system by introducing advanced sound pressure monitoring algorithms and matrix signal management technologies. It has significant economic benefits and has the advantages of low investment, quick results, and strong compatibility.

[0054] In some examples, it further includes:

[0055] When the target audio - video output device plays audio - video, keep the microphones distributed at different positions in the meeting area in an active state so that the microphones receive the audio data around themselves;

[0056] Control the audio - video playback volume of the target audio - video output device based on the sound pressure levels of the audio data received by the microphones distributed at different positions in the meeting area, so as to increase the audio - video playback volume of the target audio - video output device when the sound pressure level is too low and decrease the audio - video playback volume of the target audio - video output device when the sound pressure level is too high.

[0057] It can be understood that when the audio - video output device plays audio, all the microphones in the meeting room remain active to receive the ambient sound pressure data around. These microphones cover different areas of the meeting room, and the collected sound pressure level data can reflect the uniformity and volume state of the current sound field. Based on the multi - point sound pressure level monitoring data, calculate the adjustment range of the playback volume of the target speaker: if the sound pressure level in a certain area is low, increase the audio - video playback volume; if the sound pressure level is high, decrease the playback volume to avoid harsh sounds or echoes. The system intelligently adjusts the audio - video playback volume according to the real - time monitoring data, so that participants can obtain a uniform and comfortable auditory experience no matter which position they sit in the meeting room. Thus, in addition to dynamically adjusting the amplification of the target microphone, the dynamic control of the speaker playback volume further improves the sound field uniformity of the meeting, enabling the sound to spread without dead corners in the meeting area. Through real - time monitoring and intelligent feedback, the system does not require manual volume adjustment, significantly reducing the workload of meeting management. The solution is applicable to various scenarios such as single - venue meetings, remote collaboration, video presentations, etc., providing a consistent and high - quality audio - video experience. The system relies on existing meeting microphones and speakers, without the need for additional equipment investment, combining economy and efficiency. By dynamically adjusting the playback volume, optimizing the sound field coverage, eliminating the volume difference between regions, and enhancing the interactivity and participation of the meeting.

[0058] Exemplarily, in the scenario of video presentation combined with discussion, during the product presentation session, the speaker plays the explanatory audio of the video content. At the same time, the participants may discuss in a low voice. The non - target microphones real - time monitor the sound pressure data in different areas: if the discussion sound covers the explanatory audio, the system automatically increases the playback volume; if the playback volume is too high and affects the discussion, the system moderately reduces the playback volume.

[0059] In some examples, it further includes:

[0060] Analyze the audio data and extract the target audio data in the audio data that matches the speech audio obtained by sound collection;

[0061] Control the amplification volume of the target microphone based on the sound pressure levels of the target audio data received by the other microphones distributed at different positions in the meeting area.

[0062] It is understandable that, based on the original solution, an audio data parsing and matching function is added to enable the system to distinguish environmental sounds from speech audio, further improving the accuracy and reliability of dynamic sound amplification adjustment. By parsing the audio data received by the microphone and matching it with the speech audio collected by the target microphone, the system can identify the target audio data related to the speech, and thus adjust the sound amplification volume of the target microphone based on the sound pressure level of this data.

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

[0064] Exemplarily, the target microphone collects the audio data of the speaker and extracts its characteristic parameters. Other non-target microphones synchronously collect environmental sounds, parse the audio and extract its characteristics. The system matches the audio characteristics collected by non-target microphones with the speech audio characteristics of the target microphone, filtering out environmental noise or irrelevant sound sources. Only the target audio data that matches the speech audio is retained. According to the target audio data received by non-target microphones at different positions, the sound pressure level distribution of each region is calculated. If the sound pressure level of the target audio in a certain region is low, the system automatically increases the sound amplification volume; if the sound pressure level is too high, the system appropriately reduces the sound amplification volume to ensure uniform sound coverage. The system continuously monitors and adjusts the sound amplification effect, and optimizes the volume adjustment strategy in real time according to the changes in the sound field. Thus, through feature parsing and matching, the system only performs sound amplification control on the speech audio, filtering out the interference of environmental noise or irrelevant sound sources, and improving the accuracy of the sound amplification effect. The sound amplification volume is dynamically adjusted based on the sound pressure level of the target audio data, achieving intelligent adaptation to complex sound fields. The system can still maintain clear and balanced speech volume in a noisy environment or under multi-source interference, enhancing the auditory experience of participants. Without adding new hardware devices, it fully exploits the potential of existing microphones and audio-visual systems, and realizes intelligent function expansion through software upgrade. It is applicable to complex meeting scenarios, such as multi-speaker rotation, conference rooms with high background noise, and audio-visual mixed playback scenarios.

[0065] In some examples, it further includes:

[0066] Parse the audio data and extract target audio data that matches the played audio-visual data from the audio data;

[0067] Control the audio-visual playback volume of the target audio-visual output device based on the sound pressure levels of the target audio data received by the microphones distributed at different positions in the meeting area.

[0068] It can be understood that in a further complex meeting scenario, the audio data not only needs to match the speaking audio, but may also need to match the target audio-visual data (such as the audio part of the played video), so as to more precisely control the playback volume of the audio-visual output device. By parsing the audio data and extracting the target audio data that matches the played audio-visual data, the system can dynamically adjust the playback volume to ensure that the sound of the video content is evenly covered and not interfered by other environmental factors.

[0069] It can be understood that the system parses the audio data received by the non-target microphones and extracts feature information such as its spectral characteristics and time-domain characteristics. Compare the extracted data with the audio part of the played audio-visual to perform feature matching, and filter out the target audio data that is consistent with the audio-visual content. Based on the target audio data, calculate the sound pressure level distribution in different regions and evaluate the uniformity of sound coverage. Ignore other ambient sounds or speaking sounds and only optimize the sound field for the audio of the played audio-visual. According to the distribution of the sound pressure levels, adjust the playback volume of the target audio-visual output device in real time: when the sound pressure level is insufficient, increase the playback volume; when the sound pressure level is too high, appropriately reduce the playback volume to ensure a comfortable listening experience. The system can process the speaking audio and the audio-visual playback audio simultaneously, perform parsing and matching respectively, and dynamically adjust their sound pressure levels to ensure that the sound sources are clear and evenly covered. Thus, through parsing and matching, only monitor the sound pressure level and adjust the volume for the target audio of the played audio-visual to avoid interference from environmental noise or irrelevant sound sources. Dynamically adjust the playback volume of the target audio-visual device to ensure that contents such as video commentary and background sound effects are evenly covered in different regions and improve the participation experience. The system can process the speaking audio and the audio-visual audio separately in a multi-source scenario to achieve parallel optimization and enhance the intelligence of the system. Make full use of the existing meeting equipment, without additional hardware investment, and achieve the optimization of complex audio scenarios through software algorithm upgrades. The system can dynamically sense changes in the sound pressure level and adjust the playback volume in real time to ensure that the sound is always in the best state.

[0070] In some examples, 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 meeting area includes:

[0071] In the case that 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 positions in the meeting area meet the preset sound pressure range.

[0072] It can be understood that in some meeting scenarios, based on the sound pressure levels received by other microphones distributed in the meeting area, the amplification volume of the target microphone can be dynamically adjusted to meet the requirements of the preset sound pressure range. This mechanism ensures uniform and clear sound propagation in the meeting room by real-time monitoring of sound pressure level changes and combining dynamic volume adjustment strategies, without being too low or too high. The system determines whether each area of the meeting room meets the set sound pressure range (e.g., 50 - 70 dB SPL) according to the sound pressure levels collected by multiple microphones in real time. If the sound pressure level in some areas is lower than the lower limit of the preset range, it indicates insufficient sound 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 decreased. The system continuously monitors the adjusted sound pressure level through a real-time feedback mechanism until the sound pressure levels of all areas meet the preset range. This closed-loop control process ensures the timeliness and accuracy of the adjustment. By adjusting the amplification volume of the target microphone, the coverage effect of sound in the meeting area is optimized, avoiding the situation where participants cannot hear clearly or the sound is too loud.

[0073] Exemplarily, the system collects audio data in real time through distributed microphones and calculates the sound pressure levels at the positions of each microphone. By comparing with a preset range (such as 50 - 70 dB SPL), it determines 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 amplification 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 decreases the amplification volume of the target microphone until the sound pressure level enters the range. While adjusting the volume, the system continuously monitors the change in the sound pressure level and further optimizes the amplitude and speed of volume adjustment based on the latest data to ensure uniform sound propagation. When the sound pressure levels at all monitoring points meet the preset range, the system enters a stable state and maintains the volume unchanged. Taking the speaking sound field in 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 relatively small, and the sound pressure level in some areas is only 45 dB SPL, lower than the preset range (50 - 70 dB SPL). The system automatically increases the amplification volume of the target microphone to raise the sound pressure level monitored by the distributed microphones within the range. Finally, the sound pressure in all areas reaches 55 - 65 dB SPL, and the participants have a comfortable listening experience. Or in some cases, such as during a heated discussion, the speaker's volume is high, and the sound pressure level in some areas reaches 80 dB SPL, exceeding the preset range (50 - 70 dB SPL). The system decreases the amplification volume of the target microphone to gradually lower the sound pressure level and avoid harsh sounds or echoes. After adjustment, the sound pressure levels in all areas return to the range of 65 - 70 dB SPL, and the meeting environment is more comfortable. Thus, whether in a large - space scenario or with multiple speaker switches, 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, it adjusts the amplification volume of the target microphone to optimize the sound coverage within the entire 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, without the need for participants to deliberately raise or lower their voices, and there will be no situation of unclear or harsh sounds. The system realizes intelligent adjustment based on existing microphones, speakers, and signal processing equipment without additional hardware investment.

[0074] 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 positions in the meeting area includes:

[0075] Query the arrangement positions of the other microphones in the meeting area to determine the nearest microphone closest to the amplification device and the farthest microphone farthest away;

[0076] Adjust the amplification volume of the target microphone 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.

[0077] It is understandable that by querying the arrangement positions of other microphones in the meeting area, the microphone closest to the target sound amplification device and the farthest microphone are dynamically determined. Based on the sound pressure levels received by the microphones at these two extreme positions, the amplification volume of the target microphone is dynamically adjusted until their sound pressure levels both meet the preset sound pressure range. This mechanism ensures the uniformity of sound coverage and avoids the problems of excessive or insufficient sound in certain areas. The system can query the physical positions of each microphone according to the fixed arrangement of microphones in the meeting room and record their distances from the target sound amplification device. Determine the two microphones closest and farthest from the sound amplification device as the key reference points for evaluating the sound coverage effect. Real-time monitor the sound pressure levels received by the closest and farthest microphones and determine whether they are within the preset range (e.g., 50 - 70 dB SPL). In the case where the sound pressure level does not meet the range, adjust the amplification volume of the target microphone. 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 meet the preset range. Ensure that the sound from the closest point to the farthest point in the coverage area is uniform and clear. Thus, by dynamically adjusting the amplification volume of the target microphone, ensure uniform sound coverage from the closest point to the farthest point. The dual-point monitoring mechanism of the closest and farthest microphones prevents the sound amplification device from being too loud in the close-range area or having insufficient coverage in the long-range area. The system has a closed-loop adjustment function and can monitor and optimize the amplification volume in real time during the speech. The solution is compatible with meeting rooms of different scales and adapts to various meeting environments through position calibration and dynamic adjustment. It can ensure uniform sound propagation without manual intervention, improving the auditory comfort and participation of the participants.

[0078] In some examples, it further includes:

[0079] Assign temporary addresses to multiple user intelligent mobile terminals that join the local area network, establish communication links with the multiple user intelligent mobile terminals based on the temporary addresses, and respectively obtain the location information of the multiple user intelligent mobile terminals based on the communication links;

[0080] Determine the actual distribution area of users in the meeting area based on the obtained location information of the multiple user intelligent mobile terminals;

[0081] If it is determined that the actual distribution area of users in the meeting area is larger than the microphone distribution area, when using the target microphone for sound collection and amplification, maintain the audio collection state of the multiple user intelligent mobile terminals so that the multiple user intelligent mobile terminals receive the audio data around themselves;

[0082] Control the amplification volume of the target microphone based on the sound pressure levels of the audio data collected by the multiple user intelligent mobile terminals distributed at different positions in the meeting area.

[0083] It is understandable that by allocating 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 can be dynamically obtained. Especially when the user distribution area exceeds the coverage range of fixed microphones, the intelligent mobile terminal is used as an auxiliary audio acquisition device to further optimize the conference sound coverage effect and the amplification volume control of the target microphone.

[0084] It is understandable 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 obtains the location information of the intelligent mobile terminal through the communication link to determine the actual distribution range of users. When the actual distribution range of users is larger than the coverage area of fixed microphones, 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 amplification 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 from the data collected by the fixed microphone and the intelligent mobile terminal, the amplification volume of the target microphone is dynamically adjusted to optimize the conference sound effect.

[0085] Exemplarily, the user's intelligent mobile terminal accesses the system through a local area network. The system assigns a temporary address to each terminal and establishes a communication link. The system obtains the location information of the user's intelligent mobile terminal through the communication link and draws a user distribution map. If the user distribution range exceeds the coverage area of the fixed microphone, the system activates the audio collection function of the intelligent terminal. The intelligent terminal collects the ambient audio data, which together with the data collected by the fixed microphone constitutes the complete sound field distribution information. The system judges the uniformity of sound coverage according to the sound pressure level differences at different positions. 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 decreased. By integrating the data of the fixed microphone and the intelligent terminal, the sound coverage in the entire meeting area is ensured to be uniform and clear. The system dynamically adjusts the amplification volume according to the real-time changing user positions and sound pressure level data to optimize the sound effect. Thus, the monitoring blind area of the fixed microphone is supplemented by the intelligent terminal to achieve a more comprehensive sound field coverage. Based on the actual distribution positions of the users and the comprehensive sound pressure data, the amplification volume is dynamically adjusted to ensure uniform distribution of sound in the meeting area. The system can dynamically adapt to changes in the user distribution range, and can achieve real-time optimization whether it exceeds the coverage range of the fixed microphone or in the case of a group discussion scenario. Without additional hardware investment, the audio collection capabilities of the user's intelligent mobile terminal are fully utilized to reduce the system deployment cost. Whether the users are distributed in the meeting center or the edge area, they can obtain clear and balanced sound effects, enhancing the sense of participation and experience. Taking the case of a meeting where the user distribution exceeds the range of the fixed microphone as an example, 6 microphones are fixedly arranged in a meeting room to cover the meeting center area, but some participants are distributed in the edge or remote areas. The users' smartphones access the system through the local area network. The system obtains the user distribution information and finds that some users are outside the coverage range of the fixed microphone. The system activates the audio collection function of the smartphones, collects the sound pressure level data in the edge area through the mobile phones, and dynamically adjusts the amplification volume of the target microphone to ensure sound coverage in the edge area. By dynamically assigning temporary addresses to the intelligent mobile terminals, establishing communication links and obtaining location information, this solution integrates the audio collection capabilities of the intelligent terminals into the meeting audio management system. In the case where the user distribution exceeds the coverage range of the fixed microphone, the audio collection function is supplemented by the intelligent terminal, and the amplification volume of the target microphone is dynamically adjusted based on its sound pressure level, significantly improving the meeting audio coverage range and sound effect quality, providing a flexible and intelligent audio solution for modern meeting scenarios.

[0086] In some examples, it further includes:

[0087] Based on the communication link, obtain the user information of the multiple user intelligent mobile terminals respectively;

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

[0089] Assign communication addresses to the audio and video input devices, intelligent mobile terminals, and audio and video output devices in the local area network, and construct a matrix signal routing network to route the audio and video signals of the target intelligent mobile terminal to the specified audio and video output device in a matrix manner.

[0090] It can be understood that this audio and video signal communication management method aims to solve problems such as volume adjustment relying on manual operation, uneven sound coverage, and mismatch between the speaker's position and the microphone's position in traditional conference systems. By utilizing the fixed microphones in the conference room, the intelligent mobile terminals of the participants, and the matrix audio and video signal routing, intelligent acquisition, dynamic adjustment, and efficient transmission of audio signals are achieved. Assign communication addresses to the audio and video input devices (fixed microphones, user intelligent mobile terminals) and audio and video output devices (speakers, display devices) in the local area network, and construct a matrix signal routing network. Through matrix routing, the audio and video signals are flexibly transmitted from the input device to the specified output device. Fixed microphone acquisition: Multiple fixed microphones distributed in the conference area are used for sound collection, amplification, and ambient sound pressure level monitoring. Intelligent mobile terminal acquisition: When the distribution area of the participants exceeds the coverage range of the fixed microphones, or the position of the speaker does not match the position of the microphone, the intelligent mobile terminals of the participants (such as mobile phones, tablets) are used for audio acquisition and position information acquisition. Analyze the collected audio data to extract the target audio data that matches the speech audio or the played audio and video data. Filter out environmental noise and irrelevant sound sources to ensure the accuracy of audio processing. Based on the sound pressure levels of the target audio data collected by the fixed microphones and intelligent mobile terminals distributed in different positions, adjust the amplification volume of the target microphone in real time. Ensure uniform sound coverage in the conference area and avoid excessive or too low sound. In the case where the position of the speaker does not match the position of the fixed microphone, determine the target intelligent mobile terminal as the audio and video input device based on the user information. Through matrix routing, transmit the audio and video signals of the target intelligent mobile terminal to the specified audio and video output device to achieve flexible audio acquisition and transmission.

[0091] Exemplarily, after the system starts up, communication addresses are assigned to all fixed microphones, speakers, and user intelligent mobile terminals. An equipment matrix is constructed to prepare for the routing control of audio and video signals. The user intelligent mobile terminal accesses the local area network, and the system assigns a temporary address to it. Through the communication link, the location information and user information of the intelligent mobile terminal are obtained. The fixed microphone continuously collects ambient audio data for sound collection, amplification, and sound pressure level monitoring. The intelligent 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 analyzed to extract the target audio data that matches the speech audio or the played audio and video data. The sound pressure levels of the target audio data received at each location (fixed microphone and intelligent mobile terminal) are calculated. It is judged whether the sound pressure level is within the preset range to evaluate the uniformity of sound coverage. Based on the sound pressure level data, the amplification volume of the target microphone is adjusted to ensure that the sound pressure level in each area meets the preset range. When playing audio and video content, the playback volume of the audio and video output device is adjusted to ensure uniform sound coverage. When the speaker position does not match the fixed microphone position, based on the user information, the intelligent mobile terminal of the speaker is determined as the target audio and video input device. Through matrix routing, the audio and video signals of this terminal are transmitted to the specified audio and video output device to achieve efficient audio transmission. The system continuously monitors the changes in the sound pressure level and user location, and dynamically adjusts the volume and signal routing strategy to ensure that the conference sound effect is always in the best state.

[0092] Thus, by obtaining the location information of the user's intelligent mobile terminal, the system can understand the actual distribution of the participants in real time. When the participants are outside the coverage area of the fixed microphone, the audio collection function of the intelligent mobile terminal is used to supplement the audio monitoring blind area. The sound coverage area is expanded to ensure that all participants can clearly hear the meeting content. The adaptability of the system is improved to meet the meeting requirements of different scales and layouts. In the case where the position of the speaker does not match the position of the fixed microphone, the system determines the intelligent mobile terminal of the speaker as the audio and video input device based on the user information. Through matrix routing, the audio and video signals of the speaker are efficiently transmitted to the designated output device. The speaker does not need to deliberately move to the position of the fixed microphone, which improves the flexibility and efficiency of the meeting. It ensures clear speech quality without being restricted by the position. By combining the audio data of the fixed microphone and the intelligent mobile terminal, the system comprehensively monitors the sound field in the meeting area. Based on the comprehensive sound pressure level data, the amplification volume is dynamically adjusted to ensure uniform sound distribution. It avoids the problems of uneven sound coverage, too loud or too soft sound in some areas. The auditory comfort of the participants is improved and the meeting experience is enhanced. The audio data is analyzed to extract the target audio data that matches the speech audio or the played audio and video data. The environmental noise and irrelevant sound sources are filtered to ensure the accuracy of volume adjustment and signal transmission. The accuracy of audio processing is improved and noise interference is avoided. The quality of the audio signal is improved to ensure the smooth progress of the meeting. 5. Make full use of existing equipment to reduce the implementation cost. By using the existing fixed microphones in the meeting room and the intelligent mobile terminals of the participants, there is no need to add additional hardware devices. Through software function expansion, the intelligence and function of the system are improved. The system deployment and maintenance costs are reduced, which has significant economic advantages. The utilization rate of the equipment is improved and resource waste is avoided.

[0093] Taking the example where participants are outside the coverage area of the fixed microphone in a large-scale meeting, that is, in a large-scale meeting, there are many participants distributed in various corners of the meeting room, and some areas are outside the coverage area of the fixed microphone. The smartphones of the participants are connected to the meeting local area network, and the system assigns a temporary address to them and obtains the location information. The system finds that the distribution area of the participants is larger than the coverage area of the fixed microphone, and activates the audio collection function of the smartphones. By combining the audio data of the fixed microphone and the smartphones, the system dynamically adjusts the amplification volume of the target microphone to ensure uniform sound coverage throughout the venue. Thus, all participants can clearly hear the meeting content and the sound is evenly transmitted. The system makes full use of the intelligent terminals of the participants without additional equipment.

[0094] According to some embodiments, in the case where the participants are far away and the received sound is small, the voice signal is routed to the intelligent mobile terminal of the participant itself in a matrix manner, and the participant can directly play the audio content through their intelligent device, thus ensuring a clear listening experience.

[0095] Exemplarily, when the meeting room is large or the participants are in the coverage blind area of the sound reinforcement equipment (such as the corners of the room, sub-venues, etc.), the sound attenuation results in a lower sound pressure level in these areas, and the participants cannot clearly hear the speech content. The system can directly route the audio signal of the target microphone to the intelligent mobile terminal of the participants, enabling them to receive and play the sound through their own devices. The system utilizes the audio-visual signal matrix routing technology to route the audio signal of the speaker to the intelligent terminal of the designated participant. Each participant joins the local area network through their intelligent terminal and serves as a receiving device to form a distributed audio coverage system. After receiving the routing signal, the intelligent mobile terminal of the participant plays it through the built-in speaker or earphone. The system can dynamically adjust the playback volume of the terminal to ensure coordination with the overall meeting sound environment. The participants can independently control the volume of their intelligent terminals to suit their personal preferences without affecting other participants.

[0096] Exemplarily, the intelligent mobile terminal of the participant accesses the local area network, and the system assigns a unique communication address to each terminal. The system establishes an audio transmission path from the target microphone to the intelligent terminal through matrix routing. The system determines which areas of the participants cannot clearly hear the sound based on the sound pressure level data collected by the fixed microphone and the intelligent mobile terminal. The system sets the terminals of the participants in these areas as the target receiving devices for the audio signal. The speech audio signal collected by the target microphone is transmitted to the intelligent terminal of the designated participant through matrix routing. After receiving the signal, the intelligent terminal plays the sound through its built-in speaker or earphone. The system dynamically adjusts the routing path and the playback volume of the terminal according to the real-time monitored sound field changes to ensure that the meeting sound effect is always optimal. Thus, the coverage deficiency of the fixed microphone and speaker is supplemented by the intelligent mobile terminal, solving the problem that the remote participants cannot hear clearly. The participants can adjust the volume of the terminal by themselves to meet their personal listening needs without affecting other participants. Without adding hardware devices, only by utilizing the existing intelligent terminals of the participants can the sound coverage be optimized, reducing the system cost. The participants directly listen to the sound through the terminal, improving their sense of participation and attention to the meeting content. It is applicable to various scenarios such as large meetings, small discussions, and multi-sub-venue collaborations, meeting the needs of different scales.

[0097] Exemplarily, the intelligent mobile terminal of the user can receive and play the audio signal without installing a dedicated application (App), by using the built-in browser of the terminal or common network protocols. This method avoids the burden of the user installing additional applications, while ensuring the ease of use and compatibility of the system. For example, the participant accesses the URL provided by the meeting system (such as a web page within the local area network) through the browser of the terminal. The meeting system pushes the audio signal to the terminal through WebRTC technology or real-time audio stream (such as HTTP Live Streaming, HLS). The user does not need to download any software, and only needs to open the web page to receive and play the audio through the browser.

[0098] According to some embodiments, intelligent mobile terminals distributed in the meeting 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 utilizes the multi-point audio collection function of distributed terminals to achieve environmental noise reduction and speech signal enhancement through audio data fusion technology. For example, when a participant speaks, multiple terminals closest to the speaker collaborate to collect the 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 stable audio collection. Thus, breaking through the limitation of single-terminal signal collection and combining the dynamic collaboration of multiple distributed terminals significantly improves the audio signal quality. It is applicable to meeting environments where the distribution of participants changes dynamically and the noise is complex.

[0099] According to some embodiments, it may also include a "pre-listening synchronization" function, which provides a real-time audio preloading mechanism at the user terminal through distributed caching technology, enabling the terminal to obtain a synchronized audio playback effect 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 slices according to microsecond-level timestamps to ensure seamless playback even when the network fluctuates. Before signal pushing, the system loads a small amount of audio preview data for the user terminal to maintain playback synchronization during network latency fluctuations. For example, in a remote meeting scenario, participants at different locations can hear the speech simultaneously without delay or stuttering. Thus, the playback synchronization is improved: even under complex network conditions, each participant can still obtain a real-time and synchronized audio experience. The reliability is enhanced: the dependence on network fluctuations for real-time audio transmission is reduced through the caching mechanism.

[0100] The above describes the audio and video signal communication management method in the embodiments of the present application. Next, the audio and video signal communication management system in the embodiments of the present application will be described.

[0101] Please refer to Figure 2 , an embodiment of the audio and video signal communication management system described in the embodiments of the present application may include:

[0102] A matrix unit 201, configured to assign communication addresses to audio and video input devices and audio and video output devices in the local area network, so that audio and video signals from multiple audio and video input devices are routed to specified multiple audio and video output devices through a matrix method. The audio and video input devices include multiple microphones fixedly distributed in the meeting area;

[0103] The monitoring unit 202 is configured to keep the activation states of a plurality of other microphones other than the target microphone distributed in the meeting area when using the target microphone for sound collection and amplification, so that the other microphones receive the audio data around themselves.

[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 positions in the meeting area.

[0105] In summary, the audio and video signal communication management system provided by the above embodiments assigns communication addresses to the audio and video input devices and audio and video output devices in the local area network, so that the audio and video signals from multiple audio and video input devices are routed to the specified multiple audio and video output devices in a matrix manner. The audio and video input devices include a plurality of microphones fixedly distributed in the meeting area. When using the target microphone for sound collection and amplification, keep the activation states of a plurality of other microphones other than the target microphone distributed in the meeting area, so that the other microphones receive the audio data around themselves. 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 positions in the meeting area. Traditional microphones are only used for a single amplification function in meetings, while this solution expands the collection function of other non-target microphones into an ambient sound pressure monitoring and feedback mechanism through the collaborative work of multiple microphones. It not only avoids the waste of idle devices but also maximizes the value of the devices. The system can intelligently adjust the volume according to the real-time sound pressure level data during the speech process without relying on manual intervention. This intelligent control based on existing devices improves the technical level of the conference sound system. The solution performs matrix management on the audio input / output devices in the existing conference room without affecting the functions and usage habits of the existing system. It can be deployed through software configuration, avoiding the complexity of large-scale hardware transformation of the conference environment. The solution does not require additional new hardware, but makes full use of the microphones fixedly distributed in the existing conference room, realizes sound pressure monitoring and dynamic amplification adjustment through software logic and signal processing, and completes the intelligent upgrade of the system by introducing advanced sound pressure monitoring algorithms and matrix signal management technologies. It has remarkable economy and has the advantages of less investment, quick effect, and strong compatibility.

[0106] Above Figure 2 The audio and video signal communication management system in the embodiments of the present application has been described from the perspective of modular functional entities. Next, the audio and video signal communication management system in the embodiments of the present application will be described in detail from the perspective of hardware processing. Please refer to Figure 3 An embodiment of the audio and video signal communication management system 300 in the embodiments of the present application includes:

[0107] An input device 301, an output device 302, a processor 303, and a memory 304, where the number of processors 303 can be one or more. Figure 3 Here, one processor 303 is taken as an example. In some embodiments of the present application, the input device 301, the output device 302, the processor 303, and the memory 304 can be connected through a bus or other means, where Figure 3 here, connection through a bus is taken as an example.

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

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

[0110] When using the target microphone for sound collection and amplification, keep the activation states of multiple other microphones other than the target microphone in the conference area, so that the other microphones receive the audio data around themselves;

[0111] 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 positions in the conference area.

[0112] By invoking the operation instructions stored in the memory 304, the processor 303 is further configured to execute Figure 1 any one of the corresponding embodiments.

[0113] Please refer to Figure 4 , Figure 4 which is a schematic diagram of an embodiment of the electronic device provided by the embodiment of the present application.

[0114] As Figure 4 shown, the 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 on 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] Assign communication addresses to the audio - video input devices and audio - video output devices in the local area network, so as to route the audio - video signals from multiple audio - video input devices to the specified multiple audio - video output devices in a matrix manner. The audio - video input devices include multiple microphones fixedly distributed in the conference area;

[0116] When using the target microphone for sound collection and amplification, keep the activation states of multiple other microphones other than the target microphone distributed in the meeting area, so that the other microphones receive the audio data around themselves;

[0117] 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 positions in the meeting area.

[0118] In the specific implementation process, when the electronic device processor 420 executes the computer program 411, it can implement Figure 1 any one of the corresponding embodiments.

[0119] Since the electronic device introduced in this embodiment is the device used in implementing an audio and video signal communication management system in an embodiment of the present application, based on the method introduced in the embodiment of the present application, those skilled in the art can understand the specific implementation manners and various variations of the electronic device in this embodiment. Therefore, the specific implementation of how this electronic device implements the method in the embodiment of the present application will not be introduced in detail here. As long as the device used by those skilled in the art to implement the method in the embodiment of the present application belongs to the scope protected by the present application.

[0120] Please refer to Figure 5 , Figure 5 which is a schematic diagram of an embodiment of a computer-readable storage medium provided by an embodiment of the present application.

[0121] As Figure 5 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] Assign communication addresses to the audio and video input devices and audio and video output devices in the local area network, so that the audio and video signals from multiple audio and video input devices are routed to the specified multiple audio and video output devices through a matrix method. The audio and video input devices include multiple microphones fixedly distributed in the meeting area;

[0123] When using the target microphone for sound collection and amplification, keep the activation states of multiple other microphones other than the target microphone distributed in the meeting area, so that the other microphones receive the audio data around themselves;

[0124] 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 positions in the meeting 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 processes or functions described in the embodiments of the present application are generated in whole or in part. The computer may be a general-purpose computer, a special-purpose computer, a computer network, or other programmable devices. 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 a website, computer, server, or data center to another website, computer, server, or data center by wire (such as coaxial cable, optical fiber, digital subscriber line (DSL)) or wirelessly (such as infrared, wireless, microwave, etc.). The computer-readable storage medium may be any available medium that can be stored by a computer or a data storage device such as a server or data center that includes one or more integrated available media. The available medium may be a magnetic medium (such as a floppy disk, hard disk, magnetic tape), an optical medium (such as a DVD), or a semiconductor medium (such as a solid state disk (SSD)), etc.

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

[0127] In several embodiments provided in the present 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 illustrative. For example, the division of the units is only a logical function division, and there may be other division methods in actual implementation. For example, multiple units or components may be combined or integrated into another system, or some features may be ignored or not executed. Another point is that the couplings, direct couplings, or communication connections shown or discussed with each other may be indirect couplings or communication connections through some interfaces, devices, or units, and may be in electrical, mechanical, or other forms.

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

[0129] In addition, in each embodiment of the present application, each functional unit can be integrated into a processing unit, or each unit can exist physically alone, or two or more units can be integrated into one unit. The above-mentioned integrated unit can be implemented in the form of hardware or in the form of a software functional unit.

[0130] If the above-mentioned integrated unit is implemented 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 solution of the present application, in essence, or the part that contributes to the prior art, or all or part of the technical solution, 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 described in each embodiment of the present application. The foregoing storage medium includes: various media such as USB flash drives, mobile hard disks, read-only memory (ROM), random access memory (RAM), magnetic disks, or optical discs that can store program codes.

[0131] As described above, the above embodiments are only used to illustrate the technical solutions of the present application and are not intended to limit them; although 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 modify the technical solutions described in the foregoing embodiments, or perform equivalent replacements on some of the technical features; 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 each embodiment 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, wherein the audio and video input devices include multiple microphones fixedly distributed in the conference area; When the target microphone is used for sound collection 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 around themselves; The amplification volume of the target microphone is controlled 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.

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 positions 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 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.

3. The method according to claim 1, characterized in that Also includes: Parsing the audio data, and extracting target audio data matching the speech audio obtained by sound recording from the audio data; 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, 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.

4. The method according to claim 2, characterized in that: Also includes: Parsing the audio data, and extracting target audio data matching 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.

5. The method according to claim 1, characterized in that The method of 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 level of the audio data received by the other microphones distributed at different positions in the conference area meets 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 at the farthest distance; 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.

6. The method according to any one of claims 1 to 5, characterized in that Also includes: Allocating temporary addresses to multiple user intelligent mobile terminals that join the local area network, establishing communication links with the multiple user intelligent mobile terminals based on the temporary addresses, and respectively acquiring location information of the multiple user intelligent 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 intelligent 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, 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; 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 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.

7. The method according to claim 6, characterized in that Also includes: Based on the communication link, respectively obtain user information of the multiple user intelligent mobile terminals; In the case where the microphone position does not match the position of the selected user to speak, determining the target user intelligent 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.

8. An audio and video signal communication management system, characterized in that: The system comprises: A matrix unit, 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, wherein the audio and video input devices include multiple microphones fixedly distributed in the conference area; A monitoring unit, configured to maintain the activation state of a plurality of other microphones distributed in the conference area other than the target microphone when the target microphone is used for sound collection and amplification, so that the other microphones receive the audio data around themselves; 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.

9. 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 7.

10. A storage medium, characterized in that: The storage medium includes a stored program, wherein when the program is running, 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 7.

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