Control method and device for audio power amplifier
Through technical means such as environmental noise, distance and biometric data, the volume and mode of the audio amplifier are intelligently adjusted, which solves the problems of low management efficiency and insufficient energy efficiency management of traditional audio amplifiers, and achieves more efficient audio management and user experience.
Patent Information
- Application Number
- CN202411337882.0
- Authority / Receiving Office
- CN · China
- Patent Type
- Patents(China)
- Current Assignee / Owner
- Filing Date
- 2024-09-25
- Publication Date
- 2025-01-10
- Estimated Expiration
- 2044-09-25
AI Technical Summary
The management efficiency of traditional audio amplifiers is not high, cannot adapt to different environments and user needs, and insufficient energy efficiency management.
Obtain ambient noise data and audio signal characteristics through the microphone, determine the music type and adjust the volume mode; use ultrasonic sensor to obtain user distance data and adjust the volume in real time; switch to low-power mode when there is no audio input or no one is active; combine biological characteristics Identification and three-dimensional scanning technology, record user preferences and common time periods, optimize audio settings; built-in sensors monitor equipment parameters and warning for potential faults.
It realizes the best auditory effect of volume in different environments and music types, ensuring that users have a unified experience regardless of location, automatic power consumption management, improving system intelligence, reducing user manual intervention, and extending device life.
Smart Images

Figure CN118870253B_ABST
Abstract
Description
Technical Field
[0001] The present application relates to the field of computer technology, and more specifically, to a control method, device, electronic device and storage medium for an audio power amplifier. Background Art
[0002] With the development of science and technology, audio amplifiers, as one of the key components of modern audio systems, have become increasingly diverse in their functions. They must not only meet basic audio amplification needs, but also have a high degree of intelligence and automation.
[0003] Traditional audio amplifiers often rely on manual volume adjustment and amplification mode selection, which is not only complex to operate but also unable to adapt to the ever-changing usage environment. In addition, the energy efficiency management of amplifiers is also a major challenge.
[0004] From the above, it can be seen that how to improve the management efficiency of audio amplifiers still needs to be further resolved. Summary of the invention
[0005] The present application provides a control method, device, electronic device and storage medium for an audio power amplifier, which can solve the problem of low management efficiency of audio power amplifiers in the related art. The technical solution is as follows:
[0006] According to one aspect of the present application, a control method for an audio amplifier includes: obtaining corresponding environmental noise data based on a microphone arranged in the audio, obtaining audio signal characteristics output by the audio, determining the corresponding music type based on the audio signal characteristics, and determining the output volume mode of the audio based on the environmental noise data and the music type; judging whether there is a user in the room, and if there is a user, obtaining the distance data corresponding to the user in real time, and adjusting the output volume mode based on the distance data; after the audio is turned on, controlling the audio to perform a real-time detection mode, and if no audio input or human activity is detected, controlling the audio to switch to a preset low power consumption mode, and if audio input or human activity is detected, controlling the audio to switch to a normal working mode.
[0007] In an exemplary embodiment, when there is a user, the method further includes: obtaining biometric data corresponding to the user, identifying and determining a corresponding target user based on the biometric data; retrieving a corresponding target audio setting from a pre-stored user preference database based on the target user; and switching an output volume mode corresponding to the audio based on the target audio setting.
[0008] In an exemplary embodiment, the method further includes: obtaining commonly used time periods corresponding to when the audio is turned on, and recording the user's playback preferences during the commonly used time periods, wherein there are multiple commonly used time periods, and the playback preferences include the types of music most frequently played by different users and the corresponding sound effect settings; associating the commonly used time periods with the playback preferences, and training the associated commonly used time periods and the playback preferences as training sets for a prediction model, and determining a corresponding prediction model; obtaining a corresponding time series, inputting the time series into the prediction model, obtaining the corresponding predicted user's future usage pattern, and controlling the audio based on the future usage pattern.
[0009] In an exemplary embodiment, in the process of adjusting the output volume mode based on the distance data, the method also includes: acquiring the distance data between the user and the audio system through an ultrasonic sensor; retrieving a preset distance division interval set, matching the distance data with the distance division interval set, and determining a corresponding target distance division interval; determining corresponding target volume interval data based on a mapping relationship between the target distance division interval and a preset mapping relationship, determining the corresponding target volume data through a linear interpolation method, and controlling the output volume of the audio system based on the target volume data, wherein the mapping relationship is a mapping relationship between the distance division interval and the volume interval data.
[0010] In an exemplary embodiment, the method also includes: obtaining a three-dimensional model of the room through a three-dimensional scanning device, and using acoustic simulation software to generate a corresponding sound wave distribution map, wherein the sound wave distribution map includes a sound wave propagation path and a sound field distribution; in the process of adjusting the output volume mode of the audio system, obtaining the corresponding latest sound wave distribution map, and determining the corresponding sound wave distribution change based on the latest sound wave distribution map and the sound wave change map; displaying the sound wave distribution change to the user, obtaining the latest output volume mode set by the user and storing it.
[0011] In an exemplary embodiment, the method further includes: obtaining the device parameters of the audio system in real time through a built-in sensor; matching the device parameters with a preset threshold interval, and if there is a mismatch, determining that a potential fault exists, and sending a warning message to a user or technician.
[0012] According to one aspect of the present application, a control device for an audio power amplifier includes:
[0013] an output volume mode determination module, which obtains corresponding environmental noise data based on a microphone arranged in the speaker, obtains audio signal characteristics output by the speaker, determines a corresponding music type based on the audio signal characteristics, and determines an output volume mode of the speaker based on the environmental noise data and the music type;
[0014] Output volume mode adjustment mode, judging whether there is a user in the room, if there is a user, obtaining the distance data corresponding to the user in real time, and adjusting the output volume mode based on the distance data;
[0015] The mode switching module controls the audio system to perform real-time detection mode after the audio system is turned on. If no audio input or human activity is detected, the module controls the audio system to switch to a preset low-power mode. If audio input or human activity is detected, the module controls the audio system to switch to a normal working mode.
[0016] According to one aspect of the present application, an electronic device includes at least one processor and at least one memory, wherein the memory stores computer-readable instructions; the computer-readable instructions are executed by one or more of the processors, so that the electronic device implements the control method for an audio power amplifier as described above.
[0017] According to one aspect of the present application, a storage medium stores computer-readable instructions thereon, and the computer-readable instructions are executed by one or more processors to implement the control method for an audio power amplifier as described above.
[0018] According to one aspect of the present application, a computer program product includes computer-readable instructions, which are stored in a storage medium. One or more processors of an electronic device read the computer-readable instructions from the storage medium, load and execute the computer-readable instructions, so that the electronic device implements the control method for an audio power amplifier as described above.
[0019] The beneficial effects of the technical solution provided by this application are:
[0020] By identifying environmental noise data and music genres, the volume is ensured to always maintain the best hearing effect in different environments and music genres; the volume is adjusted based on user distance data to ensure a consistent listening experience no matter where the user is in the room.
[0021] It realizes automatic power consumption management and automatically enters low power consumption mode when there is no audio input or no human activity to save energy. It improves the intelligence of the system through real-time detection and intelligent adjustment and reduces the need for manual intervention by users. BRIEF DESCRIPTION OF THE DRAWINGS
[0022] In order to more clearly illustrate the technical solutions in the embodiments of the present application, the following briefly introduces the drawings required for use in the description of the embodiments of the present application. Obviously, the drawings described below are only some embodiments of the present application, and those skilled in the art can obtain other drawings based on these drawings without creative work.
[0023] Figure 1 It is a schematic diagram of the implementation environment involved in this application;
[0024] Figure 2 is a flow chart showing a control method for an audio power amplifier according to an exemplary embodiment;
[0025] Figure 3 is a flow chart from S001 to S003 in a control method for an audio power amplifier according to an exemplary embodiment;
[0026] Figure 4 is a flow chart from S121 to S123 in a control method for an audio power amplifier according to an exemplary embodiment;
[0027] Figure 5 The present invention is a structural block diagram of a control device for an audio power amplifier according to an exemplary embodiment. DETAILED DESCRIPTION
[0028] The embodiments of the present application are described in detail below, and examples of the embodiments are shown in the accompanying drawings, wherein the same or similar reference numerals throughout represent the same or similar elements or elements having the same or similar functions. The embodiments described below with reference to the accompanying drawings are exemplary and are only used to explain the present application, and cannot be interpreted as limiting the present application.
[0029] It will be understood by those skilled in the art that, unless expressly stated, the singular forms "one", "said", and "the" used herein may also include plural forms. It should be further understood that the term "comprising" used in the specification of the present disclosure refers to the presence of the features, integers, steps, operations, elements, and / or components, but does not exclude the presence or addition of one or more other features, integers, steps, operations, elements, components, and / or groups thereof. It should be understood that when we refer to an element as being "connected" or "coupled" to another element, it may be directly connected or coupled to the other element, or there may be intermediate elements. In addition, the "connection" or "coupling" used herein may include wireless connection or wireless coupling. The term "and / or" used herein includes all or any unit and all combinations of one or more associated listed items.
[0030] In order to make the objectives, technical solutions and advantages of the present application clearer, the implementation methods of the present application will be further described in detail below with reference to the accompanying drawings.
[0031] Figure 1 The present invention is a schematic diagram of an implementation environment involved in a control method for an audio power amplifier. The implementation environment includes a collection end and a service end.
[0032] See also Figure 2The present application embodiment provides a control method for an audio power amplifier, the method is applicable to an electronic device, the electronic device may be Figure 1 The server side in the implementation environment is shown.
[0033] In the following method embodiments, for ease of description, the execution subject of each step of the method is taken as an electronic device as an example for illustration, but this does not constitute a specific limitation.
[0034] like Figure 2 As shown, the method may include the following steps:
[0035] S100, obtaining corresponding environmental noise data based on a microphone installed in the speaker, obtaining characteristics of an audio signal output by the speaker, determining a corresponding music type based on the characteristics of the audio signal, and determining an output volume mode of the speaker based on the environmental noise data and the music type.
[0036] Among them, the output volume mode is pre-set in the system; a microphone is installed on the audio equipment to collect environmental noise data, and then signal processing technology (such as FFT transform) can be used to analyze the environmental noise data and extract noise characteristics. The currently playing audio signal is collected by the signal processor inside the audio, and the audio signal can be classified to identify the music type.
[0037] Adjusting the volume according to the ambient noise level can ensure that clear audio content can be heard in different noise environments. In addition, the music type is automatically identified based on the audio signal characteristics, so as to select the sound effect mode that best suits the current music type. At the same time, the volume and sound effects are automatically adjusted according to the ambient noise data and music type, and the system provides a personalized listening experience.
[0038] S110, determining whether there is a user in the room, if there is a user, obtaining distance data corresponding to the user in real time, and adjusting the output volume mode based on the distance data.
[0039] An ultrasonic sensor can be used to detect the presence of a user. When the ultrasonic sensor detects a user, distance data acquisition is initiated, and sound waves can be emitted and reflected sound waves are received, and the round-trip time is calculated to determine the distance. In the presence of a user, by acquiring the user's distance data in real time, the system can sense changes in the user's position and make corresponding adjustments to improve the user experience. When the user is away from the speaker, the volume is automatically lowered to avoid interference caused by unnecessary excessive volume; when the user is close to the speaker, the volume is automatically increased to ensure that the volume is moderate.
[0040] S120, after the audio is turned on, the audio is controlled to perform a real-time detection mode. If no audio input or human activity is detected, the audio is controlled to switch to a preset low power consumption mode. If audio input or human activity is detected, the audio is controlled to switch to a normal working mode.
[0041] Among them, whether there is audio signal input can be monitored to realize audio input detection, and user activity can be detected by motion sensor or infrared sensor. If there is neither audio input nor human activity, it is judged as idle state. When the system detects idle state, it automatically enters low power consumption mode to reduce power consumption; when the system detects audio input or user activity, it automatically switches back to normal working mode; thereby realizing automatic power consumption management, it automatically enters low power consumption mode when there is no audio input or human activity, and saves energy.
[0042] In the embodiment of the present application, the system has multiple preset output volume modes, and the user can participate in the preset output volume mode, such as Figure 3 As shown, the following steps are also performed:
[0043] S001, obtain a three-dimensional model of the room through a three-dimensional scanning device, and use acoustic simulation software to generate a corresponding sound wave distribution map.
[0044] Among them, the sound wave distribution map includes the sound wave propagation path and sound field distribution; first, use a three-dimensional scanning device to comprehensively scan the room, capture the room's geometric structure and physical characteristics, and create a detailed three-dimensional model. This three-dimensional model can reflect the actual size of the room, the location of walls, doors, windows, and other fixtures; next, use acoustic simulation software to simulate the behavior of sound waves in this environment based on the three-dimensional model, and the generated sound wave distribution map not only shows the propagation path of the sound wave from the sound source to the receiving point, but also reveals the sound pressure level at different locations (that is, the sound field distribution).
[0045] S002, in the process of adjusting the output volume mode of the speaker, obtaining the corresponding latest sound wave distribution map, and determining the corresponding sound wave distribution change based on the latest sound wave distribution map and the sound wave change map.
[0046] Among them, when the output mode of the sound system changes, the acoustic simulation is rerun to generate the latest sound wave distribution map, and then the sound wave distribution map is compared with the latest sound wave distribution map to identify changes in the sound wave distribution, such as the sound becoming stronger or weaker in certain areas.
[0047] S003, displaying the sound wave distribution change to the user, obtaining the latest output volume mode set by the user and storing it.
[0048] The changes in sound wave distribution are presented to the user in the form of a graphical interface, such as using color coding to indicate changes in sound pressure level. A user interface is provided to allow the user to adjust the output volume mode, and the sound wave distribution diagram is updated in real time after each adjustment. The optimal output mode selected by the user is stored so that it can be automatically loaded the next time it is used.
[0049] By executing steps S001 to S003, the overall sound quality in the room can be improved by precisely controlling the distribution of sound waves and reducing unnecessary reflections and standing waves. Visual tools can be used to help identify acoustic problem areas in the room, and users can adjust the audio system according to their preferences. The system can remember the user's preferred settings and automatically apply them to future usage scenarios.
[0050] In addition, when a user is detected indoors, the following steps are performed:
[0051] S111, obtaining biometric data corresponding to the user, and identifying and determining the corresponding target user based on the biometric data.
[0052] Among them, when the user approaches the audio system or actively triggers recognition, the device collects the user's biometric data. By using biometric recognition devices (such as fingerprint readers, facial recognition cameras, iris scanners, etc.) to capture the user's biometric information, the system will compare the collected biometric information with the biometric information of registered users in the database to find the record with the highest match.
[0053] Biometric recognition technology can ensure that only authorized users can access the audio system, increasing security; it can also quickly identify specific target users, ensuring that the audio system can automatically adjust to the settings suitable for the user based on different user identities.
[0054] S112: Retrieving corresponding target audio settings from a pre-stored user preference database based on the target user.
[0055] Among them, after the target user is successfully identified, the system queries the audio setting information in the target user's personal preference database, and can retrieve the user's audio preference settings from the server or local database through the identified user ID, and enable the audio system to automatically adapt to the auditory preferences of different users without manual adjustment each time, thereby improving convenience.
[0056] S113, switching the output volume mode corresponding to the sound system based on the target sound system settings.
[0057] Among them, the audio settings obtained from the database are applied to the current audio equipment through the control interface of the audio system (which may be a hardware button, software interface or wireless communication protocol). After receiving the new setting instructions, the audio system adjusts the output volume mode to the specified state; thus ensuring that each user can enjoy the sound effects they are accustomed to, thereby enhancing user satisfaction.
[0058] By executing steps S111 to S113, the user identity is confirmed using biometric recognition technology, and the audio settings are automatically adjusted according to the user's preferences. This not only improves the personalized service level of the audio system, but also strengthens the security of the system.
[0059] At the same time, the system will also perform intelligent control based on the distance between the user and the speaker. In the process of adjusting the output volume mode based on distance data, the specific implementation process includes:
[0060] S114, obtaining distance data between the user and the speaker through an ultrasonic sensor.
[0061] The method for obtaining the distance data has been described above and will not be elaborated here.
[0062] S115, retrieve a preset distance division interval set, match the distance data with the distance division interval set, and determine a corresponding target distance division interval.
[0063] Among them, a set of distance division intervals are pre-defined, for example: 0 to 1 meter, 1 to 2 meters, 2 to 3 meters, 3 to 4 meters, and above 4 meters, and these intervals are stored in the system. Through the preset distance division intervals, the system can quickly find the interval to which the distance data belongs, and can adjust the interval definition according to actual conditions to adapt to different usage scenarios and needs.
[0064] The collected distance data is compared with the preset distance division intervals, and the target distance division interval of the user is determined according to the matching results. The interval to which the distance data belongs is found, which can ensure that the distance data can be accurately mapped to the corresponding interval, thereby providing an accurate basis for subsequent volume adjustment.
[0065] S116, determining corresponding target volume interval data based on the target distance division interval and a preset mapping relationship, determining corresponding target volume data by a linear interpolation method, and controlling the output volume of the audio system based on the target volume data.
[0066] Among them, the mapping relationship is the mapping relationship between the distance division interval and the volume interval data. It should be pointed out here that a mapping table is set in advance to define the volume interval data corresponding to each distance division interval, for example: 0 to 1 meter: the volume is set to 20%; 1 to 2 meters: the volume is set to 40%; 2 to 3 meters: the volume is set to 60%; 3 to 4 meters: the volume is set to 80%; above 4 meters: the volume is set to 100%.
[0067] At the same time, by dividing the intervals according to the target distance, the system can automatically determine the appropriate volume interval through mapping relationships without the need for manual settings by the user. Different mapping relationships can also be pre-set according to the auditory needs of different users to provide personalized volume adjustments.
[0068] If the actual distance between the user and the speaker falls between the two intervals, a linear interpolation method is used to calculate the target volume data. For example, if the user is 1.5 meters away, the system will calculate the target volume data based on the linear relationship between the 1-2 meter interval (40%) and the 2-3 meter interval (60%): , for example, when the user distance is 1.5 meters, the calculation formula is: ,Through the linear interpolation method, the volume can be accurately adjusted through the ,interpolation method even if the user distance changes slightly, ensuring a smooth volume adjustment ,process and avoiding the discomfort caused by abrupt changes.
[0069] Finally, the system monitors the user's distance in real time and dynamically adjusts the volume according to the change in distance. By adjusting the volume in real time, the system can better adapt to the user's movement, thereby providing a more comfortable and personalized audio experience and improving the intelligence level of the system.
[0070] In the embodiments of the present application, Figure 4 As shown, the method also includes:
[0071] S121, obtaining common time periods corresponding to when the audio is turned on, and recording the user's playing preferences in the common time periods.
[0072] There are multiple commonly used time periods, and the playback preferences include the types of music most frequently played by different users and the corresponding sound effect settings.
[0073] Record the timestamp when the speaker is turned on, and count the frequency of users turning on the speaker in different time periods. Based on the statistical data, define the user's "common time periods", such as 6:00-8:00 in the morning, 7:00-10:00 in the evening, etc., and record the types of music that users play most frequently and the corresponding sound effect settings during the common time periods. By recording the user's speaker usage in different time periods, you can understand the user's habits and preferences.
[0074] S122, associating the commonly used time periods with the playback preferences, and using the associated commonly used time periods and the playback preferences as training sets for a prediction model, and determining a corresponding prediction model.
[0075] Among them, commonly used time periods are paired with playback preferences to form a training data set, and then a suitable machine learning model, such as a neural network, is selected. The model is trained using the organized training data set to learn the correlation between commonly used time periods and playback preferences. The accuracy and generalization ability of the model are then evaluated through methods such as cross-validation to ensure that the model can accurately predict future usage patterns.
[0076] By training the model, the system can predict the user's usage patterns in a certain period of time in the future, and based on the prediction results, the system can automatically adjust the settings of the speaker.
[0077] S123, obtaining a corresponding time series, inputting the time series into a prediction model, obtaining a corresponding predicted future usage pattern of the user, and controlling the audio system based on the future usage pattern.
[0078] By obtaining the current time in real time and using it as an input time series, and inputting the current time into the trained prediction model, the system can predict the user's future usage pattern in real time based on the current time. The future usage pattern includes the type of music the user may play and the sound effect settings; based on the predicted future usage pattern, the system automatically adjusts the volume of the speaker, sound effect settings, etc.
[0079] By executing steps S121 to S123, the system can predict future usage patterns based on the user's playback preferences during common time periods and automatically adjust the audio settings accordingly; this not only improves the intelligence level of the audio system, but also significantly improves the user experience, ensuring that users can get the best hearing effects in different time periods.
[0080] After the speaker is used for the first time, follow these steps:
[0081] S124 obtains the device parameters of the audio system in real time through built-in sensors.
[0082] Various built-in sensors, such as temperature sensors, current sensors, vibration sensors, etc., are installed in audio equipment to monitor the operating status of the equipment. These sensors collect the operating parameters of the audio equipment in real time, such as temperature, current, voltage, vibration, etc., and then transmit the collected equipment parameters to the central processing unit or cloud server for processing via wireless or wired means.
[0083] S125, matching the device parameters with the preset threshold intervals, and if they do not match, when it is determined that there is a potential fault, sending a warning message to the user or technician.
[0084] Among them, a series of threshold ranges are pre-set according to the technical specifications of the audio equipment and the manufacturer's recommended values. For example, the upper temperature limit is set to 50°C, the upper current limit is set to 2A, etc.; the device parameters collected in real time are compared with the pre-set threshold ranges to check whether there are any situations beyond the normal range.
[0085] When device parameters exceed the preset threshold range, the system determines that there is a potential fault, generates an early warning message, and sends it to the user or technical support personnel through various means (such as SMS, email, application notification, etc.). The early warning message should include the specific device parameter abnormality, possible cause analysis, and recommended countermeasures.
[0086] During the data collection stage, the temperature sensor: monitors the temperature inside the audio equipment in real time; the current sensor: monitors the working current of the audio equipment in real time; the vibration sensor: monitors the vibration of the audio equipment in real time.
[0087] In the matching and judgment stage, the temperature threshold: set the normal operating temperature range to 20°C to 50°C; the current threshold: set the normal operating current range to 0A to 2A; the vibration threshold: set the normal operating vibration range to 0mm / s to 10mm / s.
[0088] During the execution of the warning information sending stage: if the temperature is detected to be over 50°C, a warning message is sent; if the current is detected to be over 2A, a warning message is sent; abnormal vibration: if the vibration is detected to be over 10mm / s, a warning message is sent.
[0089] The built-in sensors can obtain the operating parameters of the audio equipment in real time to ensure that the system can detect abnormal situations in time. Through the pre-set threshold range, the system can automatically determine whether the equipment parameters are out of the normal range and realize standardized monitoring. When potential faults are found, the system can promptly send early warning information to users or technicians to help take quick measures to avoid greater losses.
[0090] The following is an embodiment of the device of the present application, which can be used to execute the control method for the audio power amplifier involved in the present application. For details not disclosed in the embodiment of the device of the present application, please refer to the method embodiment of the control method for the audio power amplifier involved in the present application.
[0091] See also Figure 5 In the embodiment of the present application, a control device for an audio power amplifier is provided, including but not limited to:
[0092] The output volume mode determination module 200 obtains corresponding environmental noise data based on a microphone installed in the speaker, obtains audio signal characteristics output by the speaker, determines the corresponding music type based on the audio signal characteristics, and determines the output volume mode of the speaker based on the environmental noise data and the music type;
[0093] Output volume mode adjustment mode 210, determining whether there is a user in the room, if there is a user, obtaining the distance data corresponding to the user in real time, and adjusting the output volume mode based on the distance data;
[0094] The mode switching module 220 controls the audio system to perform a real-time detection mode after the audio system is turned on. If no audio input or human activity is detected, the mode switching module 220 controls the audio system to switch to a preset low power consumption mode. If audio input or human activity is detected, the mode switching module controls the audio system to switch to a normal working mode.
[0095] In an exemplary embodiment, the device further includes but is not limited to:
[0096] A target user determination module obtains biometric data corresponding to the user and performs identification based on the biometric data to determine the corresponding target user;
[0097] A target audio setting retrieval module, for retrieving corresponding target audio settings from a pre-stored user preference database based on a target user;
[0098] The output volume mode switching module is used to switch the output volume mode corresponding to the sound based on the target sound setting.
[0099] In an exemplary embodiment, the device further includes but is not limited to:
[0100] A common time period acquisition module is used to obtain the common time period corresponding to the sound on, and record the user's playback preferences in the common time period, where there are multiple common time periods, and the playback preferences include the most frequently played music types and corresponding sound effect settings of different users;
[0101] A prediction model determination module associates the commonly used time periods with the playback preferences, and uses the associated commonly used time periods and playback preferences as training sets for the prediction model, and is used to determine the corresponding prediction model;
[0102] The future usage pattern acquisition module acquires the corresponding time series, inputs the time series into the prediction model, is used to acquire the future usage pattern of the corresponding predicted user, and controls the audio based on the future usage pattern.
[0103] In an exemplary embodiment, the device further includes but is not limited to:
[0104] A distance data acquisition module is used to acquire the distance data between the user and the speaker through an ultrasonic sensor;
[0105] The target distance division interval determination module retrieves a preset distance division interval set, matches the distance data with the distance division interval set, and uses it to determine the corresponding target distance division interval;
[0106] The output volume control module determines the corresponding target volume interval data based on the target distance division interval and the pre-set mapping relationship, determines the corresponding target volume data by a linear interpolation method, and controls the output volume of the audio based on the target volume data, wherein the mapping relationship is a mapping relationship between the distance division interval and the volume interval data.
[0107] In an exemplary embodiment, the device further includes but is not limited to:
[0108] A sound wave distribution map generation module, which obtains a three-dimensional model of the room through a three-dimensional scanning device, and uses acoustic simulation software to generate a corresponding sound wave distribution map, wherein the sound wave distribution map includes a sound wave propagation path and a sound field distribution;
[0109] A sound wave distribution change determination module, which obtains the corresponding latest sound wave distribution map during the process of adjusting the output volume mode of the audio system, and determines the corresponding sound wave distribution change based on the latest sound wave distribution map and the sound wave change map;
[0110] The display module is used to display the sound wave distribution changes to the user, obtain the latest output volume mode set by the user and store it.
[0111] In an exemplary embodiment, the device further includes but is not limited to:
[0112] The device parameter acquisition module is used to obtain the device parameters of the audio system in real time through the built-in sensors;
[0113] The matching module is used to match the device parameters with the preset threshold interval. If there is a mismatch, it is determined that there is a potential fault and an early warning message is sent to the user or technician.
[0114] It should be noted that the control device for the audio power amplifier provided in the above embodiment only uses the division of the above-mentioned functional modules as an example when controlling the audio power amplifier. In actual applications, the above-mentioned functions can be assigned to different functional modules as needed, that is, the internal structure of the control device for the audio power amplifier will be divided into different functional modules to complete all or part of the functions described above.
[0115] In addition, the control device for an audio power amplifier provided in the above embodiment and the embodiment of the control method for an audio power amplifier belong to the same concept, wherein the specific manner in which each module performs operations has been described in detail in the method embodiment and will not be repeated here.
[0116] According to one aspect of the present application, an electronic device includes at least one processor and at least one memory, wherein the memory stores computer-readable instructions; the computer-readable instructions are executed by one or more of the processors, so that the electronic device implements the control method for an audio power amplifier as described above.
[0117] In addition, an embodiment of the present application provides a storage medium on which computer-readable instructions are stored. The computer-readable instructions are executed by one or more processors to implement the control method for an audio power amplifier as described above.
[0118] In an embodiment of the present application, a computer program product is provided. The computer program product includes computer-readable instructions. The computer-readable instructions are stored in a storage medium. One or more processors of an electronic device read the computer-readable instructions from the storage medium, load and execute the computer-readable instructions, so that the electronic device implements the control method for an audio power amplifier as described above.
[0119] Compared with related technologies, the output volume mode is adjusted according to the ambient noise and music type, making the audio output more in line with the actual auditory experience needs; the volume is dynamically adjusted using distance data to ensure that the user can get the best hearing effect no matter where in the room; the user is identified through biometric data, and the audio settings are adjusted according to personal preferences to enhance the personalized experience; the user's playback habits in a specific time period are learned, and the operation mode of the audio is optimized accordingly to provide more personalized services; when there is no audio input or no activity, the audio automatically enters low power mode to save energy; once activity or audio input is detected, the audio immediately switches back to normal working mode to ensure the immediate availability of the audio; sound wave distribution maps are generated through 3D scanning and acoustic simulation to help users better understand the propagation of sound in the room and adjust the audio settings accordingly; let users understand the changes in sound effects before and after audio adjustment to increase user participation; continuous monitoring of key performance indicators of the audio, early warning of potential failures, help preventive maintenance, and extend the service life of the audio.
[0120] It should be understood that, although the steps in the flowchart of the accompanying drawings are displayed in sequence as indicated by the arrows, these steps are not necessarily executed in sequence in the order indicated by the arrows. Unless otherwise specified herein, there is no strict order restriction on the execution of these steps, and they can be executed in other orders. Moreover, at least a part of the steps in the flowchart of the accompanying drawings may include multiple sub-steps or multiple stages, and these sub-steps or stages are not necessarily executed at the same time, but can be executed at different times, and their execution order is not necessarily sequential, but can be executed in turn or alternately with other steps or at least a part of the sub-steps or stages of other steps.
[0121] The above description is only a partial implementation method of the present application. It should be pointed out that for ordinary technicians in this technical field, several improvements and modifications can be made without departing from the principles of the present application. These improvements and modifications should also be regarded as the scope of protection of the present application.
Claims
1. A control method for an audio power amplifier, characterized in that: include: Acquire corresponding environmental noise data based on a microphone installed in the speaker, acquire audio signal characteristics output by the speaker, determine the corresponding music type based on the audio signal characteristics, and determine the output volume mode of the speaker based on the environmental noise data and the music type; Determine whether there is a user in the room, and if there is a user, obtain the distance data corresponding to the user in real time, and adjust the output volume mode based on the distance data; After the speaker is turned on, the speaker is controlled to enter the real-time detection mode. If no audio input or human activity is detected, the speaker is controlled to switch to the preset low-power mode. If audio input or human activity is detected, the speaker is controlled to switch to the normal working mode. The method also includes: A three-dimensional model of the room is obtained by a three-dimensional scanning device, and a corresponding sound wave distribution map is generated by using acoustic simulation software, wherein the sound wave distribution map includes a sound wave propagation path and a sound field distribution; When the output mode of the sound system changes, re-running the acoustic simulation to generate a latest sound wave distribution map, and then comparing the sound wave distribution map with the latest sound wave distribution map to identify changes in the sound wave distribution, including changes in the sound wave distribution including stronger or weaker sound in certain areas; The sound wave distribution changes are displayed to the user in the form of a graphical interface, and the latest output volume mode set by the user is obtained and stored.
2. The method according to claim 1, characterized in that In the case where the user exists, the method further includes: Obtaining biometric data corresponding to the user, and identifying and determining the corresponding target user based on the biometric data; Retrieving corresponding target audio settings from a pre-stored user preference database based on the target user; The output volume mode corresponding to the sound is switched based on the target sound setting.
3. The method according to claim 2, characterized in that The method further comprises: Get the common time periods corresponding to when the speaker is turned on, and record the user's playback preferences during the common time periods. There are multiple common time periods, and the playback preferences include the most frequently played music types and corresponding sound effect settings of different users. Associating the commonly used time periods with the playback preferences, and using the associated commonly used time periods and the playback preferences as training sets for a prediction model, and determining a corresponding prediction model; A corresponding time series is obtained, the time series is input into the prediction model, a future usage pattern of a corresponding predicted user is obtained, and the audio is controlled based on the future usage pattern.
4. The method according to claim 2, characterized in that In the process of adjusting the output volume mode based on the distance data, the method further includes: Acquire the distance data between the user and the speaker through the ultrasonic sensor; Retrieving a preset distance division interval set, matching the distance data with the distance division interval set, and determining a corresponding target distance division interval; The corresponding target volume interval data is determined based on the target distance division interval and the pre-set mapping relationship, the corresponding target volume data is determined by a linear interpolation method, and the output volume of the audio is controlled based on the target volume data, wherein the mapping relationship is a mapping relationship between the distance division interval and the volume interval data.
5. The method according to claim 1, characterized in that The method further comprises: Obtain audio equipment parameters in real time through built-in sensors; The device parameters are matched with a preset threshold interval. If they do not match, it is determined that there is a potential fault and an early warning message is sent to the user or technician.
6. A control device for an audio power amplifier, characterized in that: The control method for an audio power amplifier according to any one of claims 1 to 5, wherein the control device comprises: an output volume mode determination module, which obtains corresponding environmental noise data based on a microphone arranged in the speaker, obtains audio signal characteristics output by the speaker, determines a corresponding music type based on the audio signal characteristics, and determines an output volume mode of the speaker based on the environmental noise data and the music type; An output volume mode adjustment module determines whether there is a user in the room, and if there is a user, obtains the distance data corresponding to the user in real time, and adjusts the output volume mode based on the distance data; A mode switching module controls the audio system to perform a real-time detection mode after the audio system is turned on. If no audio input or human activity is detected, the audio system is used to switch to a preset low-power mode. If audio input or human activity is detected, the audio system is used to switch to a normal working mode. Wherein, the control device further includes: A sound wave distribution map generation module, which obtains a three-dimensional model of the room through a three-dimensional scanning device, and uses acoustic simulation software to generate a corresponding sound wave distribution map, wherein the sound wave distribution map includes a sound wave propagation path and a sound field distribution; a sound wave distribution change determination module, which reruns the acoustic simulation to generate a latest sound wave distribution map when the output mode of the sound system changes, and then compares the sound wave distribution map with the latest sound wave distribution map to identify changes in the sound wave distribution, wherein the changes in the sound wave distribution include the sound becoming stronger or weaker in certain areas; The display module is used to display the sound wave distribution changes to the user, obtain the latest output volume mode set by the user and store it.
7. An electronic device, characterized in that: include: at least one processor and at least one memory, wherein: The memory has computer-readable instructions stored thereon; The computer-readable instructions are executed by one or more of the processors, so that the electronic device implements the control method for an audio power amplifier according to any one of claims 1 to 5.
8. A storage medium having computer-readable instructions stored thereon, characterized in that: The computer-readable instructions are executed by one or more processors to implement the control method for an audio power amplifier according to any one of claims 1 to 5.
Citation Information
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