A conference scene switching control method and device and a storage medium

By comparing local and global variables to generate a set of difference parameters, and updating only the parameters of the target audio component, the signal jitter problem of the audio processor during scene switching in a multi-conference room scenario is solved, achieving efficient and stable scene switching.

CN120762336BActive Publication Date: 2026-01-06SHENZHEN TENDZONE INTELLIGENT TECH
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Patent Information

Application Number
CN202511271393.4
Authority / Receiving Office
CN · China
Patent Type
Patents(China)
Current Assignee / Owner
Filing Date
2025-09-08
Publication Date
2026-01-06
Estimated Expiration
2045-09-08

AI Technical Summary

Technical Problem

When switching between multiple meeting rooms, the existing audio processor causes signal jitter or brief silence, affecting the continuity of the meeting. This is because the DSP needs to reload all parameters, interrupting the audio processing flow of other meeting rooms.

Method used

By storing a subset of parameters of scene-related components in local variables and comparing them with global variables to generate a set of differing parameters, only the parameters of the target audio component are updated. The processing logic parameters of the audio component are configured using DSP functions to achieve the switching of meeting scenes.

Benefits of technology

It eliminates the global interruption problem in multi-meeting room scenarios, ensures no signal jitter during meeting switching, reduces switching latency to the millisecond level, reduces parameter transmission volume, and improves scene calling efficiency.

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Abstract

The application discloses a conference scene switching control method and device, and a storage medium, and comprises the following steps: in response to a scene switching instruction, an audio component corresponding to the scene switching instruction is acquired; a parameter value of the audio component is saved as a local variable; a parameter value difference between the local variable and a global variable is acquired, a difference parameter set is generated according to the parameter value difference, wherein, when the audio processor is started, a parameter value of a default audio component is acquired, and the parameter value is saved as a global variable; a DSP function is called to configure a processing logic parameter of the audio component according to the difference parameter set, so that a conference scene switching function is realized. The application stores a parameter subset of a scene-related component through a local variable, compares the local variable with a global variable to generate a difference set, only updates a target audio component parameter, ensures that there is no signal jitter and the switching delay is reduced to the millisecond level during conference switching, greatly reduces the parameter transmission amount, and improves the efficiency of the calling scene.
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Description

Technical Field

[0001] This application relates to the field of audio processing technology, and in particular to a control, device and storage medium for switching meeting scenarios. Background Technology

[0002] With the rapid development of audio processing technology, modern audio processors often adopt an ARM+DSP collaborative architecture to balance control flexibility and real-time processing capabilities. To reduce costs, existing designs typically only configure a memory chip on the ARM side to store the executable program and scene configuration files (such as JSON format), while the DSP side does not have independent storage; the algorithm parameters required for its operation must be sent to the ARM in real time via the SPI interface. For example, in a conference scenario, users can switch audio effects with a single click by calling different scene configuration files (such as "Conference Mode" and "Tea Break Mode") to meet diverse needs.

[0003] Since the scene configuration file contains all algorithm parameters, when a single audio processor serves multiple conference rooms, any conference room calling the scene will trigger a global parameter update. The DSP needs to reload all parameters, which will cause the audio processing flow of other conference rooms to be forcibly interrupted, resulting in signal jitter or brief silence, affecting the continuity of the meeting.

[0004] The above content is only used to help understand the technical solution of this application and does not represent an admission that the above content is prior art. Summary of the Invention

[0005] The main purpose of this application is to provide a control method, device and storage medium for switching meeting scenarios, which aims to solve the technical problem of abnormal audio parameter switching in existing multi-meeting room scenarios.

[0006] To achieve the above objectives, this application proposes a control method, device, and storage medium for switching conference scenes. The control method for switching conference scenes is applied to an audio processor, and the method includes:

[0007] In response to a scene switching command, the audio component corresponding to the scene switching command is obtained;

[0008] Save the parameter values ​​of the audio component as local variables;

[0009] Obtain the parameter value difference between the local variable and the global variable, and generate a difference parameter set based on the parameter value difference. Specifically, when the audio processor is started, obtain the parameter value of the default audio component and save the parameter value as a global variable.

[0010] The DSP function is called to configure the processing logic parameters of the audio component according to the set of differences in parameters, thereby enabling the conference scene switching function.

[0011] In one embodiment, the step of saving the parameter values ​​of the audio component as local variables includes:

[0012] Obtain the binary data conversion rule corresponding to the audio component from the preset parameter mapping table, wherein the preset parameter mapping table stores the mapping relationship between all audio components and their corresponding binary data conversion rules in advance;

[0013] Based on the binary data conversion rules, the parameter values ​​of the audio component are converted into binary data blocks, and the binary data blocks are saved as local variables.

[0014] In one embodiment, the step of obtaining the parameter value difference between the local variable and the global variable, and generating a set of difference parameters for the parameter value difference, includes:

[0015] The parameter values ​​of the same audio component are extracted from the local variables and the global variables respectively, and the target audio components with different parameter values ​​are located by using a hierarchical hash algorithm.

[0016] The difference parameter set is generated based on the differences in parameter values ​​of the target audio component.

[0017] In one embodiment, the step of generating the difference parameter set based on the parameter value differences of the target audio component includes:

[0018] Obtain the floating threshold of the audio component, wherein the floating threshold is calculated by obtaining the threshold calculation model of the audio component;

[0019] The difference parameter set is obtained by the difference in parameter values ​​and the floating threshold.

[0020] In one embodiment, the step of calling the DSP function to configure the processing logic parameters of the audio component according to the difference parameter set, thereby realizing the conference scene switching function, includes:

[0021] Determine the audio component corresponding to the set of differences in parameters, wherein the audio component may be one or more.

[0022] The meeting scene is switched by changing the processing logic parameters of the audio component to the difference parameter set.

[0023] In one embodiment, before the step of changing the processing logic parameters of the audio component to the difference parameter set to switch the conference scene, the method further includes:

[0024] Determine the update type of the audio component;

[0025] If the update type is an immediate update component, execute the step of changing the processing logic parameters of the audio component to the difference parameter set to switch the meeting scenario;

[0026] If the update type is a smooth transition component, the step of changing the processing logic parameters of the audio component to the difference parameter set is executed using a pre-set smooth transition mechanism to switch the meeting scene.

[0027] In one embodiment, if the update type is a smooth transition component, the step of changing the processing logic parameters of the audio component to the difference parameter set to switch the conference scene using a pre-set smooth transition mechanism includes:

[0028] The update priority is determined based on the component type of the audio component;

[0029] The processing logic parameters of the audio component are changed to the set of differences in sequence according to the update priority.

[0030] In one embodiment, if the update type is a smooth transition component, the step of changing the processing logic parameters of the audio component to the difference parameter set to switch the conference scene using a pre-set smooth transition mechanism includes:

[0031] Obtain the update curve of the audio component, the update curve including a linear gradient curve, a logarithmic gradient curve and an S-shaped gradient curve;

[0032] Based on the update curve, the processing logic parameters of the audio component are updated to the difference parameter set.

[0033] In addition, to achieve the above objectives, this application also proposes a control device for switching meeting scenarios, the device comprising: a memory, a processor, and a computer program stored in the memory and executable on the processor, the computer program being configured to implement the steps of the control method for switching meeting scenarios as described above.

[0034] In addition, to achieve the above objectives, this application also proposes a storage medium, which is a computer-readable storage medium, on which a computer program is stored, and when the computer program is executed by a processor, it implements the steps of the control method for switching meeting scenes as described above.

[0035] One or more technical solutions proposed in this application have at least the following technical effects:

[0036] The technical solution of this application responds to a scene switching command and obtains the audio component corresponding to the scene switching command; saves the parameter values ​​of the audio component as local variables; obtains the parameter value differences between the local variables and global variables, and generates a difference parameter set based on the parameter value differences. Specifically, when the audio processor is started, the parameter values ​​of the default audio component are obtained and saved as global variables; a DSP function is called to configure the processing logic parameters of the audio component according to the difference parameter set, thereby realizing the meeting scene switching function. This application stores a subset of parameters of scene-related components in local variables, compares them with global variables to generate a difference set, and enables the signal processor to update only the parameters of the target audio component, eliminating the global interruption problem in multi-meeting room scenarios, ensuring no signal jitter during meeting switching and reducing switching latency to milliseconds, while significantly reducing the amount of parameter transmission and further improving the efficiency of scene calling. Attached Figure Description

[0037] The accompanying drawings, which are incorporated in and form part of this specification, illustrate embodiments consistent with this application and, together with the description, serve to explain the principles of this application.

[0038] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, for those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0039] Figure 1 A flowchart illustrating the control method for switching meeting scenarios in this application (Example 1);

[0040] Figure 2 This is a schematic diagram of the hardware operating environment involved in the control method for switching meeting scenarios in this application embodiment.

[0041] The purpose, features, and advantages of this application will be further explained in conjunction with the embodiments and with reference to the accompanying drawings. Detailed Implementation

[0042] It should be understood that the specific embodiments described herein are merely illustrative of the technical solutions of this application and are not intended to limit this application.

[0043] To better understand the technical solution of this application, a detailed description will be provided below in conjunction with the accompanying drawings and specific implementation methods.

[0044] The main solution of this application embodiment is as follows: responding to a scene switching command and obtaining the audio component corresponding to the scene switching command; saving the parameter value of the audio component as a local variable; obtaining the parameter value difference between the local variable and the global variable, generating a difference parameter set based on the parameter value difference, wherein, when starting the audio processor, the parameter value of the default audio component is obtained and the parameter value is saved as a global variable; calling the DSP function to configure the processing logic parameters of the audio component according to the difference parameter set, thereby realizing the meeting scene switching function.

[0045] Since the existing scenario configuration file contains all algorithm parameters, when a single audio processor serves multiple conference rooms, any conference room calling the scenario will trigger a global parameter update. The DSP needs to reload all parameters, which will cause the audio processing flow of other conference rooms to be forcibly interrupted, resulting in signal jitter or brief silence, affecting the continuity of the meeting.

[0046] This application provides a solution that stores a subset of parameters of scene-related components in local variables, compares them with global variables to generate a difference set, and enables the signal processor to update only the parameters of the target audio component. This eliminates the global interruption problem in multi-conference scenarios, ensures no signal jitter during conference switching and reduces switching latency to the millisecond level, while significantly reducing the amount of parameter transmission and further improving the efficiency of calling the scenario.

[0047] Based on this, embodiments of this application provide a control method for switching meeting scenarios, referring to... Figure 1 and Figure 2 , Figure 1 This is a flowchart illustrating steps S10-S40 of the first embodiment of the meeting scene switching control method of this application. The meeting scene switching control method includes steps S10-S40:

[0048] Step S10: In response to the scene switching command, obtain the audio component corresponding to the scene switching command;

[0049] Step S20: Save the parameter values ​​of the audio component as local variables;

[0050] Step S30: Obtain the parameter value difference between the local variable and the global variable, and generate a difference parameter set based on the parameter value difference. In this step, when the audio processor is started, the parameter value of the default audio component is obtained and the parameter value is saved as a global variable.

[0051] Step S40: Call the DSP function to configure the processing logic parameters of the audio component according to the set of difference parameters, thereby realizing the meeting scene switching function.

[0052] In this embodiment, in the traditional architecture of multiple conference rooms sharing a single audio processor, which is an ARM+DSP heterogeneous architecture, where the ARM is used for system management and scheduling and the DSP is used for hardware real-time audio processing, frequent parameter updates in a multi-conference room scenario will significantly reduce system availability. Therefore, the current audio processor, in its application, loads a scene startup file, saves the scene startup file to a global variable, and then starts the conference scene application function.

[0053] Based on this, the system receives user-initiated meeting scene switching commands and uses the corresponding audio scene for the meeting area. The scene switching command is an operation command that triggers the audio processor to switch from the current meeting scene to a new scene. This is achieved through user interface button triggering or automatic triggering by a preset timer, and is used to initiate the scene switching process.

[0054] Therefore, when responding to a scene switching command and obtaining the audio component for that command, the audio component can be determined based on the scene configuration file. This scene configuration file contains the audio component used in the meeting scene corresponding to the scene switching command. Specifically, the scene configuration file stores the audio component parameter values ​​for a specific scene; it is a JSON or XML format file. Different meeting areas have corresponding scene configuration files for their respective meeting scenes. Each scene configuration file only contains audio component parameters directly associated with the corresponding meeting scene, avoiding redundant data storage. Specifically, based on the configuration of this scenario configuration file, the corresponding conference room scenario hardware deployment can be divided into two cases. In one case, conference scenario one can be configured with three components: noise gate, compressor, and automatic gain control, while conference scenario two can be configured with two components: audio manager and limiter. In this hardware deployment, different scenarios can be switched based on the included audio components without interference. In the other hardware deployment, both conference scenario one and conference scenario two have mixer components. The mixer component of conference scenario one only has mixing parameters for microphones 1-8, while the mixer component of conference scenario two includes mixing parameters for microphones 9-16. By partially modifying the audio components, the switching of conference scenarios can be achieved without interference.

[0055] Based on this, the obtained audio components are stored as local variables. These local variables are memory variables that temporarily store the parameter values ​​in the scene configuration file. They are implemented using a storage space independent of global variables and are isolated from other scene parameters to prevent data conflicts.

[0056] The global variable refers to a persistent variable that stores the parameter values ​​of all currently active audio components. It is implemented using non-volatile memory or a database and is used to maintain the complete parameter state during system operation. The difference parameter set refers to a set of difference parameters generated by comparing local variables and global variables. It uses a hash table or bitmap data structure to record the changed items, and only the changed parameter values ​​are transmitted and executed, reducing the amount of data processing.

[0057] This application generates a set of difference parameters by comparing local and global variables, updating only the parameters of the audio components that have changed, rather than reloading all parameters globally. This mechanism effectively avoids signal interruption caused by global parameter refresh when switching between multiple conference room scenarios, ensuring the continuous and stable operation of the audio processing workflow in scenarios that are not switched.

[0058] Specifically, the working process and principle of this application are as follows: The audio processor responds to a scene switching command based on the currently initiated meeting scene. It obtains the scene configuration file corresponding to the scene switching command and stores it as a local variable. The scene configuration file only contains parameter values ​​of audio components associated with the current scene, thus reducing the amount of data to be processed. The local variable is compared with global variables, and the target audio component whose parameter value has changed is determined based on the comparison result. This step can accurately locate the parameters that need to be updated, avoiding unnecessary global parameter updates. Finally, the parameter values ​​of the target audio component are generated into a difference parameter set and sent to the signal processor for execution, realizing the switching of the current scene. This incremental update method can significantly reduce data transmission volume and processing time.

[0059] In a preferred embodiment, the audio processor operates within the currently initiated meeting scenario. Upon receiving a scenario switching command, the audio processor responds to the command. The audio processor retrieves the scenario configuration file corresponding to the scenario switching command from a preset storage location. The scenario configuration file is stored in JSON or XML format and contains only audio component parameter values ​​related to the current scenario. The audio processor parses the retrieved scenario configuration file and stores it as a local variable. The audio processor maintains a global variable to store all currently effective audio component parameter values. The audio processor compares the local variable with the global variable item by item. During the comparison, the audio processor matches the unique identifiers of the audio components, comparing whether the parameter values ​​under the same identifier are consistent. For audio components with inconsistent parameter values, the audio processor marks them as target audio components. The audio processor extracts the new parameter values ​​of all target audio components and generates a difference parameter set. The difference parameter set is in binary format and contains audio component identifiers and corresponding new parameter values. The audio processor sends the difference parameter set to the signal processor via the SPI interface. The signal processor receives the difference parameter set, parses out the new parameter values ​​of the target audio components, updates the corresponding processing logic parameters, and completes the scenario switching.

[0060] This embodiment achieves efficient scene switching for the audio processor in a multi-conference room scenario. By transmitting only the changed parameters, the amount of data transmitted is significantly reduced, shortening the parameter update time. This incremental update mechanism avoids audio processing interruptions caused by global parameter overwriting, ensuring the continuity and stability of audio signals in other conference rooms. Simultaneously, the method of precisely locating changed parameters improves the accuracy of parameter updates and reduces unnecessary processing resource consumption. Furthermore, by separating the storage of global and local variables, the parameter management process is simplified, improving the system's maintainability and scalability.

[0061] Furthermore, before responding to a scene switching command based on the currently initiated meeting scene, a default audio component is obtained and its parameter values ​​are saved as global variables. The combination of the default audio components can be represented as a scene startup file, which includes the parameter values ​​of the default audio components in the signal topology diagram. The parameter set of the global variables is sent to the signal processor for execution. A global variable storage space and multiple local variable storage spaces are provided in a preset storage area. Specifically, starting the meeting scene essentially means using all applied audio components as the default audio components. That is, the scene startup file contains a complete parameter set of all audio components in the signal topology diagram, ensuring that the parameters in the global variable storage space cover all components. The global variable storage space is physically isolated from the multiple local variable storage spaces to avoid data interference between different scene configurations. When the parameter set is sent to the signal processor, a binary structure transmission method is used to reduce data parsing time. The space allocation strategy of the preset storage area adopts a fixed address mapping mechanism, and the address ranges of the global variable storage space and the local variable storage space do not overlap.

[0062] Specifically, during the meeting scenario initialization phase, the scenario startup file is loaded into the global variable storage space via the ARM-based storage chip, forming a benchmark dataset containing parameters of all audio components. The signal processor receives the global variable parameter set via the SPI interface, writes the parameter values ​​into the DSP-based register, and completes the initialization configuration of the audio processing algorithm. The complete parameter set stored in the global variable storage space provides a comparison benchmark for subsequent scenario switching, while the local variable storage space only stores differentiated parameters associated with the current scenario. When performing scenario switching, the comparison operation between the local and global variables is implemented through parameter identifier indexing, requiring only traversal of parameter entries in the local variable storage space, reducing the amount of data processing. The global variable storage space uses non-volatile storage media to ensure that parameter data is not lost after power failure, while the local variable storage space uses high-speed cache media to support fast switching between multiple scenarios.

[0063] In practice, upon startup, the audio processor reads a scene startup file from the storage chip. This file contains initial parameter values ​​for all audio components in the signal topology diagram. The audio processor saves these parameter values ​​to global variables, which are located in the global variable storage space of a preset storage area. The audio processor then sends the complete set of parameters from the global variables to the signal processor via the SPI interface. The signal processor receives and loads these parameters, completing the initialization configuration.

[0064] In addition, besides the global variable storage space, multiple local variable storage spaces are also provided in the preset storage area. These local variable storage spaces are used to temporarily store local variables for different meeting scenarios, facilitating quick switching and comparison.

[0065] For example, when a user needs to switch from "Meeting Mode" to "Tea Break Mode," the audio processor first reads the scene configuration file or audio component set for "Tea Break Mode" and stores it in a free local variable storage space. Then, the audio processor compares this local variable with global variables, identifying only the parameters that have changed. The audio processor only sends the changed parameters to the signal processor, achieving efficient scene switching.

[0066] This embodiment enables an efficient scene switching process. By using global variables to store the complete configuration and local variables to store the current scene configuration, parameters that need to be updated can be quickly identified. This reduces data transmission volume, shortens scene switching time, and improves system response speed. Simultaneously, since only changed parameters are updated, other unchanged audio processing flows are not interrupted, effectively avoiding signal jitter or brief silence issues, and ensuring the continuity and stability of audio processing across multiple conference rooms.

[0067] Furthermore, based on the applied global variables, users can change the parameter values ​​of audio components via the web input page to ensure the meeting scenario meets the current startup requirements. However, updating these global parameters will forcibly interrupt the audio processing flow in all meeting rooms, causing signal jitter or brief silence, affecting meeting continuity. If a user modifies the parameters of a single audio component and directly triggers a global parameter update, it will disrupt the audio processing flow in other meeting rooms.

[0068] Based on this, in response to the audio component parameter modification command, the corresponding audio component parameter value is obtained, and the audio component parameter value is saved to a global variable to generate an instantaneous difference set of audio component parameter values, which is then sent to the signal processor for execution. This allows for rapid modification of audio component parameters without the need for repeated startups, thus improving the efficiency of starting up meeting scenarios.

[0069] The audio component parameter modification command is triggered by the user through a web interface or an external control signal. The parameter value includes the component identifier and the modified value. The global variable is stored in the ARM's memory chip, and specifically, the data structure of the global variable is synchronized with the parameter set in the scene startup file. The real-time difference set only contains the identifier and value of the modified parameter and is transmitted to the DSP via the SPI interface. After parsing the difference set, the signal processor only updates the processing logic parameters of the target audio component, without reloading all parameters.

[0070] Specifically, when a user adjusts the microphone gain parameter in a conference room, the system captures the parameter modification command and extracts the gain value and the corresponding microphone identifier. The gain value is written into the corresponding microphone parameter item in a global variable to ensure that the modified value is correctly inherited during subsequent scene switching. Simultaneously, a difference set data packet containing only the microphone identifier and gain value is generated. After receiving the difference set, the DSP locates the corresponding microphone processing module in the signal topology diagram and directly updates its gain register value. Since other unmodified parameters are not included in the difference set, the DSP does not need to interrupt other ongoing conference room audio processing flows. For example, when the gain value is adjusted from 0dB to +3dB, the difference set only transmits this 3dB increment value, and the DSP completes the parameter update within 20ms, avoiding any perceptible audio interruption.

[0071] In a specific implementation, the system responds to audio component parameter modification commands and obtains the corresponding audio component parameter values. For example, during a meeting, if a user adjusts the microphone gain via the control panel, the system receives the gain adjustment command.

[0072] The audio component parameter values ​​are saved to global variables. Specifically, the system updates the corresponding parameter position in the global variables with the new microphone gain value. An instantaneous set of differences in the audio component parameter values ​​is generated and sent to the signal processor for execution. Further, the system compares the new and old gain values, generates a difference data packet containing only the changed parameters, and sends it to the DSP chip in real time via the SPI interface.

[0073] Based on this, this application enables real-time adjustment of audio parameters in a single conference room without interrupting audio processing in other conference rooms. This improves system flexibility and user experience while avoiding audio interruptions caused by global parameter updates. Furthermore, by transmitting only the difference data, the amount of data transmitted is reduced, thus improving system response speed.

[0074] Furthermore, considering the efficiency of comparing local and global variables, i.e., the step of obtaining the parameter value differences between the local and global variables and generating a set of difference parameters for the parameter value differences, includes:

[0075] The parameter values ​​of the same audio component are extracted from the local variables and the global variables respectively, and the target audio components with different parameter values ​​are located by using a hierarchical hash algorithm.

[0076] The difference parameter set is generated based on the differences in parameter values ​​of the target audio component.

[0077] This embodiment addresses the problem that when determining target audio components by comparing local and global variables, the lack of structured indexing and consistency comparison of parameter groups can lead to inefficient parameter comparison, inaccurate identification of actually changed audio components, and consequently, redundant parameters in the difference set. This increases the processing burden on the signal processor and affects the real-time performance of scene switching. Therefore, this embodiment uses audio component identifiers as a benchmark, extracts parameter values ​​of the same audio component from the local and global variables, and filters out audio components with identical parameter values ​​to obtain target audio components with inconsistent parameter values.

[0078] The audio components are uniquely identified by an audio component identifier, which is generated using hexadecimal encoding rules. Each identifier corresponds to an independent functional module in the signal topology diagram; for example, a reverb unit is identified as 0xA1, and a noise reducer as 0xB2. The parameter group contains all configurable parameters of the component, including gain, filter coefficients, and delay time. In the specific comparison process, a hierarchical hash algorithm is used to define target audio components with different parameter values. Specifically, a hash table is used to establish a mapping relationship between audio components and parameter groups, and the hierarchical hash function uses the FNV-1a algorithm for fast lookup. Consistency comparison employs a byte-by-byte comparison mechanism. When any difference in parameter value is detected, the audio component corresponding to that parameter value is determined as the target to be updated. Furthermore, a linked list structure is used to dynamically remove nodes that match the hash table to achieve filtering, while retaining the parameter values ​​of the differing nodes to generate the target list.

[0079] Specifically, the global variable storage space stores all parameters currently in operation, while the local variable storage space stores the configuration parameters for the scene to be switched. When performing parameter comparison, the audio component identifiers in the local variable storage space are traversed, and the corresponding parameter group in the global variables is quickly located using a hash table. Each audio component's parameter group is encapsulated into an independent data packet, with the packet header containing an identifier and parameter length information. For example, the reverb parameter packet contains 32 bytes of reflection time parameters and 16 bytes of attenuation coefficient. By comparing the payload areas of the two data packets byte by byte, when the gain value in the 5th byte changes from 0x7F to 0x8A, the component is immediately added to the target list. After traversing all identifiers, the target list only contains the identifiers of components whose parameters have changed, such as the components corresponding to identifiers 0xA1 and 0xC3. The signal processor only needs to load the difference parameters of these two components, avoiding the processing delay caused by reloading all parameters, thereby ensuring the continuity of each audio channel when switching between multi-conference room scenes.

[0080] In practical applications, the parameter values ​​of the audio component identifier are extracted from both local and global variables based on the audio component identifier, and parameter groups are generated on a per-audio-component basis. For example, for the audio component "EQ1", parameter groups identified as "EQ1" are searched in both local and global variables. The consistency of parameter values ​​in parameter groups under the same audio component identifier is compared. Specifically, the parameter values ​​of "EQ1" in the local variables are compared one by one with the parameter values ​​of "EQ1" in the global variables.

[0081] The target audio component is obtained by filtering audio components with consistent parameter values. If a parameter value of "EQ1" is inconsistent between local and global variables, then "EQ1" is determined as the target audio component.

[0082] Based on this, it is possible to quickly locate audio components whose parameters have changed, avoiding a full update of all audio components. This targeted parameter update method reduces data transmission volume, improves scene switching efficiency, and reduces interference with other unchanged audio components, ensuring the continuity and stability of audio processing.

[0083] In addition, considering the impact of environmental noise on the audio effects of the meeting scene to be switched between the current meeting scene and the meeting scene to be switched, a set of difference parameters for the audio components is generated by obtaining the floating threshold of the audio components. That is, the step of generating the set of difference parameters based on the parameter value differences of the target audio components includes:

[0084] Obtain the floating threshold of the audio component, wherein the floating threshold is calculated by obtaining the threshold calculation model of the audio component;

[0085] The difference parameter set is obtained by the difference in parameter values ​​and the floating threshold.

[0086] In this embodiment, a corresponding floating threshold is obtained according to the component type of the audio component. The floating threshold can also be defined as the dynamic threshold of the audio component. Specifically, a corresponding threshold calculation model is set according to the component type of the audio component. The component type of the audio component is defined in the form of parameter values, including continuous parameters, discrete parameters, and correlated parameters. That is, a corresponding threshold calculation model is set according to the parameter value representation of the audio component, and the floating threshold of the audio component is calculated according to the threshold calculation model. Specifically, based on the continuous parameters, the threshold calculation model is based on basic sensitivity plus environmental factors, for example, volume parameter: basic 0.5dB, amplified to 1.2dB in a noisy environment. The threshold calculation model for the discrete parameters is to directly compare the original values, for example, a strict match between on / off states (0 / 1). The threshold calculation model for the correlated parameters is limited to the following: if the upstream parameter changes, the downstream parameter threshold is reset to zero (forced update), for example, when the EQ center frequency changes, the Q value is updated accordingly.

[0087] Based on this, a floating threshold for the audio component is obtained according to the threshold calculation model of the audio component, and a difference parameter set is calculated based on the floating threshold and the difference in parameter values. The difference in parameter values ​​is essentially the numerical difference between the parameter values ​​of the same audio component in global and local variables, or a difference in processing logic parameters of the audio component, such as a difference in level. Therefore, the difference parameter values ​​generated based on the parameter value differences and the floating threshold are essentially adjustment values ​​based on switching the processing logic parameters of the current audio component in the current application scenario to the processing logic parameters of another meeting scenario.

[0088] Considering the readability and execution efficiency of the scene configuration file (i.e., the DSP executes the scene configuration file when it needs to be sent), and taking into account the execution efficiency caused by the amount of data sent, the scene configuration file undergoes format conversion. Specifically, the step of saving the parameter values ​​of the audio component as local variables includes:

[0089] Obtain the binary data conversion rule corresponding to the audio component from the preset parameter mapping table, wherein the preset parameter mapping table stores the mapping relationship between all audio components and their corresponding binary data conversion rules in advance;

[0090] Based on the binary data conversion rules, the parameter values ​​of the audio component are converted into binary data blocks, and the binary data blocks are saved as local variables.

[0091] In this embodiment, the parameter values ​​stored in the global and local variables are transmitted and parsed in text format. This causes the signal processor to consume additional resources for format conversion when executing the difference parameter set, affecting the real-time performance and efficiency of scene switching. Therefore, binary data blocks corresponding to the parameter values ​​of the audio components can be obtained from a preset parameter mapping table. The obtained binary database is then used to generate a binary format file stored as a binary structure. This binary structure includes global and local variables, i.e., global and local variables based on the binary structure.

[0092] The preset parameter mapping table establishes a mapping relationship between parameter values ​​and binary data blocks through unique parameter identifiers. The data type and storage length of each parameter value are predefined in the mapping table. The binary data blocks represent parameter values ​​using a fixed-length byte sequence, eliminating redundant characters in the text format. The binary structure stores global and local variables using a contiguous memory allocation method, and its storage space layout is aligned with the memory access mode of the signal processor.

[0093] Specifically, when global or local variables need to be generated, the parameter values ​​are converted into corresponding binary data blocks according to the parameter mapping table. For example, a floating-point parameter with a gain of -3.5dB corresponds to a 4-byte binary data block in the mapping table, which is then converted to hexadecimal representation using the IEEE 754 standard. Multiple binary data blocks are concatenated in order of parameter identifiers to form a binary format file, which is directly stored in the storage space as a binary structure. When the signal processor executes the difference parameter set, it directly reads the parameter values ​​of the target audio component in the binary structure through memory address offsets, without the need for text parsing or data type conversion. This processing mechanism can reduce the processing time of a single parameter from milliseconds to microseconds, making it particularly suitable for conference switching scenarios involving hundreds of parameters.

[0094] In the specific conversion process, the preset parameter mapping table can be implemented using a hash table data structure. The keys in the hash table are parameter identifiers of the audio components, and the values ​​are the corresponding binary data blocks. For example, the audio gain parameter can be mapped to a 16-bit binary data block.

[0095] Furthermore, the binary format file can use a custom file format, such as the ".bin" extension. The file header contains metadata such as version information and timestamps, followed by the binary data of global and local variables.

[0096] Specifically, the binary structure can be designed to contain two main fields: a global variable array and a local variable array. Each array element corresponds to a set of parameters for an audio component. For example:

[0097] struct BinaryStruct {

[0098] GlobalVar globalVars[MAX_COMPONENTS];

[0099] LocalVar localVars[MAX_COMPONENTS];

[0100] };

[0101] GlobalVar and LocalVar are custom structures used to store parameters for each audio component.

[0102] This embodiment achieves efficient storage and fast access to audio component parameters. Converting parameter values ​​into binary data blocks reduces data storage space and improves data transmission efficiency. A pre-defined parameter mapping table simplifies the conversion process from parameter values ​​to binary data. Organizing the binary data into structured binary files facilitates data exchange between the ARM and DSP. By distinguishing between global and local variables, differentiated updates during scene switching are implemented, avoiding unnecessary global parameter reloading and thus reducing audio processing interruptions and signal jitter.

[0103] As shown above, based on a pre-created preset parameter mapping table, in this embodiment, since the data types and storage lengths of different parameters are not clearly identified, the signal processor needs to traverse all binary data blocks for dynamic parsing, resulting in parameter loading delay and low data parsing efficiency.

[0104] Therefore, by setting the parameter values ​​of the audio components, corresponding binary data conversion rules are set, and a preset parameter mapping table is generated based on the correspondence between the parameter values ​​and the binary data conversion rules. The binary data conversion rules are based on the parameter value conversion rules of the corresponding audio components. Therefore, the preset parameter mapping table includes a unique parameter identifier, parameter data type, and parameter storage length, thereby limiting the binary data block.

[0105] The unique identifiers for the parameters are implemented using integer values, with each identifier corresponding to a unique parameter item for a specific audio component. The parameter data types are defined through an enumeration type, including three basic data types: floating-point, integer, and boolean. The parameter storage length is recorded in bytes; the storage length for floating-point parameters is fixed at 4 bytes, while the storage length for integer parameters is set to 2 or 4 bytes depending on the value range. The preset parameter mapping table is stored using a hash table structure, establishing a key-value pair index relationship through the unique parameter identifiers.

[0106] Specifically, when the signal processor receives a set of difference parameters, it directly locates the starting address and parsing rules of the binary data block according to the preset parameter mapping table. For example, the gain parameter of an audio component is identified as 0x0001 in the mapping table, its data type is floating-point, and its storage length is 4 bytes. The signal processor reads 4 bytes of data from the offset address 0x0010 of the binary structure and parses it in floating-point format. Through the predefined data structure, the signal processor does not need to dynamically determine the data type and directly converts the binary data into the target parameter value. This conversion process can shorten the parameter parsing time to a fixed period, avoid processing delays caused by data type differences, and ensure uninterrupted audio signal transmission during scene switching.

[0107] Specifically, a corresponding binary data block is assigned to the parameter values ​​of the audio components. A unique binary data block is allocated to each audio component's parameter value. For example, for volume parameters, 8 bits of binary data can be used to represent volume levels from 0 to 255. For equalizer parameters, 16 bits of binary data can be used to represent the gain values ​​for different frequency bands.

[0108] A preset parameter mapping table is generated based on the correspondence between the parameter values ​​and binary data blocks. The preset parameter mapping table includes a unique identifier for each parameter, the parameter data type, and the parameter storage length. In a specific implementation, an array of structures can be created, each structure containing the following fields:

[0109] Parameter unique identifier: represented by a 32-bit unsigned integer;

[0110] Parameter data type: Use 8-bit unsigned integers, such as 0 for integer, 1 for floating-point, etc.

[0111] Parameter storage length: The number of bytes occupied by the parameter is represented by a 16-bit unsigned integer.

[0112] By setting the preset parameter mapping table, efficient mapping between audio component parameters and binary data is achieved. The establishment of this preset parameter mapping table simplifies the parameter transmission and parsing process, reducing data transmission volume. Simultaneously, the unified data structure definition improves the standardization and scalability of parameter management. This method effectively reduces communication overhead between the ARM and DSP, accelerates scene switching speed, and enhances the overall performance of the audio processor.

[0113] Furthermore, scene configuration files can be configured based on the currently applicable meeting areas. That is, by determining the environmental parameters of each meeting room, the default parameter values ​​of the corresponding audio components are changed according to the environmental parameters, and the audio components with changed default parameters are used to generate the scene configuration file for that meeting room.

[0114] The environmental parameters include at least one of the following: conference room space dimensions, wall sound absorption coefficient, and background noise level, acquired through acoustic sensors deployed in the conference room or through manual input. The default parameter values ​​for the audio components include at least one of the following: microphone array gain coefficient, filter cutoff frequency, and reverberation cancellation algorithm decay time. The scene configuration file stores parameter values ​​using a key-value pair structure, where the key is the audio component identifier and the value is the adjusted parameter value.

[0115] Specifically, when the meeting rooms where the audio processors are deployed have significant spatial differences—for example, the first meeting room has an area of ​​50 square meters and a background noise level of 35 dB, while the second meeting room has an area of ​​200 square meters and a background noise level of 45 dB—after obtaining these environmental parameters through measurement or an input interface, the system automatically adjusts the decay time of the reverberation cancellation algorithm from the default 1.2 seconds to 0.8 seconds for the first meeting room and 1.5 seconds for the second. Simultaneously, it adjusts the threshold of the noise reduction component from the default 40 dB to 30 dB for the first meeting room and 38 dB for the second. The adjusted parameter values ​​are saved as scene configuration files for the corresponding meeting rooms, such as "Meeting Room A.json" and "Meeting Room B.json". When a scene configuration file for a specific meeting room is called, only the set of differences matching the environment of that meeting room needs to be transmitted, avoiding interruptions to audio processing in other meeting rooms due to global parameter updates. For example, when meeting room A switches to tea break mode, only the parameter differences for the reverberation cancellation algorithm and noise reduction component need to be transmitted, while the audio processing flow of meeting room B remains unaffected.

[0116] The above is an example of setting a scene configuration file based on the conference room environment. Specifically, the above example essentially involves determining the environmental parameters of each conference room and changing the default parameter values ​​of the corresponding audio components according to these parameters. These environmental parameters include the conference room area, shape, wall material, seating layout, etc. For example, for a larger conference room, the volume gain and reverberation time can be increased; for a rectangular conference room, the left and right channel balance can be adjusted; and for wall materials with good sound absorption, high-frequency compensation can be appropriately increased.

[0117] Based on this, a scene configuration file for the conference room is generated using the audio component with modified default parameters. Specifically, the adjusted audio component parameter values ​​are written into a JSON-formatted configuration file, with each conference room corresponding to an independent scene configuration file. The scene configuration file contains the identifier of the audio component and its corresponding parameter values.

[0118] Furthermore, multiple scenario modes can be preset for different types of meetings, such as "presentation mode," "discussion mode," and "video conferencing mode." The audio component parameter values ​​for each mode are optimized and adjusted according to the meeting room environment parameters.

[0119] Based on this, customized scene configuration files can be generated for the specific environmental characteristics of each meeting room, improving the accuracy and adaptability of audio processing. This enables automatic optimization of audio parameters for different meeting room environments, enhancing the accuracy and adaptability of audio processing. Simultaneously, generating customized scene configuration files simplifies the audio system debugging process and improves system configuration efficiency. Furthermore, multiple preset meeting modes further enhance the flexibility of the audio system, enabling it to quickly adapt to the needs of different types of meetings.

[0120] Furthermore, based on the calculated difference parameter set generated from the parameter value differences between local and global variables, the meeting scene switching function is achieved by calling the DSP to execute the difference parameter set, i.e., calling the DSP function to configure the processing logic parameters of the audio component according to the difference parameter set, thereby realizing the meeting scene switching function. This includes the following steps:

[0121] Determine the audio component corresponding to the set of differences in parameters, wherein the audio component may be one or more.

[0122] The meeting scene is switched by changing the processing logic parameters of the audio component to the difference parameter set.

[0123] In this embodiment, based on the heterogeneous architecture of the audio processor, the ASP receives scene switching instructions and processes the parameter values ​​of the audio components based on these instructions to obtain a set of differential parameters. The DSP then executes the set of differential parameters to achieve scene switching. Specifically, the DSP essentially performs the process of adjusting the processing logic parameters of the audio components. This process requires parsing the received set of differential parameters to identify the audio components whose processing logic parameters need adjustment, and then adjusting the processing logic parameters of the audio components to the parameter values ​​in the set of differential parameters. That is, upon receiving the set of differential parameters, the parameters are parsed to obtain the audio component identifier, and the parameter values ​​obtained from parsing the set are used to update the processing logic parameters of the target audio component to achieve scene application.

[0124] The DSP receives the difference parameter set via an SPI interface. Specifically, the ASP sends the difference parameter set to the DSP for execution via the SPI interface. This difference parameter set only contains parameter values ​​that have changed in the audio components associated with the current scene. When parsing the difference parameter set, the binary structure is converted into recognizable parameter key-value pairs. For example, binary data blocks are parsed into floating-point numbers or integer values ​​according to a preset parameter mapping table. The audio component identifier is read based on the unique identifier field in the parameter key-value pairs, and a hash table is used to quickly match the target audio component. When updating the parameter values ​​to the target audio component, only the corresponding storage address in the processing logic parameters is modified, keeping the data at other parameter addresses unchanged.

[0125] In a specific implementation, the signal processor (DSP) receives a set of difference parameters from the ARM terminal via an SPI interface. This set of difference parameters is generated by the ARM terminal by comparing local and global variables and contains only audio component information where parameter values ​​have changed. When parsing the difference set, the signal processor converts binary data blocks into specific parameter values ​​according to a preset parameter mapping table; for example, it parses 4 bytes of binary data into gain values. By extracting the identifier field from the parameter key-value pairs, the signal processor locates the corresponding audio component processing logic module in its internal registers. The parsed parameter values ​​are directly written to the parameter storage area of ​​the target audio component; for example, the gain value is updated to the coefficient register of the reverb algorithm, without reloading other unchanged parameters. Thus, the signal processor only performs incremental updates to the difference parameters, avoiding processing interruptions caused by global parameter loading and ensuring the continuity of audio processing.

[0126] In a preferred embodiment, during implementation, the signal processor receives a differential parameter set data packet from the control terminal via the SPI communication interface. This data packet is encapsulated in a binary encoding format. The parsing module decodes the data packet, separating the audio component identifier field and the corresponding parameter value field. Based on the identifier matching rules, the register address space of the target audio component is located in the DSP memory mapping table. The updated parameter value is written to the corresponding register via direct memory access, while triggering a parameter verification mechanism to ensure that the value takes effect immediately after being written. During execution, only parameters with discrepancies are incrementally updated; audio components not affected maintain their original operating state.

[0127] Furthermore, before the step of changing the processing logic parameters of the audio component to the difference parameter set to switch the conference scene, the method further includes:

[0128] Determine the update type of the audio component;

[0129] If the update type is an immediate update component, execute the step of changing the processing logic parameters of the audio component to the difference parameter set to switch the meeting scenario;

[0130] If the update type is a smooth transition component, the step of changing the processing logic parameters of the audio component to the difference parameter set is executed using a pre-set smooth transition mechanism to switch the meeting scene.

[0131] In this embodiment, to address the issue of smooth transitions during audio scene switching, a gradual transition mechanism is implemented. This prevents the direct updating of all audio component parameter values ​​during scene transitions. Instead, it gradually adjusts the audio component parameter values ​​over a short, perceptible time period, achieving a smooth audio transition. Specifically, the audio components are categorized based on their functionality. This categorization includes immediate update types and smooth transition types, where parameter values ​​are updated slowly using a pre-defined smooth transition mechanism, depending on the component's type.

[0132] Specifically, the smooth transition mechanism includes a gradient curve design, which sets a gradient curve for updating parameter values ​​for audio components designated as smooth transition types. The gradient curve includes linear gradients, logarithmic gradients, and S-curves, etc. The most suitable gradient method can be set according to the attribute characteristics of different audio components. That is, if the update type is a smooth transition component, the pre-set smooth transition mechanism is used to execute the step of changing the processing logic parameters of the audio component to the difference parameter set, thereby switching the meeting scene, including:

[0133] Obtain the update curve of the audio component, the update curve including a linear gradient curve, a logarithmic gradient curve and an S-shaped gradient curve;

[0134] Based on the update curve, the processing logic parameters of the audio component are updated to the difference parameter set.

[0135] In this embodiment, a corresponding update curve is set according to the characteristics of the audio component. Specifically, the update curve includes a linear gradient curve, a logarithmic gradient curve, and an S-shaped gradient curve. In another embodiment, the update curve of the audio component can be generated by calling an update curve generator. The update curve generator can select a matching gradient model based on a curve template library, including a linear trajectory model suitable for volume parameters, a logarithmic decay model suitable for equalizer parameters, and an S-shaped gradual change model suitable for reverberation parameters. Furthermore, using the selected model, starting from the current parameter value and ending at the target value of the difference parameter set; or, using the current parameter value combined with the target value generated from the difference parameter set to generate the endpoint parameter value, an intermediate parameter sequence is automatically calculated on a preset transition time axis, and the update curve of the audio component is obtained based on the calculation result.

[0136] Specifically, the calculation process of the update curve incorporates an anti-shake filter to eliminate instantaneous distortion caused by parameter jumps. The final generated parameter sequence is written to the DSP's gradient register through an asynchronous channel, and frame-by-frame interpolation is automatically performed by the hardware, resulting in a scene transition effect that is imperceptible to the ear.

[0137] Furthermore, priorities are set for different types of audio parameters, thereby prioritizing smooth transition updates for parameters of key audio components. Specifically, if the update type is a smooth transition component, a pre-set smooth transition mechanism is used to change the processing logic parameters of the audio component to the differential parameter set, thereby switching the meeting scenario. This includes:

[0138] The update priority is determined based on the component type of the audio component;

[0139] The processing logic parameters of the audio component are changed to the set of differences in sequence according to the update priority.

[0140] In this embodiment, after determining the corresponding audio components whose processing logic parameters need to be updated by parsing the difference parameter set, a priority mapping table is queried according to the audio components to determine the execution order of different audio components in the parameter update sequence. In the priority mapping table, speech enhancement components can be set to the highest priority to ensure that the clarity of human voices remains stable during scene switching; environmental sound effect components can be set to the second priority, allowing their parameters to be adjusted with a lag. Specifically, relevant limitations can be made based on the actual meeting scenario.

[0141] Based on this, according to the determined priorities of the audio components, an ordered audio component update queue is generated by the scheduler, and the difference parameter sets are sequentially injected into the parameter buffers of each audio component in the audio component update queue. Changes to the processing logic parameters of high-priority audio components take effect within a single audio processing cycle, while the processing logic parameters of low-priority components are updated in batches using a time-slice rotation method. This avoids processing overload caused by concurrent updates of multiple components and maintains the continuity of the conference audio stream.

[0142] The above describes the specific implementation process of this application embodiment. This application stores a subset of parameters of the scene-related components in local variables, compares them with global variables to generate a difference set, and enables the signal processor to update only the parameters of the target audio components, eliminating the global interruption problem in multi-conference scenarios, ensuring no signal jitter during conference switching and reducing the switching latency to the millisecond level, while significantly reducing the amount of parameter transmission and further improving the efficiency of calling scenarios.

[0143] It should be noted that the above examples are only for understanding this application and do not constitute a limitation on the control method for switching meeting scenarios in this application. Any simple modifications based on this technical concept are within the protection scope of this application.

[0144] This application provides a control device for switching meeting scenarios. The control device for switching meeting scenarios includes: at least one processor; and a memory communicatively connected to the at least one processor; wherein the memory stores instructions that can be executed by the at least one processor, and the instructions are executed by the at least one processor to enable the at least one processor to execute the control method for switching meeting scenarios and the signal execution method for switching scenarios in the first embodiment described above.

[0145] The following is for reference. Figure 2 The diagram illustrates a structural schematic of a control device suitable for implementing meeting scene switching in the embodiments of this application. The control device for meeting scene switching in the embodiments of this application may include, but is not limited to, mobile terminals such as mobile phones, laptops, digital broadcast receivers, PDAs (Personal Digital Assistants), PADs (Portable Application Description), etc., and fixed terminals such as desktop computers. Figure 2 The control device for switching meeting scenarios shown is merely an example and should not impose any limitations on the functionality and scope of use of the embodiments of this application.

[0146] like Figure 2As shown, the control device for switching conference scenes may include a processing unit 1001 (e.g., a central processing unit, a graphics processing unit, etc.), which can perform various appropriate actions and processes according to a program stored in a read-only memory (ROM) 1002 or a program loaded from a storage device 1003 into a random access memory (RAM) 1004. The RAM 1004 also stores various programs and data required for the operation of the control device for switching conference scenes. The processing unit 1001, the ROM 1002, and the RAM 1004 are interconnected via a bus 1005. An input / output (I / O) interface 1006 is also connected to the bus. Typically, the following systems can be connected to I / O interface 1006: input devices 1007 including, for example, touchscreens, touchpads, keyboards, mice, image sensors, microphones, accelerometers, gyroscopes, etc.; output devices 1008 including, for example, liquid crystal displays (LCDs), speakers, vibrators, etc.; storage devices 1003 including, for example, magnetic tapes, hard disks, etc.; and communication devices 1009. Communication device 1009 allows the control device for conference scene switching to communicate wirelessly or wiredly with other devices to exchange data. Although the figure shows a control device for conference scene switching with various systems, it should be understood that it is not required to implement or have all the systems shown. More or fewer systems can be implemented alternatively.

[0147] Specifically, according to the embodiments disclosed in this application, the processes described above with reference to the flowcharts can be implemented as computer software programs. For example, embodiments disclosed in this application include a computer program product comprising a computer program carried on a computer-readable medium, the computer program containing program code for performing the methods shown in the flowcharts. In such embodiments, the computer program can be downloaded and installed from a network via a communication device, or installed from storage device 1003, or installed from read-only memory 1002. When the computer program is executed by processing device 1001, it performs the functions defined in the methods of the embodiments disclosed in this application.

[0148] The meeting scene switching control device provided in this application, employing the meeting scene switching control method in the above embodiments, can solve the technical problem of abnormal audio parameter switching in existing multi-meeting room scenarios. Compared with the prior art, the beneficial effects of the meeting scene switching control device provided in this application are the same as those of the meeting scene switching control method provided in the above embodiments, and other technical features in this meeting scene switching control device are the same as those disclosed in the previous embodiment method, and will not be repeated here.

[0149] It should be understood that the various parts disclosed in this application can be implemented using hardware, software, firmware, or a combination thereof. In the description of the above embodiments, specific features, structures, materials, or characteristics can be combined in any suitable manner in one or more embodiments or examples.

[0150] The above description is merely a specific embodiment of this application, but the scope of protection of this application is not limited thereto. Any variations or substitutions that can be easily conceived by those skilled in the art within the scope of the technology disclosed in this application should be included within the scope of protection of this application. Therefore, the scope of protection of this application should be determined by the scope of the claims.

[0151] This application provides a storage medium, which is a computer-readable storage medium having computer-readable program instructions (i.e., a computer program) stored thereon. The computer-readable program instructions are used to execute the control method for switching meeting scenes and the signal execution method for switching scenes in the above embodiments.

[0152] The computer-readable storage medium provided in this application may be, for example, a USB flash drive, but is not limited to, electrical, magnetic, optical, electromagnetic, infrared, or semiconductor systems, devices, or any combination thereof. More specific examples of computer-readable storage media may include, but are not limited to: electrical connections having one or more wires, portable computer disks, hard disks, random access memory (RAM), read-only memory (ROM), erasable programmable read-only memory (EPROM or flash memory), optical fibers, portable compact disk read-only memory (CD-ROM), optical storage devices, magnetic storage devices, or any suitable combination thereof. In this embodiment, the computer-readable storage medium may be any tangible medium containing or storing a program that can be used by or in conjunction with an instruction execution system, system, or device. The program code contained on the computer-readable storage medium may be transmitted using any suitable medium, including but not limited to: wires, optical cables, RF (Radio Frequency), or any suitable combination thereof.

[0153] The aforementioned computer-readable storage medium may be included in the control device for switching conference scenarios; or it may exist independently and not be assembled into the control device for switching conference scenarios.

[0154] The aforementioned computer-readable storage medium carries one or more programs. When the aforementioned one or more programs are executed by the control device for switching meeting scenarios, the control device for switching meeting scenarios implements the technical content of the control method for switching meeting scenarios and the signal execution method for switching scenarios as described above.

[0155] Computer program code for performing the operations of this application can be written in one or more programming languages ​​or a combination thereof, including object-oriented programming languages ​​such as Java, Smalltalk, and C++, and conventional procedural programming languages ​​such as the "C" language or similar programming languages. The program code can be executed entirely on the user's computer, partially on the user's computer, as a standalone software package, partially on the user's computer and partially on a remote computer, or entirely on a remote computer or server. In cases involving remote computers, the remote computer can be connected to the user's computer via any type of network—including a Local Area Network (LAN) or a Wide Area Network (WAN)—or can be connected to an external computer (e.g., via the Internet using an Internet service provider).

[0156] The flowcharts and block diagrams in the accompanying drawings illustrate the architecture, functionality, and operation of possible implementations of systems, methods, and computer program products according to various embodiments of this application. In this regard, each block in a flowchart or block diagram may represent a module, segment, or portion of code containing one or more executable instructions for implementing a specified logical function. It should also be noted that in some alternative implementations, the functions indicated in the blocks may occur in a different order than those indicated in the drawings. For example, two consecutively indicated blocks may actually be executed substantially in parallel, and they may sometimes be executed in reverse order, depending on the functions involved. It should also be noted that each block in the block diagrams and / or flowcharts, and combinations of blocks in the block diagrams and / or flowcharts, can be implemented using a dedicated hardware-based system that performs the specified function or operation, or using a combination of dedicated hardware and computer instructions.

[0157] The modules described in the embodiments of this application can be implemented in software or hardware. The names of the modules do not necessarily limit the functionality of the unit itself.

[0158] The readable storage medium provided in this application is a computer-readable storage medium that stores computer-readable program instructions (i.e., a computer program) for executing the above-described control method for switching meeting scenarios, which can solve the technical problem of abnormal audio parameter switching in existing multi-meeting room scenarios. Compared with the prior art, the beneficial effects of the computer-readable storage medium provided in this application are the same as the beneficial effects of the control method for switching meeting scenarios and the signal execution method for switching scenarios provided in the above embodiments, and will not be repeated here.

Claims

1. A control method of a conference scene switching, characterized by, The conference scene switching control method is applied to an audio processor, and comprises the following steps: in response to a scene switching instruction, obtaining an audio component corresponding to the scene switching instruction; saving a parameter value of the audio component as a local variable; obtaining a parameter value difference between the local variable and a global variable, generating a difference parameter set according to the parameter value difference, wherein a parameter value of a default audio component is obtained when the audio processor is started, and the parameter value is saved as a global variable; calling a DSP function to configure a processing logic parameter of the audio component according to the difference parameter set, thereby realizing a conference scene switching function.

2. The control method of conference scene switching according to claim 1, wherein, The step of saving the parameter value of the audio component as a local variable comprises: obtaining a binary data conversion rule corresponding to the audio component in a preset parameter mapping table, wherein a mapping relationship between all audio components and corresponding binary data conversion rules is pre-stored in the preset parameter mapping table; based on the binary data conversion rule, converting the parameter value of the audio component into a binary data block, and saving the binary data block as the local variable.

3. The control method of conference scene switching according to claim 1, wherein, The step of obtaining the parameter value difference between the local variable and the global variable and generating the difference parameter set of the parameter value difference comprises: extracting the parameter values of the same audio component in the local variable and the global variable respectively, and locating a target audio component with different parameter values through a hierarchical hash algorithm; generating the difference parameter set according to the parameter value difference of the target audio component.

4. The control method of conference scene switching according to claim 3, wherein, The step of generating the difference parameter set according to the parameter value difference of the target audio component comprises: obtaining a floating threshold value of the audio component, wherein the floating threshold value is calculated by obtaining a threshold calculation model of the audio component; obtaining the difference parameter set through the parameter value difference and the floating threshold value.

5. The control method of conference scene switching according to claim 1, wherein, The step of calling a DSP function to configure a processing logic parameter of the audio component according to the difference parameter set, thereby realizing a conference scene switching function, comprises: determining the audio component corresponding to the difference parameter set, the audio component being one or more; changing the processing logic parameter of the audio component to the difference parameter set, thereby switching the conference scene.

6. The control method of conference scene switching according to claim 5, wherein, Before the step of changing the processing logic parameter of the audio component to the difference parameter set, thereby switching the conference scene, further comprising: determining an update type of the audio component; if the update type is an immediate update component, performing the step of changing the processing logic parameter of the audio component to the difference parameter set, thereby switching the conference scene; if the update type is a smooth transition component, performing the step of changing the processing logic parameter of the audio component to the difference parameter set, thereby switching the conference scene, by a pre-set smooth transition mechanism.

7. The control method of conference scene switching according to claim 6, wherein, If the update type is a smooth transition component, performing the step of changing the processing logic parameter of the audio component to the difference parameter set, thereby switching the conference scene, by a pre-set smooth transition mechanism, comprises: determining an update priority according to a component type of the audio component; The processing logic parameters of the audio component are changed to the difference parameter set in sequence by the update priority.

8. The control method of conference scene switching according to claim 6, wherein, If the update type is a smooth transition component, the step of changing the processing logic parameters of the audio component to the difference parameter set to switch the conference scene by a preset smooth transition mechanism, comprising: An update curve of the audio component is obtained, the update curve comprising a linear gradient curve, a logarithmic gradient curve and an S-shaped gradient curve; Based on the update curve, the processing logic parameters of the audio component are updated to the difference parameter set.

9. A control device for conference scene switching, characterized by The conference scene switching control device stores a computer program, and the computer program is executed by a processor to implement the conference scene switching control method of any one of claims 1-8.

10. A storage medium, characterized by The storage medium stores a computer program, and the computer program is executed by a processor to implement the conference scene switching control method of any one of claims 1-8.

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