A network audio integrated machine, an audio signal processing system and method

Abstract

The application discloses an audio signal processing system and method. The system comprises a network audio integrated machine, a microphone array and an audio playing device. The network audio integrated machine is integrated with a switch, a DSP processor and a master control chip. The master control chip is used for controlling the working state of the DSP processor, the microphone array, the switch and the audio playing device. The master control chip is also used for controlling the switch to connect the network audio integrated machine to a preset network, obtaining the IP address of the network audio integrated machine, receiving a service request, responding to the service request, sending a management interface corresponding to the network audio integrated machine, receiving a triggering operation on the management interface, and controlling the DSP processor to configure an audio channel of the network audio integrated machine. The audio channel is used for indicating an audio signal transmission link between the microphone array and the audio playing device. The application solves the complicated problems of wiring, configuration and maintenance of a traditional audio system, and realizes optimization of simple deployment, fast configuration and easy maintenance.

Description

Technical Field

[0001] This application relates to the field of audio processing technology, and in particular to an audio signal processing system and method. Background Technology

[0002] Current network audio devices are deployed in a distributed manner, and after deployment, host computer software needs to be installed in the interface system to configure the product's functions. Current network audio devices suffer from drawbacks such as numerous connections, complex deployment, difficulty in quick debugging and maintenance, and the need for dedicated personnel for maintenance. Summary of the Invention

[0003] In view of this, it is necessary to provide an audio signal processing system and method to solve the problems of existing network audio systems, such as configuration dependence on host computers, complex wiring, and high debugging and maintenance costs.

[0004] To solve the above-mentioned technical problems, one technical solution adopted in this application is as follows: A network audio all-in-one machine is provided. The network audio all-in-one machine is connected to a microphone array and an audio playback device. The network audio all-in-one machine includes a switch, a DSP processor, and a main control chip. The DSP processor is connected to the microphone array, the switch, and the audio playback device respectively. The main control chip is also connected to the DSP processor, the microphone array, the switch, and the audio playback device respectively. The main control chip is used to control the working status of the DSP processor, the microphone array, the switch, and the audio playback device. The main control chip is also used to: receive service requests through a preset network; the service requests are sent by control devices accessing the preset network via IP addresses; respond to the service requests and send a management interface to the control devices, causing the control devices to display the management interface; receive user trigger operations on the management interface sent by the control devices; and configure the audio channels of the network audio all-in-one machine according to the trigger operations; the audio channels are used to indicate the audio signal transmission link between the microphone array and the audio playback device.

[0005] This system integrates a switch, DSP processor, and main control chip into a network audio all-in-one unit, significantly reducing the number and complexity of devices and facilitating management and maintenance. The DSP processor enables real-time audio signal processing, providing gain adjustment, equalization, and effects processing to optimize audio quality and meet the needs of various scenarios. Through the integrated switch, the system can easily connect to a pre-set network, enabling connection with control devices and enhancing system scalability and interoperability. Users can perform local or remote configuration and debugging via a browser without installing additional software, achieving plug-and-play functionality and cross-platform access. The main control chip can dynamically configure audio channels based on user triggers, enabling flexible audio signal transmission links to adapt to different usage requirements. Therefore, this solution solves the complexities of wiring, configuration, and maintenance in traditional audio systems, achieving overall optimization of simple deployment, fast configuration, and easy maintenance. It also improves the quality and flexibility of audio processing, simplifies user interaction and device management, making it suitable for various scenarios and applications.

[0006] In some embodiments, configuring the audio channel of the network audio all-in-one device according to a trigger operation includes: displaying audio stream information corresponding to the preset output audio channel according to a first trigger operation on the management interface, the audio stream information including multiple input nodes and multiple output nodes; wherein the input node is a microphone in a microphone array, and the output node is one of the audio playback devices; receiving a user selection operation on multiple input nodes sent by the control device, configuring the input device corresponding to the preset output audio channel; receiving a user selection operation on multiple output nodes sent by the control device, configuring the output device corresponding to the preset output audio channel. This solution supports multiple input and output nodes, allowing users to freely select input and output devices as needed, enhancing the system's usability and adaptability.

[0007] In some embodiments, after configuring the input device corresponding to the preset output audio channel, the management interface is used to: receive the audio processing algorithm module selected by the user, which performs algorithmic processing on the audio input from the input device. The audio processing algorithm module improves the quality of the audio signal, and users can customize sound effects according to personal preferences or scenario requirements, such as enhancing bass, adjusting high frequencies, applying reverb, etc., thus enhancing the user's audio experience in different situations.

[0008] In some embodiments, the switch includes a network data forwarding unit, a power supply unit, and a network audio device detection unit. The power supply unit is connected to both the network data forwarding unit and the network audio device detection unit, and the network data forwarding unit is connected to the DSP processor. The network audio device detection unit determines the power requirement of the network audio device when it detects a network audio device connected, and controls the power supply unit to provide power to the network audio device. The network data forwarding unit receives, parses, and forwards data packets from the connected network audio device to enable data communication between the connected network audio device and the DSP processor. This solution supports one-wire transmission of both data and power, thereby reducing wiring complexity.

[0009] In some embodiments, the microphone array includes a ceiling microphone array, which is communicatively connected to a DSP processor and a main control chip. The ceiling microphone array is used to collect audio signals in a preset scene and send the audio signals in the preset scene to the DSP processor. The main control chip is used to respond to a pickup area setting request input from the management interface, set the pickup range of the ceiling microphone array, and send the pickup range of the ceiling microphone array to the DSP processor. The DSP processor is used to acquire the audio signals collected within the pickup range of the ceiling microphone array, process the audio signals, and output the processed audio signals to an audio playback device for local playback of the audio signals collected within the pickup range of the ceiling microphone array. This solution can control the pickup range through a ceiling microphone array, allowing speakers to speak clearly without holding a microphone, thus solving the problem of passing microphones in meeting rooms.

[0010] In some embodiments, the microphone array further includes a host microphone, which is communicatively connected to a DSP processor. The host microphone is used to acquire the host's audio signal and send it to the DSP processor. The DSP processor is used to send the host's audio signal to the main control chip. The main control chip is used to: perform sound suppression control on the ceiling microphone array when the host's audio signal is detected; control the ceiling microphone array to resume sound pickup when the host's audio signal is not detected; or, when the host is speaking, respond to a ceiling microphone suppression request from the management interface and perform sound suppression control on the ceiling microphone array; when the host is not speaking, respond to a ceiling microphone enable request from the management interface and control the ceiling microphone array to resume sound pickup. Specifically, when the host is speaking, the system automatically suppresses the ceiling microphone array to prevent the recording of environmental noise or echo, avoid reverberation, howling, and audio interference caused by multi-channel sound pickup, and effectively improve speech clarity. After the host pauses, the ceiling microphone array resumes operation and can normally pick up the speeches of other participants.

[0011] In some embodiments, the system further includes a camera device, which is communicatively connected to the main control chip via a switch. The main control chip is also used to: acquire speaker information collected by the camera device; adjust the pickup range of the ceiling microphone array based on the speaker information; acquire the audio signal collected within the adjusted pickup range of the ceiling microphone array, and send the audio signal to the DSP processor; the DSP processor processes the audio signal and outputs the processed audio signal to an audio playback device for local playback of the audio signal collected within the adjusted pickup range of the ceiling microphone array. This solution implements microphone-camera linkage, enabling the camera device to intelligently track the speaker, and the network audio all-in-one device to adjust the pickup range of the ceiling microphone array based on the tracked speaker, thereby more accurately picking up the speaker's voice and improving voice quality and user experience.

[0012] In some embodiments, the microphone array includes an omnidirectional microphone, which is communicatively connected to a DSP processor via a switch. The omnidirectional microphone is used to acquire audio signals in a preset scenario and send these signals to the DSP processor. The DSP processor processes the audio signals in the preset scenario and outputs the processed audio signals to a remote audio device to achieve remote audio transmission. The DSP processor does not amplify the audio signals in the preset scenario acquired by the omnidirectional microphone through an audio playback device. This solution allows remote participants to clearly hear the on-site sound, improving the communication efficiency of remote audio and video conferencing. Furthermore, audio signals that do not pass through an audio playback device reduce distortion and noise, thus maintaining higher sound quality, which provides remote participants with clearer and more natural audio feedback.

[0013] In some embodiments, the microphone array further includes a host microphone, which is communicatively connected to a DSP processor. The host microphone is used to acquire the host's audio signal and send it to the DSP processor. The DSP processor is also used to: process the host's audio signal; output the processed host's audio signal to a remote audio device for remote audio transmission; and play the processed host's audio signal locally through an audio playback device. During the host's speech, the DSP processor also performs pickup suppression control on the omnidirectional microphone. This solution achieves local playback and remote synchronous transmission of the host's audio signal, balancing on-site listening experience with remote audio interaction, thus improving the integrity and practicality of the conference system. This solution also implements host pickup priority control to avoid audio channel interference, prevent feedback, and improve speech clarity.

[0014] In some embodiments, the main control chip includes a Web control module and a status control module. The Web control module receives service requests and responds to trigger operations on the management interface, while the status control module controls the operating status of the DSP processor, microphone array, switch, and audio playback device. The Web control module allows users to remotely access and manage the devices via the network without physical contact, improving system flexibility and convenience. The status control module facilitates status monitoring and management of each submodule, enhancing system stability and maintainability.

[0015] In some embodiments, receiving a service request and responding to it, returning the management interface corresponding to the network audio all-in-one device, includes: receiving a service request, the service request including a username and password; and responding to the service request and returning the management interface corresponding to the network audio all-in-one device if the username and password meet preset conditions. In this solution, access to the management interface is only possible with a correct username and password, thus improving the security of the audio signal processing system.

[0016] To solve the above-mentioned technical problems, one technical solution adopted in this application is to provide an audio signal processing system, including a microphone array, an audio playback device, and the above-mentioned network audio all-in-one machine.

[0017] To address the aforementioned technical problems, one technical solution adopted in this application is: providing an audio signal processing method applied to the aforementioned audio signal processing system. The audio signal processing system includes a network audio all-in-one unit, a microphone array, and an audio playback device. The network audio all-in-one unit is connected to both the microphone array and the audio playback device. The method includes: receiving a service request through a preset network; the service request is sent by a control device accessing the preset network via an IP address; responding to the service request by sending a management interface to the control device, causing the control device to display the management interface; receiving a user trigger operation on the management interface sent by the control device; configuring the audio channel of the network audio all-in-one unit according to the trigger operation; the audio channel is used to indicate the audio signal transmission link between the microphone array and the audio playback device.

[0018] This method integrates a switch, DSP processor, and main control chip into a network audio all-in-one unit, significantly reducing the number and complexity of devices and facilitating management and maintenance. The DSP processor enables real-time audio signal processing, providing gain adjustment, equalization, and effects processing to optimize audio quality and meet the needs of different scenarios. Through the integrated switch, the system can easily connect to a preset network, enabling connection with control devices and enhancing system scalability and interoperability. Users can perform local or remote configuration and debugging via a browser without installing additional software, achieving plug-and-play functionality and cross-platform access. The main control chip can dynamically configure audio channels based on user triggers, achieving flexible audio signal transmission links to adapt to different usage requirements. Therefore, this solution solves the complexities of wiring, configuration, and maintenance in traditional audio systems, achieving overall optimization of simple deployment, fast configuration, and easy maintenance. It also improves the quality and flexibility of audio processing, simplifies user interaction and device management, making it suitable for various scenarios and applications.

[0019] In some embodiments, configuring the audio channel of the network audio all-in-one device according to a trigger operation includes: displaying audio stream information corresponding to the preset output audio channel according to a first trigger operation on the management interface, the audio stream information including multiple input nodes and multiple output nodes; wherein the input node is a microphone in a microphone array, and the output node is one of the audio playback devices; receiving a user selection operation on multiple input nodes sent by the control device, configuring the input device corresponding to the preset output audio channel; receiving a user selection operation on multiple output nodes sent by the control device, configuring the output device corresponding to the preset output audio channel. This solution supports multiple input and output nodes, allowing users to freely select input and output devices as needed, enhancing the system's usability and adaptability. Attached Figure Description

[0020] To more clearly illustrate the technical solutions of the embodiments of this application, the drawings used in the description of the embodiments of this application will be briefly introduced below. Obviously, the drawings described below are only some embodiments of this application. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0021] Figure 1 This is a schematic diagram of the structure of an audio signal processing system provided in an embodiment of this application;

[0022] Figure 2 This is a schematic diagram of the structure of an audio signal processing system provided in another embodiment of this application;

[0023] Figure 3 This is a flowchart of an audio signal processing method provided in an embodiment of this application;

[0024] Figure 4 This is a schematic diagram illustrating the process of accessing the management interface through a browser, as provided in an embodiment of this application.

[0025] Figure 5a and Figure 5b These are schematic diagrams of the user mode and developer mode provided in the embodiments of this application, respectively.

[0026] Figure 6a and Figure 6b This is a schematic diagram of an interface in developer mode provided in an embodiment of this application;

[0027] Figures 7a to 7c This is another schematic diagram of the interface in the developer mode provided in the embodiments of this application;

[0028] Figure 8a and Figure 8b This is a schematic diagram of the interface for implementing AI tuning provided in an embodiment of this application;

[0029] Figure 9 This is a schematic diagram illustrating a scenario where the omnidirectional sound pickup device provided in this application is used in conjunction with a local sound pickup amplifier.

[0030] Figure 10 This is a schematic diagram of the ceiling microphone and handheld microphone cascaded amplification provided in this application embodiment, and the linkage with the control screen for in-field control. Detailed Implementation

[0031] To make the objectives, technical solutions, and advantages of this application clearer, the following detailed description is provided in conjunction with the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are merely illustrative and not intended to limit the scope of this application. All other embodiments obtained by those skilled in the art based on the embodiments in this application without inventive effort are within the scope of protection of this application.

[0032] It should be noted that, unless there is a conflict, the various features in the embodiments of this application can be combined with each other, all of which are within the protection scope of this application. Furthermore, although functional modules are divided in the device schematic diagram and a logical order is shown in the flowchart, in some cases, the steps shown or described can be executed in a different order than the module division in the device or the order in the flowchart. Moreover, the terms "first" and "second" used in this application do not limit the data or execution order, but only distinguish between identical or similar items with essentially the same function and effect.

[0033] Network audio devices are devices that transmit and process audio signals over a network, typically used for functions such as audio broadcasting, conferencing systems, live streaming, and voice communication. These devices digitize audio signals, enabling transmission over local area networks (LANs) or wide area networks (WANs), thus achieving more flexible and efficient audio management and processing. Network audio devices include audio interfaces / converters, audio matrix switchers, recording equipment, voice processors, and communication devices.

[0034] Current network audio devices typically employ a distributed architecture, allowing audio processing functions to be distributed across multiple nodes, improving system flexibility and scalability. However, this process involves numerous devices connected via a network, which can lead to relatively cumbersome cabling, especially in large systems. Furthermore, since all devices are network-connected, users can remotely control, configure, and monitor the entire audio system using computers or mobile devices. While the distributed architecture of network audio devices offers flexibility and convenience, users still rely on host computer software to configure all product functions. Different brands or models of devices may use different host computer software, potentially limiting the user's flexibility in system configuration and control. Moreover, for users without technical backgrounds, the complex interface and functions of the host computer software can pose a learning curve, requiring time to adapt. Therefore, current network audio devices suffer from drawbacks such as numerous connections, complex deployment, difficulty in quick debugging and maintenance, and the need for dedicated personnel for maintenance.

[0035] Therefore, it is necessary to provide a solution to address the problems of existing network audio systems, such as configuration dependence on host computers, complex wiring, and high debugging and maintenance costs.

[0036] This application provides an audio signal processing system that integrates multiple key audio processing functions and management control into a single network audio all-in-one unit. It leverages network technology to achieve flexible remote management and efficient audio processing. The system integrates a switch, a DSP processor, and a main control chip into a single network audio all-in-one unit, reducing the complexity of external connections and wiring, and simplifying installation and maintenance. The DSP processor is responsible for real-time processing of the input audio signal, including noise reduction, echo cancellation, equalization, and reverb, to improve audio clarity and quality. The main control chip is also integrated into the network audio all-in-one unit, enabling users to access the device's management interface via the network for parameter configuration, device control, and status monitoring, thus providing flexible management. Through the collaboration between the main control chip and the switch, the system achieves network access and IP address acquisition for the network audio all-in-one unit, enabling efficient device identification and interaction in a network environment. The management interface is designed to provide a user-friendly interface, allowing even users without technical backgrounds to easily control the device, monitor audio status, and perform maintenance.

[0037] Therefore, the audio signal processing system of this application integrates a switch, a DSP processor, and a main control chip into one unit, greatly reducing the number and complexity of devices and facilitating management and maintenance. With the embedded main control chip, users can perform local or remote configuration and debugging via a browser without installing additional software, achieving plug-and-play functionality and cross-platform access. This system solves the complexities of wiring, configuration, and maintenance in traditional audio systems, achieving overall optimization that is simple to deploy, fast to configure, and easy to maintain.

[0038] Please see Figure 1 , Figure 1 This is a schematic diagram of the structure of an audio signal processing system provided in an embodiment of this application. Figure 1 As shown, the audio signal processing system 100 includes a network audio all-in-one unit 10, a microphone array 20, and an audio playback device 30. The network audio all-in-one unit 10 integrates a DSP processor 11, a switch 12, and a main control chip 13. The DSP processor 11 is connected to the microphone array 20, the switch 12, and the audio playback device 30. The main control chip 13 is also connected to the DSP processor 11, the microphone array 20, the switch 12, and the audio playback device 30.

[0039] The DSP processor 11 is a microprocessor specifically designed for digital signal processing. It receives audio signals from the microphone array 20 and performs complex processing such as filtering, enhancement, reverberation, echo cancellation, and equalization to improve sound quality and clarity, aiming to enhance audio quality. The switch 12 is a network device that provides connectivity and communication between multiple network devices. It supports Power over Ethernet (PoE), allowing connected devices to not only communicate over the network but also receive power. The switch 12 reduces the amount of cabling required. The main control chip 13 is the core control unit of the audio signal processing system 100, responsible for coordinating and controlling all components of the audio signal processing system 100, including the DSP processor 11, microphone array 20, switch 12, and audio playback device 30. The main control chip 13 monitors the operating status of each device to ensure normal system operation and can promptly detect and handle faults. The main control chip 13 is connected to the network via the switch 12, managing the device's IP address allocation and network communication; it also provides a user interface, receiving user input and settings, allowing users to select audio processing algorithms and other control parameters; it can also send instructions to the DSP processor 11 to configure audio processing, ensuring that the audio signal is processed according to user requirements. The main control chip 13 plays a central control role in the system 100, ensuring that all components work together to achieve efficient audio processing and management functions.

[0040] Microphone array 20 may consist of multiple microphones used to capture ambient sound and convert it into electrical signals (audio signals). Microphone array 20 can selectively focus on specific sound sources (such as speakers) and filter out background noise. Microphone array 20 can pick up audio in various environments, such as conference rooms, auditoriums, or other venues requiring high-quality audio capture. Audio playback device 30 receives the processed audio signal output from DSP processor 11 and amplifies the audio signal to drive speakers or other output devices, ensuring that the sound is clearly received by the audience.

[0041] In this embodiment, the microphone array 20 is used to acquire audio signals and input them to the DSP processor 11. The DSP processor 11 receives the audio signals input from the microphone array 20, performs voice processing on the audio signals, and generates a processed audio signal. The audio playback device 30 receives the processed audio signal output from the DSP processor 11, amplifies the processed audio signal, and outputs the amplified audio signal. The switch 12 provides network connectivity and power to the network audio all-in-one unit 10.

[0042] In this audio signal processing system 100, the collaborative operation of various devices enables efficient audio capture, processing, and playback. The microphone array 20 is responsible for acquiring audio signals, the DSP processor 11 performs intelligent processing of the audio signals, and the audio playback device 30 amplifies the processed signal for output. The switch 12 ensures network connectivity and power supply for the entire audio signal processing system 100, enabling remote management and control of the devices. This integrated and interdependent design allows the audio signal processing system 100 to achieve flexible, efficient, and high-quality audio processing and playback functions.

[0043] In some embodiments, please refer to Figure 2This invention provides an audio signal processing system. The main control chip 13 includes an analog pickup unit, a network audio pickup unit, and an analog-to-digital converter (ADC). The analog pickup unit is connected to the DSP processor 11 via the ADC, and the network audio pickup unit is connected to both the switch 12 and the DSP processor 11. The main control chip 13 is used to acquire a first audio signal from a local analog microphone via the analog pickup unit, convert the first audio signal to digital via the ADC, and then input it to the DSP processor 11. It is also used to acquire a second audio signal sent by a remote network device via the switch 12 via the network audio pickup unit, and input the second audio signal to the DSP processor 11. This supports the access of local traditional analog microphones (such as conference microphones) as well as network audio devices (such as remote speaker columns). By integrating analog and network pickup capabilities, it achieves device compatibility and deployment flexibility. The analog pickup unit is the starting point for audio acquisition; it is used to connect to analog microphones or other audio input devices. After connection, it can be powered by 48V phantom power and converted to a digital signal via an ADC. The network audio pickup unit is a device that can connect to a network and receive audio signals from other network devices (such as computers, smart devices, or other audio sources). It has the capabilities of audio capture and digital signal processing. The network audio pickup unit supports audio stream transmission protocols over the network (such as RTP, RTSP, etc.), thereby achieving remote audio signal capture. The first and second audio signals are transmitted to the DSP processor 11 for further processing and analysis.

[0044] The switch 12 includes a network data forwarding unit, a power supply unit, and a network audio device detection unit. The power supply unit is connected to both the network data forwarding unit and the network audio device detection unit. The network data forwarding unit is connected to the DSP processor 11 via a network audio pickup unit. The network audio device detection unit determines the power requirement of the network audio device when it is detected, and controls the power supply unit to provide power to the network audio device. The network data forwarding unit receives, parses, and forwards data packets from the connected network audio device to enable data communication between the connected network audio device and the DSP processor 11. The switch 12 connects to the network audio devices and can include multiple audio devices (N devices). Each device can transmit audio data over the network via network connection and power supply (PoE can be used). The network audio device detection unit monitors the status and performance of the network audio devices to ensure real-time acquisition of device operating conditions. It may include sensors or monitoring software to collect device status information. The network data forwarding unit forwards or routes audio data to enable data transmission between different devices, ensuring normal forwarding of audio streams and stable network connections. The power supply unit provides power to each network audio component to ensure stable system operation. Switch 12 ensures smooth data flow in the network, reduces latency, and improves the reliability of audio transmission through its network data forwarding unit. Its power supply unit simplifies cabling requirements and reduces the complexity of device deployment. The network audio device detection unit allows for real-time monitoring and management of network audio devices connected to Switch 12, including ensuring proper device operation and performance monitoring. Switch 12 is crucial for network audio device systems, providing not only network connectivity and power supply but also integrated data forwarding and device detection functions.

[0045] The main control chip 13 also includes a network audio amplification unit, an analog interface, a DAC converter, and analog amplification equipment. The network audio amplification unit is connected to both the DSP processor 11 and the switch 12. One end of the analog interface is connected to the DSP processor 11, and the other end is connected to the analog amplification equipment via the DAC converter. The analog-to-digital converter (DAC) is connected to the DSP processor 11 via the analog interface and is used to convert digital audio signals into analog signals for output through analog amplification equipment (such as speakers). Specifically, the DAC converter receives digital audio signals from the DSP processor 11, which processes the audio signals, including noise reduction, equalization, and other audio effects. Then, the DAC converter converts these processed digital signals back into analog signals to drive speakers, headphones, or other analog audio output devices. The DAC converter enables the processed audio signals to be output through local audio equipment, providing a clear and high-quality audio experience. The network audio amplification unit is connected to the network audio equipment via the switch 12 and is primarily used for amplifying and processing remote audio signals. In addition to amplification, the network audio amplification unit can also have audio processing functions, such as volume adjustment, equalization settings, and other sound effects processing, to improve the final sound quality. Through switch 12, the network audio amplification unit can not only send local audio signals, but also obtain power using PoE (Power over Ethernet) technology, simplifying wiring and installation.

[0046] In this embodiment, the network audio all-in-one machine 10 also features real-time audio processing and feedback, multi-source audio input, and remote control functions. A USB sound card can be connected to the DSP processor 11. Through this USB sound card, terminal devices can connect to the network audio all-in-one machine 10, and the terminal devices (such as speakers or headphones) can instantly receive processed audio signals for real-time audio playback. Furthermore, the application APP allows users to conveniently control audio devices through terminal devices (such as computers, tablets, and mobile phones). Users can adjust volume, select sound effects, and switch audio sources through the APP, providing a user-friendly experience. Users can control audio devices associated with the network audio all-in-one machine 10 even when they are away from the audio devices. Additionally, the USB sound card can be used to connect the terminal device's audio source to the DSP processor 11, enabling multi-source input.

[0047] The DSP processor 11 is also connected to the AI ​​module and the Web control module, providing intelligent processing capabilities such as voice recognition and intelligent gain control; it can also be used for system settings and remote control through the Web management interface.

[0048] The aforementioned network audio all-in-one machine 10 uses the DSP processor 11 as its core to process audio, providing an efficient, flexible, and intelligent audio solution for the entire network audio all-in-one machine 10, which greatly improves audio quality and user experience.

[0049] The main control chip 13 is connected to the DSP processor 11, microphone array 20, switch 12, and audio playback device 30. The main control chip 13 manages and coordinates all connected components, including the DSP processor 11, microphone array 20, switch 12, and audio playback device 30, ensuring efficient collaboration among all parts of the system 100. The main control chip 13 can monitor the operating status of each component and adjust and respond promptly to different operating conditions, ensuring the stability and reliability of the system 100. For example, the main control chip 13 can send operation commands to the DSP processor 11 for tasks such as audio signal processing and effects adjustment. Simultaneously, it can coordinate the operation of the microphone array 20 to achieve multi-channel audio input. The main control chip 13 also has various communication interfaces (such as I2C, SPI, UART, etc.) to facilitate data exchange with other electronic components (such as camera equipment) and implement control logic in relevant application scenarios. The main control chip 13 can dynamically adjust the processing parameters of the DSP processor 11 according to the system status and user input to optimize audio effects. The main control chip 13 also features rapid response capabilities, allowing it to react instantly to user operations or input signals, thus enabling dynamic adjustment of system performance. As the control center of the entire network audio all-in-one machine 10, the main control chip 13 coordinates the operation of various components in the system, manages audio streams, enables user interaction, and dynamically adjusts system settings. Its efficient control capabilities and flexible configuration ensure the stability of the entire system and a user-friendly experience.

[0050] In this embodiment, the main control chip 13 includes a Web control module and a status control module. The Web control module receives service requests and responds to trigger operations on the management interface. The status control module controls the operating status of the DSP processor 11, microphone array 20, switch 12, and audio playback device 30. Through the Web control module and status control module, users can remotely configure and debug via a browser without installing additional software, achieving power-on functionality and cross-platform access. Specifically, based on this audio signal processing system 100, an audio signal processing method is also provided, which is executed by the main control chip 13, such as... Figure 3 As shown, the method includes:

[0051] S11. Receive a service request through a preset network. This service request is sent by a control device that accesses the preset network via its IP address.

[0052] The control switch connects the network audio all-in-one device to a preset network and obtains its IP address. The network audio all-in-one device then enters the preset local network through the switch, enabling it to communicate with control devices (such as computers, smartphones, or other audio devices) on the network. Once connected to the network, the system automatically obtains (e.g., via DHCP (Dynamic Host Configuration Protocol)) a unique IP address for the network audio all-in-one device, allowing control devices to easily access and control it.

[0053] S12. Respond to the service request and send the management interface to the control device so that the control device displays the management interface.

[0054] The management interface is used for parameter configuration, device control, audio status monitoring, and system maintenance of the network audio all-in-one unit 10. The web control module allows the network audio all-in-one unit 10 to interact with the user corresponding to the control device via a web interface. The web control module can receive service requests from control devices in a preset network (sent via their IP addresses), and these service requests may include parameter configuration, device control, etc. For user requests, the web control module will process them accordingly, generating and returning the management interface of the network audio all-in-one unit 10. This management interface is a user-friendly web interface that users can access through a browser. Users can use the management interface to make various settings for the audio device, such as changing the volume, selecting sound effects, and adjusting other audio parameters. Users can directly operate the device, such as turning it on / off and switching input sources. The management interface can display the real-time operating status of the audio device, such as the current audio input source, audio signal strength, and processing status, helping users understand the device's performance.

[0055] The collaboration of steps S11 and S12 provides the network audio all-in-one device 10 with crucial network access and control capabilities. Step S11 ensures that the network audio all-in-one device 10 can stably access the preset network and obtain its own IP address through the switch 12. Step S12 provides users with a powerful and easy-to-use management interface through the Web control module, enabling the control device to easily perform audio control, parameter configuration, and audio status monitoring, thereby improving the user experience and device interactivity. The control device can be an all-in-one device, a smartphone, a personal computer, a tablet computer, etc.

[0056] For example, in a conference room, the network audio all-in-one unit 10 can be directly connected to the conference speakers. This device has a built-in array microphone that can directly pick up the speaker's voice, eliminating the need for the speaker to hold a microphone. The speaker's voice is amplified, ensuring everyone in the conference room can hear it. Furthermore, the network audio all-in-one unit 10 automatically mixes, cancels echo, and suppresses noise to ensure clear speaker audio. Additionally, an intuitive management interface is provided on the user's end, displayed on the corresponding large screen of the all-in-one unit. This interface supports real-time audio tuning, zone control, automatic gain adjustment, and more, and supports both web and touchscreen operation, allowing even non-professional users to easily adjust audio parameters.

[0057] S13. Receive user-triggered operations on the management interface sent by the control device.

[0058] S14. Based on the trigger operation, configure the audio channel of the network audio all-in-one machine. This audio channel is used to indicate the audio signal transmission link between the microphone array and the audio playback device.

[0059] For example, when the mouse cursor hovers over a preset output audio channel on the management interface, the audio stream information corresponding to the preset output audio channel is displayed. This audio stream information includes multiple input nodes and multiple output nodes. The system receives user selection operations for multiple input nodes from the control device and configures the input devices corresponding to the preset output audio channel. It also receives user selection operations for multiple output nodes from the control device and configures the output devices corresponding to the preset output audio channel. The input nodes are microphones in the microphone array 20, there are multiple audio playback devices, and the output node is one of the audio playback devices 30.

[0060] In this embodiment, the management interface includes an audio input device list and an audio output device list. The audio input device list contains microphones from all connected microphone arrays 20, allowing users to view and select available input devices. The audio output device list contains all available audio playback devices 30, allowing users to select output devices. Users can select multiple input nodes from the audio input device list; each input node is a microphone from the microphone array 20. After a user selects a specific microphone, the system confirms that microphone as the input device for the audio stream, ensuring that the captured sound can be further processed. Similarly, users can select multiple output nodes from the audio output device list. Each output node is any device from the audio playback devices 30. After a user selects a device, the system determines that device as the output device for the audio stream, ensuring that the processed audio signal can be played through the selected device. The management interface supports intuitive click and selection operations, allowing users to easily browse input and output devices and select the device best suited to their current needs. After selection, the system updates the audio stream information in real time, ensuring that users receive the latest device status and its corresponding audio signal stream. Furthermore, the system provides real-time feedback, displaying the input and output devices selected by the user to ensure that the user confirms that the selection is correct, thereby improving the accuracy and efficiency of the operation.

[0061] This embodiment supports multiple input and output nodes, allowing users to freely select input and output devices as needed, enhancing the system's usability and adaptability. Through this management interface, users can easily manage audio input and output devices, ensuring smooth audio stream transmission and optimizing the audio processing experience. This design enhances the system's ease of use and flexibility, making it suitable for various application scenarios.

[0062] In some embodiments, such as Figure 4 As shown, the Web control module receives service requests from users, including usernames and passwords. If the username and password match preset conditions, the main control chip 13 responds to the service request and returns the corresponding management interface of the network audio all-in-one machine 10. This ensures that access to the management interface is only possible with the correct username and password, thus improving the security of the audio signal processing system 100.

[0063] The aforementioned management modes include a user mode and a developer mode. The user mode is for general users, such as meeting hosts, while the developer mode is for technical personnel, such as channel partners, pre-sales, and after-sales staff. The user mode allows general users to clearly understand the operating status of existing devices and quickly configure devices for control and operation. The developer mode allows users to manage multiple devices, customize audio streaming routing rules, adjust the audio output content of multiple devices, and monitor audio intercom in real time. By setting these two modes, different management options can be provided for different users, thereby improving system usability and security, reducing maintenance costs, preventing system anomalies caused by misoperation, and meeting the needs of professionals for in-depth debugging and maintenance.

[0064] For example, such as Figure 5a The diagram shows the management interface in simple mode (i.e., user mode). It mainly includes audio pickup devices and audio playback devices. The pickup devices include available wireless and wired microphones, as well as other pickup devices. Users can select the desired device for audio input from this list. Audio playback devices include speakers, etc., which users can choose to send processed audio signals for playback. In addition to pickup and playback devices, the management interface also displays USB and other linear interfaces for easy device connection and expansion. This simple management interface design allows users to intuitively select and manage audio input and output devices, suitable for audio processing needs in different scenarios. Furthermore, the overall layout is clear, making it easy for users to quickly get started.

[0065] like Figure 5b The diagram shown illustrates the management interface in professional mode (i.e., developer mode). Besides displaying the microphone, audio playback device, and interfaces, it also includes an audio processor module. This audio processor module includes input channel configuration and output channel configuration. For example, as shown... Figure 5b As shown, the input channel configuration supports 1 to 32 input channels, allowing users to select and configure multiple input sources as needed. Input channel types include analog input (for connecting traditional audio devices) and USB input (for connecting digital audio sources, ensuring high-quality audio transmission). The output channel configuration supports 1 to 24 output channels, allowing users to set audio outputs according to their needs. Output channel types include analog output (for connecting speakers or other analog devices) and USB output (for transmitting audio signals to digital devices, supporting high-fidelity sound quality). Furthermore, users can independently configure each input and output channel, such as adjusting gain and selecting audio processing algorithms, to achieve personalized audio processing effects. Real-time signal monitoring is also provided, allowing users to view the audio signal status of each channel to ensure efficient system operation. Figure 5bThe professional-mode management interface gives developers greater flexibility and control, enabling them to deeply configure audio processing parameters to meet complex audio processing needs. This design is suitable for professional audio workers, ensuring optimized audio quality and performance in various application scenarios.

[0066] In professional mode (i.e., developer mode), the management interface displays the audio stream information corresponding to a preset output audio channel when the mouse cursor hovers over it. This audio stream information includes multiple input and output nodes, allowing users to quickly understand the current audio configuration. Users can select multiple input nodes (microphones in a microphone array) through the interface, easily choosing the desired input device to ensure accurate audio signal capture. Users can also select multiple output nodes (audio playback devices), enabling flexible transmission of audio signals to desired amplification or other output devices. This allows for rapid acquisition of relevant data for the currently selected audio stream, improving efficiency. Users can quickly understand the audio stream configuration, easily select the required input and output devices, and ensure correct sound transmission to the desired amplification or other output devices. For example, as... Figure 6a As shown, by hovering the mouse over the output audio channel (specifically, the configuration channel corresponding to that output audio channel), the input channel configuration and output channel configuration corresponding to the current output audio channel can be quickly displayed, giving the user a clear understanding of the overall structure of the audio stream. For example... Figure 6b As shown, after clicking on any output channel configuration, the user can quickly select the device output source for the desired sound output, allowing the user to clearly understand the relationship between the selected device and the audio, ensuring the accuracy and effectiveness of the audio output.

[0067] In professional mode (i.e., developer mode), the management interface is also used to receive the user-selected audio processing algorithm module, which is used to process the audio input from the input device. Specifically, after receiving the user-selected audio algorithm module through the management interface, the web control module enables the selected module to process the audio input from the input device. In this way, users can configure and optimize the input audio stream by selecting an audio processing module to meet different usage needs. For example, ... Figures 7a to 7c As shown, Figure 7a In the middle, users can enter through clicks, such as Figure 7b The channel content configuration interface shown is in Figure 7cThe system performs high-level algorithm configuration for mixed audio, quickly adjusting to the optimal settings. Therefore, by clicking on the configuration channel of a node, one can access the output sound configuration interface for that node. This interface allows for quick selection of the desired sound input and the corresponding enabled / disabled algorithm modules. For example... Figure 8a and Figure 8b As shown, the system also supports one-click AI tuning. The system can monitor the current audio environment in real time, such as background noise levels, sound source location, and audio content characteristics. The AI ​​module analyzes these parameters to determine the optimal audio settings. Users simply click the "AI Auto Tuning" button, and the system automatically adjusts the audio effects, saving time and complexity associated with manual configuration. The tuning process includes dynamic equalization, volume adjustment, and low-frequency and high-frequency enhancement to provide a clear and bright sound. The one-click AI tuning function makes the network audio all-in-one device more intelligent and automated in audio adjustment, providing a convenient user experience and optimizing sound quality according to the environment and user needs.

[0068] The audio signal processing system 100 of this application embodiment can be used in conference scenarios. Through the DSP processor 11 and one-click AI audio tuning, the system 100 can provide clear and natural audio in the conference, ensuring smooth communication between participants.

[0069] In some embodiments, such as Figure 9 As shown, the audio signal processing system 100 also includes an omnidirectional microphone, which can be one of the microphone arrays 20. The omnidirectional microphone can be communicatively connected to the DSP processor 11 via the switch 12. The omnidirectional microphone is used to collect audio signals in a preset scene and send the audio signals in the preset scene to the DSP processor 11 via the switch 12; the DSP processor 11 is used to process the audio signals in the preset scene and output the processed audio signals to a remote audio device to achieve remote audio transmission. Specifically, the DSP processor 11 does not amplify the audio signals in the preset scene collected by the omnidirectional microphone through the audio playback device 30. This allows remote participants to clearly hear the on-site sound, improving the communication efficiency of remote audio and video conferencing. Furthermore, the audio signal that does not pass through the audio playback device 30 reduces distortion and noise, thus maintaining higher sound quality, which provides remote participants with clearer and more natural audio feedback.

[0070] The audio signal processing system also includes a host microphone, which can be one of the microphone arrays 20. The host microphone is communicatively connected to the DSP processor 11. The host microphone is used to acquire the host's audio signal and send it to the DSP processor 11. The DSP processor is also used to: process the host's audio signal; output the processed host's audio signal to a remote audio device for remote audio transmission; and simultaneously play the processed host's audio signal locally through the audio playback device 30. This achieves both local playback and remote synchronous transmission of the host's audio signal, balancing on-site listening experience with remote audio interaction, thus improving the integrity and practicality of the conference system. Furthermore, when the host is speaking, the DSP processor 11 also performs pickup suppression control on the omnidirectional microphone. This achieves priority control of the host's pickup, avoids audio channel interference, prevents feedback, and improves speech clarity.

[0071] The aforementioned preset scenario can be a conference scenario. For example, a company-wide meeting, including on-site audience and host, as well as audiences in remote branch venues. The host conducts the meeting using a handheld microphone, and omnidirectional microphones are deployed on-site to collect the sound in the current meeting room and transmit it to the audience in the remote branch venues. To avoid interference (such as echo, noise, etc.) between local amplification and remote conference audio pickup, the handheld microphone and omnidirectional microphone are connected to the network audio all-in-one unit 10 and managed through the network audio all-in-one unit 10. For example, when the host speaks, the network audio all-in-one unit 10 performs noise reduction, echo cancellation, and gain adjustment on the sound from the handheld microphone, and then sends the processed sound to the local speakers to ensure that the on-site audience can hear clearly; if the host's voice needs to be transmitted to the remote conference, the network audio all-in-one unit 10 sends its signal to the remote conference system. When on-site audience members speak, the omnidirectional microphones collect the speaking sound from the local meeting room and send the signal to the network audio all-in-one unit 10. The network audio all-in-one unit 10 then transmits the processed audio signal directly to the remote conference system, allowing remote audiences to hear the on-site audience members clearly. Since the omnidirectional microphone is mainly used for remote conferencing, it is not connected to the local amplification system to avoid local feedback issues.

[0072] In some embodiments, such as Figure 10As shown, the audio signal processing system 100 also includes a ceiling microphone array, which can also be one of the microphone arrays 20. The ceiling microphone array can be installed on the ceiling to capture ambient sound, making it particularly suitable for venues such as conference rooms and lecture halls. It effectively collects sound from different directions while reducing background noise, thereby improving audio quality. The ceiling microphone array is communicatively connected to the DSP processor 11. The ceiling microphone array is used to collect audio signals from a preset scene and send these signals to the DSP processor 11. The main control chip 13 responds to a pickup area setting request input from the management interface, sets the pickup range of the ceiling microphone array, and sends the pickup range to the DSP processor 11. The DSP processor 11 acquires the audio signals collected within the pickup range of the ceiling microphone array, processes the audio signals, and outputs the processed audio signals to an audio playback device for local playback. In this way, the pickup range can be controlled through the ceiling microphone array, so that the speaker can speak clearly without holding the microphone, solving the difficult scenario of passing the microphone in the venue.

[0073] Users can select relevant settings options on the web management interface, such as selecting the pickup area, setting sensitivity, and adjusting other parameters. Users can intuitively define or adjust the pickup area of ​​the ceiling microphone array through a graphical interface or slider controls. When a user sets the pickup area and submits a request, the main control chip 13 receives this instruction. The main control chip 13 parses the request content, including parameters such as the shape, size, and position of the pickup area, to determine the effective pickup range of the ceiling microphone array. The main control chip 13 can also convert the parsed pickup range parameters (such as coordinates, radius, sensitive area, etc.) into a format that the DSP processor 11 can understand. After configuration, the main control chip 13 sends this pickup range setting information to the DSP processor 11 to adapt to the new pickup range during audio signal processing. Once the pickup range of the ceiling microphone array is configured, each microphone in the array acquires audio signals according to the settings. The acquired audio signals are transmitted to the DSP processor 11 through analog pickup units for subsequent processing. The main control chip 13 acts as the central management and coordination unit, processing user requests for microphone pickup area settings and sending the settings information to the DSP processor 11 to ensure that the pickup range of the ceiling microphone array matches the user's needs. This allows speakers to speak clearly without holding the microphone, bringing significant convenience in meetings and various public occasions and solving problems inherent in traditional microphone-passing scenarios.

[0074] For example, in large conference rooms with many speakers, traditional handheld microphones are cumbersome to use. Two to four ceiling microphones can be deployed to cover the conference area. These ceiling microphones communicate with the network audio all-in-one unit 10 to achieve audio data transmission. The built-in DSP processor 11 in the network audio all-in-one unit 10 can automatically perform echo cancellation, noise suppression, automatic gain control, and intelligent mixing to ensure clear and natural sound. Furthermore, the corresponding meeting control screen management interface of the network audio all-in-one unit 10 allows users (meeting hosts, etc.) to adjust the pickup area, enable / disable specific microphones, and adjust volume gain in real time. The system can also automatically identify the current speaker and dynamically adjust the pickup direction. The system can also automatically optimize audio output and amplify it to the entire room when necessary.

[0075] The audio signal processing system 100 also includes a host microphone, which can also be one of the microphone arrays 20. The host microphone is communicatively connected to the DSP processor 11. The host microphone is used to acquire the host's audio signal and send it to the DSP processor 11 via an analog pickup unit. The DSP processor 11 sends the host's audio signal to the main control chip 13. The main control chip 13 is used to: perform pickup suppression control on the ceiling microphone array when the host's audio signal is detected; and control the ceiling microphone array to resume pickup when the host's audio signal is not detected, thus enabling the system to automatically control the pickup state of the ceiling microphone array. The main control chip 13 can also be used to: respond to a ceiling microphone suppression request from the management interface when the host is speaking and perform pickup suppression control on the ceiling microphone array; and respond to a ceiling microphone enable request from the management interface when the host is not speaking and control the ceiling microphone array to resume pickup, thus enabling the system to control the pickup state of the ceiling microphone array based on user operations on the management interface. In this embodiment, when the host speaks, the system automatically suppresses the ceiling microphone array to prevent the recording of ambient noise or echoes, avoiding reverberation, feedback, and audio interference caused by multi-channel audio pickup, effectively improving speech clarity. After the host pauses, the ceiling microphone array resumes operation and can normally pick up the speeches of other participants. The system can react according to the duration of the host's pause; for example, if the pause time is longer than a preset time, the ceiling microphone array will resume operation.

[0076] In some embodiments, the audio signal processing system 100 further includes a camera device, which is communicatively connected to the main control chip 13 via a switch 12. The main control chip 13 is used to: acquire speaker information collected by the camera device; adjust the pickup range of the ceiling microphone array according to the speaker information; acquire the audio signal collected within the adjusted pickup range of the ceiling microphone array, and send the audio signal to the DSP processor 11. The DSP processor 11 is used to process the audio signal and output the processed audio signal to the audio playback device 30, so that the audio signal collected within the adjusted pickup range of the ceiling microphone array can be played locally by the audio playback device 30. In this way, the camera device can intelligently track the speaker, and the network audio all-in-one machine 10 can adjust the pickup range of the ceiling microphone array according to the tracked speaker, thereby more accurately picking up the speaker's voice and improving voice quality and user experience.

[0077] The audio signal processing system and method provided in this application, through the integration of a network audio all-in-one machine and a management interface, comprehensively optimizes conference audio processing. This includes: improved conference efficiency and optimized user experience, where participants no longer need to hold or pass microphones, resulting in more natural and smoother speaking; simple deployment, automatic sound pickup and amplification upon startup, easy for beginners to use; automatic adjustment of volume for different speakers to avoid feedback and noise interference; and a visual operation interface for quick adjustment of audio parameters, allowing for easy management even for non-professionals. Furthermore, it reduces deployment and maintenance costs, as the switch, DSP processor, and main control chip are integrated into the network audio all-in-one machine, simplifying hardware deployment; networked deployment eliminates the need for complex audio wiring, making later maintenance more convenient; the device supports remote control via web / touchscreen, allowing maintenance personnel to debug the device from different locations, reducing labor costs. Finally, it is compatible with local sound reinforcement and remote conferencing, with the network audio all-in-one machine featuring a built-in intelligent echo cancellation algorithm to ensure clear, noise-free sound for remote participants; and for large conference rooms, different pickup areas can be set to optimize the audio experience in different meeting areas.

[0078] The above-disclosed embodiments are merely preferred embodiments of this application and should not be construed as limiting the scope of this application. Therefore, any equivalent variations made in accordance with the claims of this application shall still fall within the scope of this application.

Claims

1. A network audio all-in-one device, characterized in that, include: The network audio all-in-one machine is connected to a microphone array and an audio playback device. The network audio all-in-one machine includes a switch, a DSP processor and a main control chip. The DSP processor is connected to the microphone array, the switch and the audio playback device respectively. The main control chip is connected to the DSP processor, the microphone array, the switch and the audio playback device respectively. The switch is also connected to the microphone array. The main control chip is used to control the working status of the DSP processor, the microphone array, the switch, and the audio playback device; The main control chip is also used for: Service requests are received through a preset network; the service requests are sent by control devices that access the preset network via IP addresses. In response to the service request, a management interface is sent to the control device, causing the control device to display the management interface; Receive user-triggered operations on the management interface sent by the control device; Configure the audio channel of the network audio all-in-one machine according to the trigger operation; The audio channel is used to indicate the audio signal transmission link between the microphone array and the audio playback device.

2. The network audio all-in-one machine according to claim 1, characterized in that, The step of configuring the audio channel of the network audio all-in-one machine according to the trigger operation includes: Based on the first trigger operation of the preset output audio channel on the management interface, the audio stream information corresponding to the preset output audio channel is displayed. The audio stream information includes multiple input nodes and multiple output nodes. The input node is a microphone in the microphone array, and the output node is one of the audio playback devices. Receive user selection operations for the multiple input nodes sent by the control device, and configure the input device corresponding to the preset output audio channel; The system receives user selection operations for the multiple output nodes sent by the control device and configures the output device corresponding to the preset output audio channel.

3. The network audio all-in-one machine according to claim 2, characterized in that, After configuring the input device corresponding to the preset output audio channel, the management interface is used for: The system receives an audio processing algorithm module selected by the user, which performs algorithmic processing on the audio input from the input device.

4. The network audio all-in-one machine according to claim 1, characterized in that, The switch includes a network data forwarding unit, a power supply unit, and a network audio device detection unit. The power supply unit is connected to both the network data forwarding unit and the network audio device detection unit. The network data forwarding unit is connected to the DSP processor. The network audio device detection unit is used to determine the power required by the network audio device when a network audio device is detected to be connected, and to control the power supply unit to provide the power to the network audio device. The network data forwarding unit is used to receive, parse, and forward data packets from the accessed network audio device to enable data communication between the accessed network audio device and the DSP processor.

5. The network audio all-in-one machine according to claim 1, characterized in that, The microphone array includes a ceiling microphone array, which is communicatively connected to the DSP processor and the main control chip. The ceiling microphone array is used to collect audio signals in a preset scene and send the audio signals in the preset scene to the DSP processor; The main control chip is used to respond to the sound pickup area setting request input by the management interface, set the sound pickup range of the ceiling microphone array, and send the sound pickup range of the ceiling microphone array to the DSP processor. The DSP processor is used to acquire audio signals collected within the pickup range of the ceiling microphone array, process the audio signals, and output the processed audio signals to the audio playback device, so as to play the audio signals collected within the pickup range of the ceiling microphone array locally through the audio playback device.

6. The network audio all-in-one machine according to claim 5, characterized in that, The microphone array also includes a presenter microphone, which is communicatively connected to the DSP processor. The host microphone is used to collect the host's audio signal and send the host's audio signal to the DSP processor; The DSP processor is used to send the host's audio signal to the main control chip; The main control chip is used for: If the host's audio signal is detected, the ceiling microphone array is controlled to suppress sound pickup; if the host's audio signal is not detected, the ceiling microphone array is controlled to resume sound pickup. or, When the host speaks, the system responds to the ceiling microphone suppression request sent by the management interface and performs sound suppression control on the ceiling microphone array; when the host does not speak, the system responds to the ceiling microphone enable request sent by the management interface and controls the ceiling microphone array to resume sound pickup.

7. The network audio all-in-one machine according to claim 5, characterized in that, The system also includes a camera device, which is communicatively connected to the main control chip via the switch; The main control chip is also used for: Obtain the speaker information collected by the camera device; Adjust the pickup range of the ceiling microphone array based on the speaker information; Acquire audio signals within the pickup range of the adjusted ceiling microphone array and send the audio signals to the DSP processor; The DSP processor is used to process the audio signal and output the processed audio signal to the audio playback device, so that the audio signal collected within the pickup range of the adjusted ceiling microphone array can be played locally through the audio playback device.

8. The network audio all-in-one machine according to claim 1, characterized in that, The microphone array includes an omnidirectional microphone, which is communicatively connected to the DSP processor via the switch; the omnidirectional microphone is used to collect audio signals in a preset scene and send the audio signals in the preset scene to the DSP processor; The DSP processor is used to process the audio signals in the preset scene and output the processed audio signals in the preset scene to a remote audio device to realize remote audio transmission. The DSP processor does not amplify the audio signal in the preset scene collected by the omnidirectional microphone through the audio playback device.

9. The network audio all-in-one machine according to claim 8, characterized in that, The microphone array also includes a presenter microphone, which is communicatively connected to the DSP processor. The host microphone is used to collect the host's audio signal and send the host's audio signal to the DSP processor; The DSP processor is also used for: The host's audio signal is processed; The processed audio signal of the presenter is output to a remote audio device to achieve remote audio transmission; The processed audio signal of the host is played locally through the audio playback device; During the host's speech, the DSP processor is also used to perform sound suppression control on the omnidirectional microphone.

10. The network audio all-in-one machine according to claim 1, characterized in that, The main control chip includes a Web control module and a status control module. The Web control module is used to receive the service request and respond to the trigger operation on the management interface. The status control module is used to control the working status of the DSP processor, the microphone array, the switch and the audio playback device.

11. An audio signal processing system, characterized in that, include: Microphone array, audio playback device, and network audio all-in-one machine as described in any one of claims 1-10.

12. An audio signal processing method, characterized in that, An audio signal processing system, comprising a network audio all-in-one unit, a microphone array, and an audio playback device, wherein the network audio all-in-one unit is connected to the microphone array and the audio playback device respectively, the method comprising: Service requests are received through a preset network; the service requests are sent by control devices that access the preset network via IP addresses. In response to the service request, a management interface is sent to the control device, causing the control device to display the management interface; Receive user-triggered operations on the management interface sent by the control device; According to the trigger operation, the audio channel of the network audio all-in-one machine is configured; the audio channel is used to indicate the audio signal transmission link between the microphone array and the audio playback device.

13. The method according to claim 12, characterized in that, The step of configuring the audio channel of the network audio all-in-one machine according to the trigger operation includes: Based on the first trigger operation of the preset output audio channel on the management interface, the audio stream information corresponding to the preset output audio channel is displayed. The audio stream information includes multiple input nodes and multiple output nodes. The input node is a microphone in the microphone array, and the output node is one of the audio playback devices. Receive user selection operations for the multiple input nodes sent by the control device, and configure the input device corresponding to the preset output audio channel; The system receives user selection operations for the multiple output nodes sent by the control device and configures the output device corresponding to the preset output audio channel.

14. The method according to claim 12, characterized in that, The service request includes a username and password; responding to the service request by sending a management interface to the control device, causing the control device to display the management interface, includes: If the username and password meet the preset conditions, the service request is responded to, and a management interface is sent to the control device, causing the control device to display the management interface.

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