Indoor interactive display method and device

Abstract

The present invention discloses an indoor interactive display method and device. The present invention uses an indoor detection module to capture the sound source position and ambient sound intensity in real time, dynamically adjust the volume of the sound reinforcement matrix unit, create a gradient volume distribution, and enhance the auditory immersion. The system also adjusts the volume in real time according to the sound intensity, and activates the interactive projection unit to display the interactive content. In addition, the lighting and music output are synchronized to ensure the coordination and consistency of the audio-visual experience. This solves the problem that the indoor interactive technology in the prior art cannot meet the needs of users.

Description

Technical Field

[0001] The present invention relates to the field of indoor interactive display, and in particular to an indoor interactive display method and device. Background Art

[0002] With the rapid development of science and technology, exhibition halls, museums, science and technology museums and other places are constantly seeking innovation in display methods and visitor experience. The existing exhibition hall scene integrates a variety of advanced technologies, providing an immersive, multi-sensory interactive experience, greatly improving the display effect and user engagement. Exhibition halls are usually used for commercial displays, art exhibitions and brand promotion. With the intensification of market competition, exhibition hall managers pay more and more attention to innovative display methods to attract audiences and enhance brand image.

[0003] With the advancement of technology, interactive audio-visual display systems are playing an increasingly important role in exhibition halls, museums and science and technology museums. These venues have improved the display effect and audience experience by introducing advanced audio, visual, interactive and information technologies, solving many pain points of traditional display methods. However, the complexity and high cost of technology remain challenges, requiring reasonable planning and design to achieve the best results. These technologies not only enrich the display content, but also provide broad space for future exhibition innovation.

[0004] Although the existing interactive display system provides rich display methods and innovative user experience in exhibition halls, museums and science and technology museums, there are still some shortcomings in the existing technology. The following are some common problems:

[0005] 1. Interactive audio technology: The positioning accuracy of spatial audio and 3D audio is limited, especially in large spaces or multi-reflection environments, the positioning accuracy may be affected.

[0006] 2. Augmented reality (AR) and virtual reality (VR) technology: AR technology is highly dependent on equipment and AR-related content development is complex. Currently, most AR experiences require specific hardware devices, such as AR glasses or high-performance mobile devices, which limits the scope of the audience, and creating high-quality AR content requires a lot of time and resources, including 3D modeling, animation and interaction design, and high development costs.

[0007] 3. Interactive projection technology: Projection mapping is complex and has high maintenance costs. In addition, the accuracy of touch and gesture recognition may decrease when multiple people interact at the same time or perform fast gesture operations, affecting the user experience.

[0008] 4. Intelligent lighting control technology: The intelligent lighting system requires professional installation and debugging, and the initial installation cost is high. The operation of the system depends on the stability of the sensor and control system, and technical failures may affect the display effect.

[0009] 5. Intelligent information terminal: It requires high-quality touch screen and voice recognition equipment, which has high equipment cost. Moreover, the voice recognition commands are relatively fixed and cannot maximize the user experience.

[0010] These shortcomings need to be improved in practical applications to enhance the stability, user experience and security of the technology while controlling costs and complexity. Summary of the invention

[0011] The present invention provides an indoor interactive display method and device to solve the problem that the indoor interactive technology in the prior art cannot meet the needs of users.

[0012] In a first aspect, the present application provides an indoor interactive display method, comprising:

[0013] Obtain the location information of each sound source and each real-time ambient sound intensity data in the detection modules of each area of ​​the room;

[0014] According to the respective sound source position information, the output of each sound reinforcement matrix unit in the area corresponding to each sound source position information is adjusted so that the output volume of each sound reinforcement matrix unit decreases step by step from the installation position of each interactive projection unit to both sides in a ladder-like manner;

[0015] According to the real-time environmental sound intensity data, the output volume of each sound reinforcement matrix unit in the area corresponding to each sound source position information is dynamically adjusted, so that each sound reinforcement matrix unit controls the output volume according to the real-time environmental sound intensity data.

[0016] This application obtains the sound source location information through the detection module, and adjusts the output volume of the sound reinforcement matrix unit based on the sound source location information, creating a ladder-shaped volume distribution that decreases from the projection unit to both sides, creating an immersive auditory environment. In addition, this application also automatically adjusts the volume by dynamically analyzing the real-time environmental sound intensity data to ensure that visitors can get a clear auditory experience in different environments. This application not only optimizes the audience's interactive experience, but also improves the adaptability and efficiency of the display environment through intelligent adjustment, solving the problem that the indoor interactive technology in the existing technology cannot meet user needs.

[0017] As a preferred embodiment of the first aspect, the acquisition of each sound source position information and each real-time ambient sound intensity data in the detection module of each indoor area is specifically:

[0018] The detection modules for each indoor area include each microphone array unit and each signal processing unit;

[0019] Each microphone array unit is used to collect each sound signal in the room and transmit each sound signal to each signal processing unit;

[0020] Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain position information of each sound source;

[0021] Each of the signal processing units calculates and obtains each of the real-time environmental sound intensity data by analyzing the intensity changes of each of the sound signals.

[0022] As a preferred embodiment of the first aspect, each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain each sound source position information; each signal processing unit calculates and obtains each real-time environmental sound intensity data by analyzing the intensity change of each sound signal, specifically:

[0023] According to the time difference positioning method, each signal processing unit determines the position of each sound source of each sound signal in combination with the geometric layout of the microphone array in each microphone array unit to obtain the position information of each sound source;

[0024] Each of the signal processing units continuously monitors the strength and quality of each of the sound signals to obtain each real-time monitoring information;

[0025] According to the various real-time monitoring information, various signal processing parameters are dynamically adjusted to obtain various real-time environmental sound intensity data.

[0026] In this preferred embodiment, the present application equips the detection module in each area with a microphone array unit, which is responsible for collecting the sound signals in the area in real time and quickly transmitting these signals to the corresponding signal processing unit. The signal processing unit uses the sound source localization algorithm to process the sound signal, accurately identify and determine the specific location of each sound source, and provide key spatial information. At the same time, these units also analyze the intensity changes of the sound signal and calculate the environmental sound intensity data of each area in real time. This meticulous regional processing method can customize the response to the sound environment in different areas, thereby optimizing the sound output and enhancing the overall auditory experience. This regionalized sound data collection and processing not only enhances the accuracy of sound source localization, but also improves the sensitivity and response speed to changes in environmental sound, providing a technical basis for creating a highly interactive and immersive display environment.

[0027] As a preferred embodiment of the first aspect, it also includes identifying and activating the interactive projection units in each area of ​​the room according to the location information of each sound source and the preset task configuration information, so that the interactive projection units in each area of ​​the room display content related to the interaction with tourists.

[0028] In this preferred embodiment, the present application uses widely distributed microphone array units to monitor the indoor sound environment in real time and obtain sound source location information. Then, the signal processing unit analyzes the collected sound signal to determine the specific location of the sound source. At the same time, the intensity change of the sound signal is also analyzed to obtain real-time environmental sound intensity data. These data are used to compare with the preset task configuration information. Once a matching sound source or sound intensity condition is detected, the system triggers the interactive projection unit in the corresponding area to display customized content. This method not only improves the interactivity and pertinence of the displayed content, but also enhances the visitors' sense of participation and quality of experience.

[0029] As a preferred embodiment of the first aspect, it also includes synchronously outputting the operation of the lighting units in each area of ​​the room and the sound reinforcement matrix units in each area of ​​the room according to the real-time ambient sound intensity data, so that the music output by the sound reinforcement matrix units in each area of ​​the room is coordinated with the performance output by the lighting units in each area of ​​the room.

[0030] In this preferred embodiment, the present application captures indoor sound through a microphone array unit and transmits it to a signal processing unit for preprocessing, then applies a sound source localization algorithm to determine the location of the sound source, and then analyzes the change in sound signal intensity to obtain real-time environmental sound intensity data. These data are used by the synchronization unit for synchronization signal processing to ensure temporal consistency. Subsequently, the audio codec processing unit encodes the synchronized sound signal and sends it to the main control device through the network unit. The main control device intelligently adjusts the volume of the sound reinforcement matrix unit in combination with the sound source location and task configuration information, creates a 3D audio environment in a specific direction and position, and sends instructions to the output processing module through the network unit to achieve coordinated synchronization of music and light performances, thereby providing the audience with an immersive and interactive display experience under different environmental sound conditions.

[0031] In a second aspect, the present application provides an indoor interactive display device. The indoor interactive display device includes an acquisition module and an output module;

[0032] The acquisition module is used to obtain the location information of each sound source and each real-time ambient sound intensity data in the detection module of each area in the room;

[0033] The output module adjusts the output of each sound reinforcement matrix unit in the area corresponding to each sound source position information according to the position information of each sound source, so that the output volume of each sound reinforcement matrix unit decreases step by step from the installation position of each interactive projection unit to both sides in a ladder-like manner;

[0034] According to the real-time environmental sound intensity data, the output volume of each sound reinforcement matrix unit in the area corresponding to each sound source position information is dynamically adjusted, so that each sound reinforcement matrix unit controls the output volume according to the real-time environmental sound intensity data.

[0035] This device uses two modules to divide the work and coordinate work to better provide users with an interactive experience in the indoor environment. This application obtains the sound source location information through the detection module, and adjusts the output volume of the sound reinforcement matrix unit based on the sound source location information, creating a ladder-shaped volume distribution that decreases from the projection unit to both sides, creating an immersive auditory environment. In addition, this application also automatically adjusts the volume by dynamically analyzing the real-time environmental sound intensity data to ensure that visitors can get a clear auditory experience in different environments. This application not only optimizes the audience's interactive experience, but also improves the adaptability and efficiency of the display environment through intelligent adjustment, solving the problem that the indoor interactive technology in the existing technology cannot meet user needs.

[0036] As a preferred embodiment of the second aspect, the acquisition of each sound source position information and each real-time ambient sound intensity data in the detection module of each indoor area is specifically:

[0037] The detection modules for each indoor area include each microphone array unit and each signal processing unit;

[0038] Each microphone array unit is used to collect each sound signal in the room and transmit each sound signal to each signal processing unit;

[0039] Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain position information of each sound source;

[0040] Each of the signal processing units calculates and obtains each of the real-time environmental sound intensity data by analyzing the intensity changes of each of the sound signals.

[0041] As a preferred embodiment of the second aspect, each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain each sound source position information; each signal processing unit calculates and obtains each real-time environmental sound intensity data by analyzing the intensity change of each sound signal, specifically:

[0042] According to the time difference positioning method, each signal processing unit determines the position of each sound source of each sound signal in combination with the geometric layout of the microphone array in each microphone array unit to obtain the position information of each sound source;

[0043] Each of the signal processing units continuously monitors the strength and quality of each of the sound signals to obtain each real-time monitoring information;

[0044] According to the various real-time monitoring information, various signal processing parameters are dynamically adjusted to obtain various real-time environmental sound intensity data.

[0045] In this preferred embodiment, the present application equips the detection module in each area with a microphone array unit, which is responsible for collecting the sound signals in the area in real time and quickly transmitting these signals to the corresponding signal processing unit. The signal processing unit uses the sound source localization algorithm to process the sound signal, accurately identify and determine the specific location of each sound source, and provide key spatial information. At the same time, these units also analyze the intensity changes of the sound signal and calculate the environmental sound intensity data of each area in real time. This meticulous regional processing method can customize the response to the sound environment in different areas, thereby optimizing the sound output and enhancing the overall auditory experience. This regionalized sound data collection and processing not only enhances the accuracy of sound source localization, but also improves the sensitivity and response speed to changes in environmental sound, providing a technical basis for creating a highly interactive and immersive display environment.

[0046] As a preferred embodiment of the second aspect, it also includes identifying and activating the interactive projection units in each area of ​​the room according to the location information of each sound source and the preset task configuration information, so that the interactive projection units in each area of ​​the room display content related to the interaction with tourists.

[0047] In this preferred embodiment, the present application uses widely distributed microphone array units to monitor the indoor sound environment in real time and obtain sound source location information. Then, the signal processing unit analyzes the collected sound signal to determine the specific location of the sound source. At the same time, the intensity change of the sound signal is also analyzed to obtain real-time environmental sound intensity data. These data are used to compare with the preset task configuration information. Once a matching sound source or sound intensity condition is detected, the system triggers the interactive projection unit in the corresponding area to display customized content. This method not only improves the interactivity and pertinence of the displayed content, but also enhances the visitors' sense of participation and quality of experience.

[0048] As a preferred embodiment of the second aspect, it also includes synchronously outputting the operation of the lighting units in each area of ​​the room and the sound reinforcement matrix units in each area of ​​the room according to the real-time ambient sound intensity data, so that the music output by the sound reinforcement matrix units in each area of ​​the room is coordinated with the performance output by the lighting units in each area of ​​the room.

[0049] In this preferred embodiment, the present application captures indoor sound through a microphone array unit and transmits it to a signal processing unit for preprocessing, then applies a sound source localization algorithm to determine the location of the sound source, and then analyzes the change in sound signal intensity to obtain real-time environmental sound intensity data. These data are used by the synchronization unit for synchronization signal processing to ensure temporal consistency. Subsequently, the audio codec processing unit encodes the synchronized sound signal and sends it to the main control device through the network unit. The main control device intelligently adjusts the volume of the sound reinforcement matrix unit in combination with the sound source location and task configuration information, creates a 3D audio environment in a specific direction and position, and sends instructions to the output processing module through the network unit to achieve coordinated synchronization of music and light performances, thereby providing the audience with an immersive and interactive display experience under different environmental sound conditions. BRIEF DESCRIPTION OF THE DRAWINGS

[0050] Figure 1 : A flow chart of an embodiment of the indoor interactive display method provided by the present application;

[0051] Figure 2 : A schematic diagram of the signal processing flow provided for this application;

[0052] Figure 3 : A schematic diagram of the detection module flow provided for this application;

[0053] Figure 4 : A schematic diagram of the system architecture provided for this application;

[0054] Figure 5 : A structural schematic diagram of an embodiment of the indoor interactive display device provided by the present application; DETAILED DESCRIPTION

[0055] The following will be combined with the drawings in the embodiments of the present invention to clearly and completely describe the technical solutions in the embodiments of the present invention. Obviously, the described embodiments are only part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the present invention.

[0056] Embodiment 1

[0057] Please refer to Figure 1 , which is an indoor interactive display method provided by an embodiment of the present invention.

[0058] In this embodiment, the process of the indoor interactive display method in this application is described in detail through steps S01-S02.

[0059] S01: Acquire the location information of each sound source and each real-time ambient sound intensity data in the detection modules of each area of ​​the room;

[0060] More specifically, the acquisition of each sound source position information and each real-time ambient sound intensity data in the detection modules of each indoor area is specifically as follows:

[0061] The detection modules for each indoor area include each microphone array unit and each signal processing unit;

[0062] Each microphone array unit is used to collect each sound signal in the room and transmit each sound signal to each signal processing unit;

[0063] Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain position information of each sound source;

[0064] Each of the signal processing units calculates and obtains each of the real-time environmental sound intensity data by analyzing the intensity changes of each of the sound signals.

[0065] This application equips the detection module in each area with a microphone array unit, which is responsible for collecting the sound signals in the area in real time and quickly transmitting these signals to the corresponding signal processing unit. The signal processing unit uses the sound source localization algorithm to process the sound signal, accurately identify and determine the specific location of each sound source, and provide key spatial information. At the same time, these units also analyze the intensity changes of the sound signal and calculate the ambient sound intensity data of each area in real time. This meticulous regional processing method can provide customized responses to the sound environment in different areas, thereby optimizing the sound output and enhancing the overall auditory experience. This regionalized sound data collection and processing not only enhances the accuracy of sound source positioning, but also improves the sensitivity and response speed to changes in environmental sound, providing a technical basis for creating a highly interactive and immersive display environment.

[0066] More specifically, the detection module is composed of multiple detection modules, and each detection module is respectively composed of a microphone array unit, a signal processing unit, a synchronization unit, an audio codec processing unit, a wireless audio receiving unit and a network unit. The microphone array unit is mainly a microphone array arranged in a point array and the number of microphones varies according to different accuracy requirements and different application scenarios. The signal processing unit is mainly used to process the data collected by the microphone array. The signal processing includes: input stage, preprocessing stage, echo cancellation stage, sound source localization stage, signal processing and feature extraction stage, optimization and feedback control stage, output stage and real-time monitoring and adjustment stage. For details, please refer to the following. Figure 2As shown; wherein, the input stage mainly transfers the sound signals captured by the microphone array, including the direct sound source (such as the sound of a person speaking) and the environmental sound (including the sound output by the sound reinforcement equipment) into the input buffer; the preprocessing stage mainly uses a filter (such as a bandpass filter, a low-pass filter) to remove the noise in the environment from the sound signal in the input buffer, and then adjusts the gain of the signal to ensure that the sound intensity is within an acceptable range; the echo cancellation stage mainly uses an adaptive filter (such as LMS, NLMS) to estimate the echo signal output by the sound reinforcement equipment in real time. The signal is removed from the signal source, and then an echo path model from the sound reinforcement equipment to the microphone is established to predict and eliminate the echo. At the same time, it is detected whether there is double talk (direct sound source and echo exist at the same time), and the echo cancellation algorithm is adjusted to prevent misoperation. The sound source localization stage mainly calculates the time difference of the sound signal reaching different microphones after the sound signal is echo-cancelled (such as using the generalized cross-correlation method), and combines the microphone array coordinates (the coordinates in the coordinate system constructed after selecting a certain position as the origin of the coordinate when the equipment is deployed) to determine the direction, deflection, distance and coordinates of the sound source, and then uses beamforming technology (such as De The first stage of the signal processing and feature extraction is to analyze the sound signal processed by the sound source localization stage, extract the frequency characteristics of the sound, and then analyze the energy distribution of the signal to distinguish the human voice from the background sound, and then use the short-time Fourier transform (STFT) and other methods to analyze the distribution characteristics of the signal in time and frequency; the first stage of the optimization and feedback control is to detect the feedback (such as howling) in the system of the sound signal processed by the signal processing and feature extraction stage, and eliminate the feedback by adaptive filtering or feedback suppression technology, and then balance the signal according to the indoor acoustic characteristics to ensure the natural and clear sound. At the same time, the data is detected in real time, and the system parameters (such as gain and filter parameters) are adjusted dynamically to optimize the sound quality; the output stage is to output the signal processed by the optimization and feedback control stage to the next unit for processing; the real-time monitoring and adjustment stage mainly adjusts the system parameters through the feedback loop to ensure stability and high-quality output; the synchronization unit mainly performs signal synchronization processing on the audio data of multiple channels on the microphone array and the audio data received by the wireless audio receiving module. The audio codec processing unit is mainly used to encode the audio data processed by the synchronization unit. The wireless audio receiving unit is mainly used to receive the sound signal collected by the wireless audio module. The network unit is mainly used to package the audio data encoded by the audio codec processing unit and carrying the sound source location information and synchronization information and send them to the main control device.

[0067] More specifically, the sound signal input source mainly comes from the microphone array unit of the detection module and the wireless audio receiving unit. The input signal of the wireless audio receiving unit comes from the wireless audio acquisition unit of the wireless audio module. The signal direction in the scenario without a wireless microphone is as follows: the microphone array unit collects the sound signal in the environment (including the direct sound source and the echo), and then the sound signal is transmitted to the signal processing unit. After the sound signal is processed, the location information of the sound source will be determined according to the multi-channel audio data transmitted by the microphone array. Then the sound signal is processed by the synchronization unit for synchronization signal processing, and then the sound signal is transmitted to the audio codec processing unit for audio encoding. The encoded audio data will carry the sound source location information and synchronization information and be packaged together, and finally sent to the main control device via the network unit.

[0068] In the scenario with a wireless microphone, the signal flow is as follows: the microphone array unit collects sound signals in the environment (including direct sound sources and echoes), and at the same time the wireless audio receiving unit also receives the audio data sent by the wireless audio sending unit of the wireless audio module. These two audio data will be transmitted to the signal processing unit at the same time, where the signal transmitted from the wireless audio receiving unit will back up a copy of the audio data as a reference signal for the microphone array incoming signal, and the other copy will be processed. After the two processed audio data, one will determine the location information of the sound source based on the multi-channel audio data transmitted by the microphone array and then be processed by the synchronization unit for synchronization signal processing, and the other will be processed by the signal processing unit and also transmitted to the synchronization unit for synchronization signal processing. The sound signal is then transmitted to the audio codec processing unit for audio encoding. The encoded audio data will carry the sound source location information and synchronization information and be packaged together, and finally sent to the main control device via the network unit. The process is as follows Figure 3 shown.

[0069] In this application, in a wireless microphone scenario, the microphone array unit of the detection module works together with the wireless audio receiving unit to collect and receive audio data. The wireless audio receiving unit copies the received audio data, retains one copy as a backup, and fuses the other copy with the signal collected by the microphone array unit to form a fused sound signal, which enhances the stability and reliability of the signal. This fused signal is then sent to the signal processing unit for further processing. In the wired microphone scenario, the microphone array unit directly collects the sound signal and transmits it to the signal processing unit, ensuring the directness and real-time nature of the sound data. This method not only improves the flexibility and accuracy of sound signal processing, but also provides an adaptive solution for sound collection in different scenarios, optimizing the quality of the sound signal and the experience of interactive display.

[0070] More specifically, the system structure of the indoor interactive display method of the present application includes, in addition to the detection module, a wireless audio module, a main control device and an output processing module, such as Figure 4 shown.

[0071] The wireless audio module includes a wireless audio acquisition unit and a wireless audio transmission unit. The wireless audio acquisition unit mainly collects sound sources from wireless devices such as wireless microphones. The wireless audio transmission unit mainly modulates the audio collected by the wireless audio acquisition unit and then packages it for transmission.

[0072] The main control device includes a scheduling unit, a DSP processing unit, an audio and video codec processing unit, a storage unit and a network unit. The scheduling unit is mainly responsible for unified scheduling according to the task configuration information set by the user. The DSP processing unit mainly filters, enhances, encodes, decodes and processes the audio and video signals through various algorithms and technologies to improve the signal quality, reduce noise, increase clarity, and adapt to different output units. The audio and video codec processing unit is mainly used to encode and decode the audio and video processed by the DSP processing unit; the storage unit is mainly a storage medium, which mainly stores preset audio and video and other related media files. The network unit mainly receives and sends audio and video data packets that need to be processed.

[0073] The output processing module includes an interactive projection unit, a lighting unit, a large-screen display unit, a sound reinforcement matrix unit, an audio and video codec processing unit, and a network unit. The interactive projection unit is mainly various types of projection equipment, including terminal equipment that applies augmented reality (AR) and virtual reality (VR) technology, for outputting video-related media files. The lighting unit is mainly various types of lighting equipment used to create lights for different scene atmospheres. The large-screen display unit is mainly various types of large-screen display equipment for outputting video-related media files. The sound reinforcement matrix unit mainly refers to sound reinforcement equipment arranged in a certain array. The sound reinforcement equipment will push the task configuration information content issued by the scheduling unit of the main control device to the corresponding sound reinforcement array, wherein the sound reinforcement array relies on signal processing technology to create a 3D audio environment when outputting, so that the sound is amplified on the sound reinforcement array in a specific direction and position, and the volume decreases from the center to the sides in a trapezoidal shape to enhance the sense of immersion. The audio and video codec unit is mainly used to encode and decode the audio and video sent by the host device. The network unit mainly receives and sends audio and video data packets that need to be processed. Among them, the received audio and video data will be pushed to the corresponding output unit according to the task configuration information content issued by the scheduling unit of the main control device. The output unit mainly includes an interactive projection unit, a lighting unit, a large-screen display unit and a sound reinforcement matrix unit.

[0074] S02: According to the sound source position information, adjust the output of each sound reinforcement matrix unit in the area corresponding to each sound source position information, so that the output volume of each sound reinforcement matrix unit decreases step by step from the installation position of each interactive projection unit to both sides;

[0075] According to the real-time environmental sound intensity data, the output volume of each sound reinforcement matrix unit in the area corresponding to each sound source position information is dynamically adjusted, so that each sound reinforcement matrix unit controls the output volume according to the real-time environmental sound intensity data.

[0076] This application obtains the sound source location information through the detection module, and adjusts the output volume of the sound reinforcement matrix unit based on the sound source location information, creating a ladder-shaped volume distribution that decreases from the projection unit to both sides, creating an immersive auditory environment. In addition, this application also automatically adjusts the volume by dynamically analyzing the real-time environmental sound intensity data to ensure that visitors can get a clear auditory experience in different environments. This application not only optimizes the audience's interactive experience, but also improves the adaptability and efficiency of the display environment through intelligent adjustment, solving the problem that the indoor interactive technology in the existing technology cannot meet user needs.

[0077] According to the position information of each sound source and the preset task configuration information, the interactive projection units in each area of ​​the room are identified and activated, so that the interactive projection units in each area of ​​the room display content related to the interaction with the visitors.

[0078] This application uses widely distributed microphone array units to monitor the indoor sound environment in real time and obtain sound source location information. Then, the signal processing unit analyzes the collected sound signal to determine the specific location of the sound source. At the same time, the intensity changes of the sound signal are analyzed to obtain real-time environmental sound intensity data. These data are used to compare with the preset task configuration information. Once a matching sound source or sound intensity condition is detected, the system triggers the interactive projection unit in the corresponding area to display customized content. This method not only improves the interactivity and pertinence of the display content, but also enhances the visitors' sense of participation and quality of experience.

[0079] According to the various real-time environmental sound intensity data, the operations of the lighting units in various areas of the room and the sound reinforcement matrix units in various areas of the room are output synchronously, so that the music output by the sound reinforcement matrix units in various areas of the room is coordinated with the performances output by the lighting units in various areas of the room.

[0080] This application captures indoor sound through a microphone array unit and transmits it to a signal processing unit for preprocessing, then applies a sound source localization algorithm to determine the location of the sound source, and then analyzes the change in sound signal intensity to obtain real-time environmental sound intensity data. These data are used by the synchronization unit for synchronization signal processing to ensure temporal consistency. Subsequently, the audio codec processing unit encodes the synchronized sound signal and sends it to the main control unit through the network unit. The main control unit intelligently adjusts the volume of the sound reinforcement matrix unit based on the sound source location and task configuration information, creates a 3D audio environment in a specific direction and position, and sends instructions to the output processing module through the network unit to achieve coordinated synchronization of music and light performances, thereby providing the audience with an immersive and interactive display experience under different environmental sound conditions.

[0081] Among them, the signal flow direction of the four functions in the implementation process is specifically as follows: the microphone array unit collects the sound signal (including direct sound source and echo) in the environment in real time, and the wireless audio receiving unit also waits to receive the audio data sent by the wireless audio sending unit of the wireless audio module. When there is a sound signal input, the audio data will be transmitted to the signal processing unit, wherein the signal transmitted from the wireless audio receiving unit will back up a copy of the audio data as a reference signal of the microphone array incoming signal, and the other will be processed. After the two processed audio data, one will determine the location information of the sound source according to the multi-channel audio data transmitted by the microphone array and then be processed by the synchronization unit for synchronization signal processing, and the other will be processed by the signal processing unit and also transmitted to the synchronization unit for synchronization signal processing. Then the sound signal is transmitted to the audio codec processing unit for audio encoding, and the encoded audio data will carry the sound source location information and synchronization information and be packaged together and sent to the main control device via the network unit. The main control device will package according to the location information of the sound source combined with the task configuration information of the scheduling unit and then send it to the output processing module through the network unit. The output processing module will parse the data packets sent by the main control device, extract audio and video related data to the audio and video codec processing unit, and then output the audio and video data to the corresponding interactive projection unit, lighting unit, large-screen display unit or sound reinforcement matrix unit according to the task configuration information.

[0082] As an embodiment, in an application example, the detection modules are arranged in different areas of the exhibition hall, and the number of detection modules in each area can be one or more. At the same time, the coordinates of each detection module and the sound reinforcement matrix unit in the coordinate system generated by a certain area of ​​the exhibition hall as the coordinate origin will be recorded during deployment. The microphone array arrangement in the detection module can be a linear array, a circular array or other geometric shapes. The sound signal captured by each microphone includes a direct sound source (such as a person's speech) and an ambient sound (including the sound output by the speaker). When deployed in the exhibition hall area, the detection unit and the interactive projection unit, lighting unit and large-screen display unit of the output processing module are used in conjunction with the sound reinforcement matrix unit.

[0083] For example, the projection explanation area in the exhibition hall is equipped with the detection module, interactive projection unit and sound reinforcement matrix unit. In actual applications, the interactive projection unit includes relevant equipment implemented by applying augmented reality (AR) and virtual reality (VR) technology and equipment related to the collection recognition technology. When visitors interact, the area will project relevant content. At the same time, the volume output of the sound reinforcement matrix unit is stepped down step by step on both sides with the installation position of the projection unit as the center according to the known position information, so as to achieve an immersive experience. At the same time, the detection unit will detect the ambient sound of the area in real time and then dynamically adjust the output volume of the sound reinforcement matrix unit to increase intelligence and fun. Similarly, in the area where the lighting system or other lighting-related equipment is displayed, the lighting unit will be used in combination with the sound reinforcement matrix unit and the detection module. When performing a lighting show, the sound reinforcement matrix unit provides the output of rhythmic music. At the same time, the detection module in the audience viewing area detects the ambient sound of the viewing area in real time, increases the output sound of the sound reinforcement matrix unit in a noisy environment, and reduces the sound output of the sound reinforcement matrix unit in a quiet environment.

[0084] This application obtains the sound source location information through the detection module, and adjusts the output volume of the sound reinforcement matrix unit based on the sound source location information, creating a ladder-shaped volume distribution that decreases from the projection unit to both sides, creating an immersive auditory environment. In addition, this application also automatically adjusts the volume by dynamically analyzing the real-time environmental sound intensity data to ensure that visitors can get a clear auditory experience in different environments. This application not only optimizes the audience's interactive experience, but also improves the adaptability and efficiency of the display environment through intelligent adjustment, solving the problem that the indoor interactive technology in the existing technology cannot meet user needs.

[0085] Embodiment 2

[0086] Please refer to Figure 5 , which is an indoor interactive display device provided in an embodiment of the present application.

[0087] In this embodiment, the indoor interactive display device includes an acquisition module 10 and an output module 20 .

[0088] The acquisition module 10 is used to obtain the location information of each sound source and each real-time ambient sound intensity data in the detection modules of each area of ​​the room;

[0089] More specifically, the acquisition of each sound source position information and each real-time ambient sound intensity data in the detection modules of each indoor area is specifically as follows:

[0090] The detection modules for each indoor area include each microphone array unit and each signal processing unit;

[0091] Each microphone array unit is used to collect each sound signal in the room and transmit each sound signal to each signal processing unit;

[0092] Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain position information of each sound source;

[0093] Each of the signal processing units calculates and obtains each of the real-time environmental sound intensity data by analyzing the intensity changes of each of the sound signals.

[0094] This application equips the detection module in each area with a microphone array unit, which is responsible for collecting the sound signals in the area in real time and quickly transmitting these signals to the corresponding signal processing unit. The signal processing unit uses the sound source localization algorithm to process the sound signal, accurately identify and determine the specific location of each sound source, and provide key spatial information. At the same time, these units also analyze the intensity changes of the sound signal and calculate the ambient sound intensity data of each area in real time. This meticulous regional processing method can provide customized responses to the sound environment in different areas, thereby optimizing the sound output and enhancing the overall auditory experience. This regionalized sound data collection and processing not only enhances the accuracy of sound source positioning, but also improves the sensitivity and response speed to changes in environmental sound, providing a technical basis for creating a highly interactive and immersive display environment.

[0095] More specifically, the detection module is composed of multiple detection modules, and each detection module is respectively composed of a microphone array unit, a signal processing unit, a synchronization unit, an audio codec processing unit, a wireless audio receiving unit and a network unit. The microphone array unit is mainly a microphone array arranged in a point array and the number of microphones varies according to different accuracy requirements and different application scenarios. The signal processing unit is mainly used to process the data collected by the microphone array. The signal processing includes: input stage, preprocessing stage, echo cancellation stage, sound source localization stage, signal processing and feature extraction stage, optimization and feedback control stage, output stage and real-time monitoring and adjustment stage. For details, please refer to the following. Figure 2As shown; wherein, the input stage mainly transfers the sound signals captured by the microphone array, including the direct sound source (such as the sound of a person speaking) and the environmental sound (including the sound output by the sound reinforcement equipment) into the input buffer; the preprocessing stage mainly uses a filter (such as a bandpass filter, a low-pass filter) to remove the noise in the environment from the sound signal in the input buffer, and then adjusts the gain of the signal to ensure that the sound intensity is within an acceptable range; the echo cancellation stage mainly uses an adaptive filter (such as LMS, NLMS) to estimate the echo signal output by the sound reinforcement equipment in real time. The signal is removed from the signal source, and then an echo path model from the sound reinforcement equipment to the microphone is established to predict and eliminate the echo. At the same time, it is detected whether there is double talk (direct sound source and echo exist at the same time), and the echo cancellation algorithm is adjusted to prevent misoperation. The sound source localization stage mainly calculates the time difference of the sound signal reaching different microphones after the sound signal is echo-cancelled (such as using the generalized cross-correlation method), and combines the microphone array coordinates (the coordinates in the coordinate system constructed after selecting a certain position as the origin of the coordinate when the equipment is deployed) to determine the direction, deflection, distance and coordinates of the sound source, and then uses beamforming technology (such as De The first stage of the signal processing and feature extraction is to analyze the sound signal processed by the sound source localization stage, extract the frequency characteristics of the sound, and then analyze the energy distribution of the signal to distinguish the human voice from the background sound, and then use the short-time Fourier transform (STFT) and other methods to analyze the distribution characteristics of the signal in time and frequency; the first stage of the optimization and feedback control is to detect the feedback (such as howling) in the system of the sound signal processed by the signal processing and feature extraction stage, and eliminate the feedback by adaptive filtering or feedback suppression technology, and then balance the signal according to the indoor acoustic characteristics to ensure the natural and clear sound. At the same time, the data is detected in real time, and the system parameters (such as gain and filter parameters) are adjusted dynamically to optimize the sound quality; the output stage is to output the signal processed by the optimization and feedback control stage to the next unit for processing; the real-time monitoring and adjustment stage mainly adjusts the system parameters through the feedback loop to ensure stability and high-quality output; the synchronization unit mainly performs signal synchronization processing on the audio data of multiple channels on the microphone array and the audio data received by the wireless audio receiving module. The audio codec processing unit is mainly used to encode the audio data processed by the synchronization unit. The wireless audio receiving unit is mainly used to receive the sound signal collected by the wireless audio module. The network unit is mainly used to package the audio data encoded by the audio codec processing unit and carrying the sound source location information and synchronization information and send them to the main control device.

[0096] More specifically, the sound signal input source mainly comes from the microphone array unit of the detection module and the wireless audio receiving unit. The input signal of the wireless audio receiving unit comes from the wireless audio acquisition unit of the wireless audio module. The signal direction in the scenario without a wireless microphone is as follows: the microphone array unit collects the sound signal in the environment (including the direct sound source and the echo), and then the sound signal is transmitted to the signal processing unit. After the sound signal is processed, the location information of the sound source will be determined according to the multi-channel audio data transmitted by the microphone array. Then the sound signal is processed by the synchronization unit for synchronization signal processing, and then the sound signal is transmitted to the audio codec processing unit for audio encoding. The encoded audio data will carry the sound source location information and synchronization information and be packaged together, and finally sent to the main control device via the network unit.

[0097] In the scenario with a wireless microphone, the signal flow is as follows: the microphone array unit collects sound signals in the environment (including direct sound sources and echoes), and at the same time the wireless audio receiving unit also receives the audio data sent by the wireless audio sending unit of the wireless audio module. These two audio data will be transmitted to the signal processing unit at the same time, where the signal transmitted from the wireless audio receiving unit will back up a copy of the audio data as a reference signal for the microphone array incoming signal, and the other copy will be processed. After the two processed audio data, one will determine the location information of the sound source based on the multi-channel audio data transmitted by the microphone array and then be processed by the synchronization unit for synchronization signal processing, and the other will be processed by the signal processing unit and also transmitted to the synchronization unit for synchronization signal processing. The sound signal is then transmitted to the audio codec processing unit for audio encoding. The encoded audio data will carry the sound source location information and synchronization information and be packaged together, and finally sent to the main control device via the network unit. The process is as follows Figure 3 shown.

[0098] In this application, in a wireless microphone scenario, the microphone array unit of the detection module works together with the wireless audio receiving unit to collect and receive audio data. The wireless audio receiving unit copies the received audio data, retains one copy as a backup, and fuses the other copy with the signal collected by the microphone array unit to form a fused sound signal, which enhances the stability and reliability of the signal. This fused signal is then sent to the signal processing unit for further processing. In the wired microphone scenario, the microphone array unit directly collects the sound signal and transmits it to the signal processing unit, ensuring the directness and real-time nature of the sound data. This method not only improves the flexibility and accuracy of sound signal processing, but also provides an adaptive solution for sound collection in different scenarios, optimizing the quality of the sound signal and the experience of interactive display.

[0099] More specifically, the system structure of the indoor interactive display method of the present application includes, in addition to the detection module, a wireless audio module, a main control device and an output processing module, such as Figure 4 shown.

[0100] The wireless audio module includes a wireless audio acquisition unit and a wireless audio transmission unit. The wireless audio acquisition unit mainly collects sound sources from wireless devices such as wireless microphones. The wireless audio transmission unit mainly modulates the audio collected by the wireless audio acquisition unit and then packages it for transmission.

[0101] The main control device includes a scheduling unit, a DSP processing unit, an audio and video codec processing unit, a storage unit and a network unit. The scheduling unit is mainly responsible for unified scheduling according to the task configuration information set by the user. The DSP processing unit mainly filters, enhances, encodes, decodes and processes the audio and video signals through various algorithms and technologies to improve the signal quality, reduce noise, increase clarity, and adapt to different output units. The audio and video codec processing unit is mainly used to encode and decode the audio and video processed by the DSP processing unit; the storage unit is mainly a storage medium, which mainly stores preset audio and video and other related media files. The network unit mainly receives and sends audio and video data packets that need to be processed.

[0102] The output processing module includes an interactive projection unit, a lighting unit, a large-screen display unit, a sound reinforcement matrix unit, an audio and video codec processing unit, and a network unit. The interactive projection unit is mainly various types of projection equipment, including terminal equipment that applies augmented reality (AR) and virtual reality (VR) technology, for outputting video-related media files. The lighting unit is mainly various types of lighting equipment used to create lights for different scene atmospheres. The large-screen display unit is mainly various types of large-screen display equipment for outputting video-related media files. The sound reinforcement matrix unit mainly refers to sound reinforcement equipment arranged in a certain array. The sound reinforcement equipment will push the task configuration information content issued by the scheduling unit of the main control device to the corresponding sound reinforcement array, wherein the sound reinforcement array relies on signal processing technology to create a 3D audio environment when outputting, so that the sound is amplified on the sound reinforcement array in a specific direction and position, and the volume decreases from the center to the sides in a trapezoidal shape to enhance the sense of immersion. The audio and video codec unit is mainly used to encode and decode the audio and video sent by the host device. The network unit mainly receives and sends audio and video data packets that need to be processed. Among them, the received audio and video data will be pushed to the corresponding output unit according to the task configuration information content issued by the scheduling unit of the main control device. The output unit mainly includes an interactive projection unit, a lighting unit, a large-screen display unit and a sound reinforcement matrix unit.

[0103] The output module 20 is used to adjust the output of each sound reinforcement matrix unit in the area corresponding to each sound source position information according to the position information of each sound source, so that the output volume of each sound reinforcement matrix unit decreases step by step from the installation position of each interactive projection unit to both sides in a ladder-like manner;

[0104] According to the real-time environmental sound intensity data, the output volume of each sound reinforcement matrix unit in the area corresponding to each sound source position information is dynamically adjusted, so that each sound reinforcement matrix unit controls the output volume according to the real-time environmental sound intensity data.

[0105] This application obtains the sound source location information through the detection module, and adjusts the output volume of the sound reinforcement matrix unit based on the sound source location information, creating a ladder-shaped volume distribution that decreases from the projection unit to both sides, creating an immersive auditory environment. In addition, this application also automatically adjusts the volume by dynamically analyzing the real-time environmental sound intensity data to ensure that visitors can get a clear auditory experience in different environments. This application not only optimizes the audience's interactive experience, but also improves the adaptability and efficiency of the display environment through intelligent adjustment, solving the problem that the indoor interactive technology in the existing technology cannot meet user needs.

[0106] According to the position information of each sound source and the preset task configuration information, the interactive projection units in each area of ​​the room are identified and activated, so that the interactive projection units in each area of ​​the room display content related to the interaction with the visitors.

[0107] This application uses widely distributed microphone array units to monitor the indoor sound environment in real time and obtain sound source location information. Then, the signal processing unit analyzes the collected sound signal to determine the specific location of the sound source. At the same time, the intensity changes of the sound signal are analyzed to obtain real-time environmental sound intensity data. These data are used to compare with the preset task configuration information. Once a matching sound source or sound intensity condition is detected, the system triggers the interactive projection unit in the corresponding area to display customized content. This method not only improves the interactivity and pertinence of the display content, but also enhances the visitors' sense of participation and quality of experience.

[0108] According to the various real-time environmental sound intensity data, the operations of the lighting units in various areas of the room and the sound reinforcement matrix units in various areas of the room are output synchronously, so that the music output by the sound reinforcement matrix units in various areas of the room is coordinated with the performances output by the lighting units in various areas of the room.

[0109] This application captures indoor sound through a microphone array unit and transmits it to a signal processing unit for preprocessing, then applies a sound source localization algorithm to determine the location of the sound source, and then analyzes the change in sound signal intensity to obtain real-time environmental sound intensity data. These data are used by the synchronization unit for synchronization signal processing to ensure temporal consistency. Subsequently, the audio codec processing unit encodes the synchronized sound signal and sends it to the main control unit through the network unit. The main control unit intelligently adjusts the volume of the sound reinforcement matrix unit based on the sound source location and task configuration information, creates a 3D audio environment in a specific direction and position, and sends instructions to the output processing module through the network unit to achieve coordinated synchronization of music and light performances, thereby providing the audience with an immersive and interactive display experience under different environmental sound conditions.

[0110] Among them, the signal flow direction of the four functions in the implementation process is specifically as follows: the microphone array unit collects the sound signal (including direct sound source and echo) in the environment in real time, and the wireless audio receiving unit also waits to receive the audio data sent by the wireless audio sending unit of the wireless audio module. When there is a sound signal input, the audio data will be transmitted to the signal processing unit, wherein the signal transmitted from the wireless audio receiving unit will back up a copy of the audio data as a reference signal of the microphone array incoming signal, and the other will be processed. After the two processed audio data, one will determine the location information of the sound source according to the multi-channel audio data transmitted by the microphone array and then be processed by the synchronization unit for synchronization signal processing, and the other will be processed by the signal processing unit and also transmitted to the synchronization unit for synchronization signal processing. Then the sound signal is transmitted to the audio codec processing unit for audio encoding, and the encoded audio data will carry the sound source location information and synchronization information and be packaged together and sent to the main control device via the network unit. The main control device will package according to the location information of the sound source combined with the task configuration information of the scheduling unit and then send it to the output processing module through the network unit. The output processing module will parse the data packets sent by the main control device, extract audio and video related data to the audio and video codec processing unit, and then output the audio and video data to the corresponding interactive projection unit, lighting unit, large-screen display unit or sound reinforcement matrix unit according to the task configuration information.

[0111] As an embodiment, in an application example, the detection modules are arranged in different areas of the exhibition hall, and the number of detection modules in each area can be one or more. At the same time, the coordinates of each detection module and the sound reinforcement matrix unit in the coordinate system generated by a certain area of ​​the exhibition hall as the coordinate origin will be recorded during deployment. The microphone array arrangement in the detection module can be a linear array, a circular array or other geometric shapes. The sound signal captured by each microphone includes a direct sound source (such as a person's speech) and an ambient sound (including the sound output by the speaker). When deployed in the exhibition hall area, the detection unit and the interactive projection unit, lighting unit and large-screen display unit of the output processing module are used in conjunction with the sound reinforcement matrix unit.

[0112] For example, the projection explanation area in the exhibition hall is equipped with the detection module, interactive projection unit and sound reinforcement matrix unit. In actual applications, the interactive projection unit includes relevant equipment implemented by applying augmented reality (AR) and virtual reality (VR) technology and equipment related to the collection recognition technology. When visitors interact, the area will project relevant content. At the same time, the volume output of the sound reinforcement matrix unit is stepped down step by step on both sides with the installation position of the projection unit as the center according to the known position information, so as to achieve an immersive experience. At the same time, the detection unit will detect the ambient sound of the area in real time and then dynamically adjust the output volume of the sound reinforcement matrix unit to increase intelligence and fun. Similarly, in the area where the lighting system or other lighting-related equipment is displayed, the lighting unit will be used in combination with the sound reinforcement matrix unit and the detection module. When performing a lighting show, the sound reinforcement matrix unit provides the output of rhythmic music. At the same time, the detection module in the audience viewing area detects the ambient sound of the viewing area in real time, increases the output sound of the sound reinforcement matrix unit in a noisy environment, and reduces the output sound of the sound reinforcement matrix unit in a quiet environment.

[0113] This device uses two modules to divide the work and coordinate work to better provide users with an interactive experience in the indoor environment. This application obtains the sound source location information through the detection module, and adjusts the output volume of the sound reinforcement matrix unit based on the sound source location information, creating a ladder-shaped volume distribution that decreases from the projection unit to both sides, creating an immersive auditory environment. In addition, this application also automatically adjusts the volume by dynamically analyzing the real-time environmental sound intensity data to ensure that visitors can get a clear auditory experience in different environments. This application not only optimizes the audience's interactive experience, but also improves the adaptability and efficiency of the display environment through intelligent adjustment, solving the problem that the indoor interactive technology in the existing technology cannot meet user needs.

[0114] The specific embodiments described above further illustrate the purpose, technical solutions and beneficial effects of the present invention. It should be understood that the above description is only a specific embodiment of the present invention and is not intended to limit the scope of protection of the present invention. It is particularly pointed out that for those skilled in the art, any modification, equivalent replacement, improvement, etc. made within the spirit and principle of the present invention should be included in the scope of protection of the present invention.

Claims

1. An indoor interactive display method, characterized in that: include: Obtain the location information of each sound source and each real-time ambient sound intensity data in the detection modules of each area of ​​the room; According to the respective sound source position information, the output of each sound reinforcement matrix unit in the area corresponding to each sound source position information is adjusted so that the output volume of each sound reinforcement matrix unit decreases step by step from the installation position of each interactive projection unit to both sides in a ladder-like manner; According to the real-time environmental sound intensity data, dynamically adjust the output volume of each sound reinforcement matrix unit in the area corresponding to each sound source position information, so that each sound reinforcement matrix unit controls the output volume according to the real-time environmental sound intensity data; According to the various real-time environmental sound intensity data, the operations of the lighting units in various areas of the room and the sound reinforcement matrix units in various areas of the room are output synchronously, so that the music output by the sound reinforcement matrix units in various areas of the room is coordinated with the performances output by the lighting units in various areas of the room.

2. The indoor interactive display method according to claim 1, characterized in that: The method of obtaining the location information of each sound source and each real-time ambient sound intensity data in the detection modules of each indoor area is specifically as follows: The detection modules for each indoor area include each microphone array unit and each signal processing unit; Each microphone array unit is used to collect each sound signal in the room and transmit each sound signal to each signal processing unit; Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain position information of each sound source; Each of the signal processing units calculates and obtains each of the real-time environmental sound intensity data by analyzing the intensity changes of each of the sound signals.

3. The indoor interactive display method according to claim 2, characterized in that: Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain each sound source position information; each signal processing unit calculates and obtains each real-time environmental sound intensity data by analyzing the intensity change of each sound signal, specifically: According to the time difference positioning method, each signal processing unit determines the position of each sound source of each sound signal in combination with the geometric layout of the microphone array in each microphone array unit to obtain the position information of each sound source; Each of the signal processing units continuously monitors the strength and quality of each of the sound signals to obtain each real-time monitoring information; According to the various real-time monitoring information, various signal processing parameters are dynamically adjusted to obtain various real-time environmental sound intensity data.

4. The indoor interactive display method according to claim 1, characterized in that: Also includes: According to the position information of each sound source and the preset task configuration information, the interactive projection units in each area of ​​the room are identified and activated, so that the interactive projection units in each area of ​​the room display content related to the interaction with the visitors.

5. An indoor interactive display device, characterized in that: Includes acquisition module and output module; The acquisition module is used to obtain the location information of each sound source and each real-time ambient sound intensity data in the detection module of each area in the room; The output module adjusts the output of each sound reinforcement matrix unit in the area corresponding to each sound source position information according to the position information of each sound source, so that the output volume of each sound reinforcement matrix unit decreases step by step from the installation position of each interactive projection unit to both sides in a ladder-like manner; According to the real-time environmental sound intensity data, dynamically adjust the output volume of each sound reinforcement matrix unit in the area corresponding to each sound source position information, so that each sound reinforcement matrix unit controls the output volume according to the real-time environmental sound intensity data; According to the various real-time environmental sound intensity data, the operations of the lighting units in various areas of the room and the sound reinforcement matrix units in various areas of the room are output synchronously, so that the music output by the sound reinforcement matrix units in various areas of the room is coordinated with the performances output by the lighting units in various areas of the room.

6. The indoor interactive display device according to claim 5, characterized in that: The method of obtaining the location information of each sound source and each real-time ambient sound intensity data in the detection modules of each indoor area is specifically as follows: The detection modules for each indoor area include each microphone array unit and each signal processing unit; Each microphone array unit is used to collect each sound signal in the room and transmit each sound signal to each signal processing unit; Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain position information of each sound source; Each of the signal processing units calculates and obtains each of the real-time environmental sound intensity data by analyzing the intensity changes of each of the sound signals.

7. The indoor interactive display device according to claim 6, characterized in that: Each signal processing unit is used to process each sound signal using a sound source localization algorithm to obtain each sound source position information; each signal processing unit calculates and obtains each real-time environmental sound intensity data by analyzing the intensity change of each sound signal, specifically: According to the time difference positioning method, each signal processing unit determines the position of each sound source of each sound signal in combination with the geometric layout of the microphone array in each microphone array unit to obtain the position information of each sound source; Each of the signal processing units continuously monitors the strength and quality of each of the sound signals to obtain each real-time monitoring information; According to the various real-time monitoring information, various signal processing parameters are dynamically adjusted to obtain various real-time environmental sound intensity data.

8. The indoor interactive display device according to claim 5, characterized in that: Also includes: According to the position information of each sound source and the preset task configuration information, the interactive projection units in each area of ​​the room are identified and activated, so that the interactive projection units in each area of ​​the room display content related to the interaction with the visitors.

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