68 results about "Environmental acoustics" patented technology

The invention relates to the technical field of voice processing, can be applied to business scenes such as financial science and technology and medical health, and discloses an environmental acoustics simulation voice generation method, device and equipment and a medium, and the method comprises the steps: obtaining and separating mixed voice data, and generating original voice content and original environmental acoustics information; converting the original voice content into first text information; determining a target environment acoustic tag in combination with the original environment acoustic tag, the first text information and the target geographical location information; acquiring target environmental acoustic information from a preset sound data set based on the tag, and adjusting the amplitude characteristic of the target environmental acoustic information to match the original environmental acoustic information; and synthesizing the adjusted target environment acoustic information and the original voice content into simulated voice data. According to the method, the target geographic position information is introduced to participate in acoustic feature determination and amplitude adjustment, so that the generated simulated voice data is more consistent in geographic semantics and acoustic performance, and the authenticity and concealment of voice environment disguise are effectively improved.

The invention relates to the technical field of voice processing, in particular to a mixed voice processing method, electronic equipment and a computer readable medium. The method comprises the following steps: collecting mixed voice and environment influence parameters; carrying out bionic frequency domain analysis on the mixed voice to obtain low-frequency attenuation compensation characteristic data; performing multipath effect propagation analysis on the low-frequency attenuation compensation characteristic data through the environmental influence parameters to generate channel distortion data; performing time domain-frequency domain joint deconvolution processing on the mixed voice by using the channel distortion data to generate a direct sound component and a reflected sound component; performing adversarial training based on the direct sound component and the reflected sound component to generate anti-multipath speech enhancement data; and constructing a dynamic frequency compensation filter based on preset environmental acoustic characteristics. Through the multi-stage signal processing, frequency compensation and real-time optimization technology, the output quality of the mixed voice is improved.

The invention relates to the technical field of sound monitoring, in particular to a wild animal sound detection system, method and device based on voiceprint recognition, and the system comprises a multi-mode sound collection module, a voiceprint feature extraction engine, an acoustic fingerprint database, a species recognition decision module and an ecological behavior analysis unit, the multi-mode sound acquisition module integrates an array microphone and an infrasonic wave sensor, and supports full-band environmental acoustic signal acquisition; the voiceprint feature extraction engine extracts animal voiceprint biological features by adopting a self-adaptive frequency band segmentation algorithm; the acoustic fingerprint database stores a multi-species acoustic feature template and an environment interference noise sample; the species identification decision module constructs a voiceprint classification model based on a deep residual network; and the ecological behavior analysis unit generates a wild animal activity mode report through the acoustic event space-time distribution map. Therefore, the problems of serious environmental noise interference, low individual feature discrimination, difficulty and insufficiency in small sample species identification and the like in the prior art are solved.

The invention discloses a self-adaptive sound effect management method and system with a scene analysis module, and relates to the field of self-adaptive sound effect management, and the method comprises the following steps: S1, collecting environmental acoustic characteristics and user behavior data in real time through the scene analysis module; s2, dynamically extracting user hearing preference parameters and scene classification labels; s3, constructing a virtual audition sweet zone matched with the physical space; and S4, dynamically adjusting the phase delay and the energy distribution proportion of each frequency band, and generating an optimal loudness equalization curve meeting the multi-scene constraint. According to the method, adaptive matching of sound field characteristics is realized in a complex dynamic scene through a multi-level perception-optimization closed-loop architecture, acoustic feature extraction and physical space modeling are deeply fused, the technical boundary of traditional pure signal processing is broken through, and the problem of multi-target conflict is solved based on cross-modal collaborative learning and a global optimization algorithm.

The invention relates to the technical field of audio signal processing, and discloses a sound box and sound effect intelligent adjusting method and system based on data analysis and a storage medium, and the sound box and sound effect intelligent adjusting method based on data analysis comprises the steps: constructing a user feature modeling engine, and extracting user auditory characteristic data; constructing an environment characteristic analysis engine, and collecting environment acoustic characteristic data; constructing a content feature extraction engine, and analyzing audio content semantic data; constructing a three-dimensional fusion optimizer, inputting the three feature vectors into an auditory scene fusion model, calculating an auditory experience score through tensor fusion operation, and solving an optimal sound effect parameter by applying a multi-objective optimization algorithm; constructing a parameter generation controller, and converting the optimal sound effect parameter into a specific audio processing parameter; according to the invention, the problem of mutual interference caused by traditional separated processing is solved, and the accuracy of sound effect adjustment and the user satisfaction are improved.

The invention discloses a debugging method and system of an audio system, and relates to the technical field of audio debugging, the method comprises the steps of S1, environmental acoustic analysis, S2, basic parameter presetting, S3, real-time monitoring and dynamic adjustment, and S4, adaptive optimizing.The layout of an equalizer or a loudspeaker is adjusted through environmental acoustic analysis; presetting basic parameters of the sound amplifying device and the microphone; real-time multi-band sampling, delay calibration and gain regulation are realized; adaptive optimization is carried out, algorithms such as noise reduction are started, stored parameters are scene presetting, and one-key calling and self-checking updating are supported; the environment is analyzed, the audio system is accurately debugged, the system is dynamically optimized, the tone quality and the debugging efficiency are improved, and diversified scene requirements are met.

The invention discloses an earphone sleep aiding method, a sleep aiding earphone, a program product and a storage medium, and relates to the field of sleep aiding, and the method comprises the steps: collecting environment noise data in a current environment, analyzing the environment noise data to obtain environment acoustic features, collecting historical use data, analyzing the historical use data to obtain user preference features, and determining the user preference features; based on the environmental acoustic features and the user preference features, selecting a target sound effect type from a preset sound effect library; adjusting the target sound effect type to obtain a target audio containing a preset volume level and a preset frequency characteristic; and playing the target audio in the sleep-aiding earphone according to the preset volume level and the preset frequency characteristic. By implementing the method, the masking effect on the environmental noise can be improved when the user sleeps.

The present application relates to the technical field of speech processing, and particularly relates to a mixed speech processing method, an electronic device and a computer readable medium. The method comprises the following steps: collecting mixed speech and environmental influence parameters; performing bionics frequency domain analysis on the mixed speech to obtain low-frequency attenuation compensation feature data; performing multipath effect propagation analysis on the low-frequency attenuation compensation feature data through the environmental influence parameters to generate channel distortion data; performing time domain-frequency domain joint deconvolution processing on the mixed speech by using the channel distortion data to generate direct sound components and reflected sound components; performing adversarial training based on the direct sound components and the reflected sound components to generate anti-multipath speech enhancement data; and constructing a dynamic frequency compensation filter based on preset environmental acoustic characteristics. The present application improves the output quality of mixed speech through multi-stage signal processing, frequency compensation and real-time optimization technology.

The invention relates to the field of photovoltaic equipment abnormity early warning, and provides a photovoltaic equipment temperature abnormity prediction method and device based on a digital twinborn model, and the method comprises the steps: obtaining an environment acoustic signal, meteorological data, and the actually measured temperature data of photovoltaic equipment; performing acoustic spectrum feature extraction according to the environmental acoustic signal to obtain a virtual environmental parameter set; sending the meteorological data and the virtual environment parameter set to a preset digital twinborn model for processing to obtain an initial temperature predicted value set; determining a temperature prediction value distribution diagram according to the initial temperature prediction value set and actually measured temperature data of the photovoltaic equipment, and comparing a temperature prediction result with an abnormal temperature threshold value to obtain a temperature abnormal risk point; and processing according to the temperature prediction value distribution diagram, the temperature anomaly risk points and the meteorological data to obtain temperature anomaly early warning information of the photovoltaic equipment, and improving the accuracy and foresight of temperature anomaly prediction of the photovoltaic equipment.

The invention discloses an intelligent control method and system for dynamic shielding of a guest room, and belongs to the technical field of intelligent control. According to the method, distance data between personnel and equipment and environmental acoustic characteristic data are collected through a multi-modal sensor, a dynamic evaluation model is constructed, and the model comprises a linear fusion structure and an integrated architecture with graph structure modeling and uncertainty reasoning capabilities and is used for outputting a privacy risk evaluation value. And judging a privacy risk level according to the evaluation value, triggering a corresponding optical, acoustic or physical shielding response, and generating a shielding control instruction. The instruction is issued to compatible shielding execution equipment through a protocol adaptation mechanism to realize dynamic shielding control; and the system automatically enters a low-power-consumption mode when the low-risk condition is met. According to the invention, privacy level intelligent discrimination and multi-level shielding response control based on scene perception are realized, the adaptability, the control precision and the energy efficiency level of the system are improved, and the method is suitable for scenes with privacy requirements, such as intelligent guest rooms and conference rooms.

Disclosed is an audio calibration method that is applied to an audio calibration system. The audio calibration system includes a terminal, a server and an audio playback device, and the method includes: playing, by the terminal, at least one kind of test signal in response to a sound collection command, and performing ambient sound collection to obtain target audio data; uploading, by the terminal, the target audio data to the server, performing, by the server, environmental acoustic parameter recognition on the target audio data to obtain target environmental acoustic parameters; generating, by the server, a calibration parameter of a digital equalizer based on the target environmental acoustic parameters to obtain a target calibration parameter; sending, by the server, the target calibration parameter to the terminal, and transmitting, by the terminal, the received target calibration parameter to the audio playback device.

Various implementations include an audio system having: an input configured to receive audio content; an output configured to playback audio in a listening environment; one or more microphones configured to detect environmental acoustic signals in the listening environment; and at least one processor coupled to the input and to the output, the at least one processor configured to determine an environmental noise signal based on the detected environmental acoustic signals in the listening environment, and select a portion of the audio content to be enhanced relative to other portions of the audio content, wherein the selected portion is a dialogue portion and is selected based on at least one of i) a center channel of the audio content or ii) a correlated portion of left and right channels of the audio content.

The invention relates to the technical field of earphone control, and discloses a multi-scene adaptive wireless earphone dynamic control method, which comprises the following steps: identifying environmental noise through phonon entangled state density and phase coherence length by adopting a quantum sensor array; physiological signals of a user are collected through the anti-interference electroencephalogram electrode, and the auditory state is analyzed; a millimeter wave radar is used to construct an environmental acoustic model, and quantum-classical hybrid calculation is combined to realize rapid sound field simulation; establishing a multi-objective optimization function to dynamically adjust noise reduction, equalization and spatial sound effect parameters; and realizing smooth transition control based on scene change prediction. According to the invention, the control accuracy of the wireless earphone during multi-scene adaptation can be improved.

The invention relates to the field of digital information transmission, and discloses a safe box encryption method based on voice and voiceprint recognition, which comprises the following steps of: synchronously acquiring environmental acoustic fingerprints of a user terminal and a safe box, and combining a voiceprint transient entropy characteristic extracted by a voice signal and a physical layer modulation characteristic of a wireless channel during sound production; and constructing a triple physical layer verification mechanism to cooperatively generate a one-time session key. According to the method, the voiceprint biological characteristics and the channel environment characteristics are deeply coupled, qualitative change from pure data protection to interaction process authenticity verification is realized, meanwhile, the context constraint of key negotiation is enhanced by using the space-time anchoring effect of the environmental acoustic fingerprints, and the transmission security in a near-field voice interaction scene is remarkably improved.

The invention relates to the technical field of engineering machinery control, and discloses a hydraulic control system under a variable load piling working condition, which comprises a multi-dimensional state acquisition module, a fluid parameter observation module, a signal reconstruction decoupling module, a virtual impedance mapping module and a fluid phase locking control module. The acquisition module acquires a basic physical state; the observation module solves the effective volume elastic modulus of the hydraulic oil based on a fluid continuity equation; the reconstruction module utilizes the modulus to correct signal lag, reconstructs external load force through an inverse dynamic model and decouples environmental acoustic impedance; the mapping module dynamically generates a target mechanical impedance parameter according to the environment impedance; and the control module calculates an ideal instruction based on the target parameter, corrects the gain by utilizing the fluid modulus and locks the rebound peak phase to adjust the action time sequence of the valve core. Control distortion caused by fluid parameter drift is solved, active flexible adaptation to variable load geology is achieved, energy hedging is avoided, and operation efficiency is improved.

The invention relates to the technical field of acoustic monitoring, in particular to an intelligent water affair voice broadcasting method and system based on intelligent audio frequency monitoring, and the method comprises the following steps: collecting sound pressure level data and sound wave frequency content through an underwater acoustic sensor, recognizing abnormal water flow sound wave features, calculating the abnormal level of a water affair event, and screening the types of water flow sound signals; and a characteristic frequency band interval is divided, and a water environment acoustic characteristic set is generated. According to the invention, through extraction and analysis of the sound pressure level data and the sound wave frequency, the abnormal level of the water affair event can be accurately monitored and predicted in real time, the potential water affair problem can be rapidly dealt with, the risk to the environment and public safety is effectively reduced, the fundamental tone frequency is optimized, the definition and understandability of voice broadcast are improved, and the voice broadcast experience is improved. The method is more obvious especially in a multi-noise environment, and the sound transmission quality is effectively optimized and misunderstanding caused by noise interference is remarkably reduced through detailed analysis of background noise and adjustment of sound signals to adapt to the noise environment.

This application discloses a planar scanning-based immersion battery thermal runaway identification device and method, relating to the field of battery energy storage safety monitoring technology. The device consists of a fully enclosed immersion tank, immersion liquid, ultrasonic phased array sensors located at the corners of the inner wall of the tank, and a signal processing module. The sensors are installed below the liquid surface and above the individual battery cells, emitting fan-shaped ultrasonic beams to form a horizontal logical scanning plane covering the entire diagonal length of the bottom surface of the module. This identification method employs active scanning and physical feature extraction logic: first, an environmental acoustic impedance reference spectrum is established using phased array beams; then, echo changes within the scanning plane are monitored in real time. This invention effectively eliminates the interference of battery bulging and mechanical vibration on traditional monitoring methods, achieving refined characterization and centimeter-level precise positioning of the gas-generating physical characteristics during the thermal runaway evolution process, significantly improving the intelligent safety level of immersion energy storage systems.

The present application relates to the technical field of speech processing, which can be applied to business scenarios such as financial technology and medical health, and discloses an environmental acoustic simulation speech generation method, device, equipment and medium, comprising: obtaining and separating mixed speech data to generate original speech content and original environmental acoustic information; converting the original speech content into first text information; determining a target environmental acoustic label in combination with the original environmental acoustic label, the first text information and target geographic location information; obtaining target environmental acoustic information from a preset sound data set based on the label and adjusting its amplitude characteristics to match the original environmental acoustic information; and combining the adjusted target environmental acoustic information with the original speech content to form simulation speech data. The present application introduces target geographic location information to participate in acoustic feature determination and amplitude adjustment, so that the generated simulation speech data is more consistent in geographic semantics and acoustic performance, effectively improving the authenticity and concealment of speech environment camouflage.

The invention relates to the technical field of voice recognition, and particularly discloses a building indoor acoustic noise reduction and sound absorption optimization system based on deep learning. The system comprises a sound field perception and data acquisition module, a cross-environment acoustic feature extraction module, a meta-learning driving model rapid adaptation module, a multi-target acoustic optimization decision module and a distributed sound absorption actuator control module, rapid model adaptation is realized through meta-learning, parameters are finely adjusted by using a small amount of field data, and the sound absorption performance is improved. And in combination with a multi-objective optimization algorithm, acoustic indexes such as voice definition, noise suppression and space sense maintenance are balanced, and finally, the sound absorption unit is dynamically regulated and controlled through the distributed actuator, so that efficient and accurate indoor acoustic environment optimization is achieved.

The invention relates to the technical field of environmental acoustics, discloses a road totally-enclosed sound barrier sound field prediction method and system, belongs to the field of environmental acoustics, and has very high practicability in sound field prediction and noise pollution evaluation of public road traffic noise after a totally-enclosed sound barrier is implemented. On the basis of the acoustic propagation principle, a new prediction method is provided for a road traffic noise source and a totally-closed sound barrier, a sound field is divided into two subsystems including a barrier internal space and an external space, and equivalent acoustic models of the two subsystems and a calculation method and recommendation parameters of the noise source sound power level are given respectively. According to the method, the sound environment of the high-rise building close to the urban road can be accurately predicted and calculated, and a basis is provided for noise pollution evaluation and control. And meanwhile, the method can be matched with a national standard method or commercial acoustic software for use, and has relatively high applicability.

The invention provides a multi-modal and multi-task-based conjoint analysis system and method, and belongs to the technical field of environmental acoustic perception, a bimodal feature decoupling architecture is adopted, dynamic separation of modal invariant features and modal specific features is realized in text-audio cross-modal fusion, and a scene adapter automatically adjusts bimodal interaction strength; for time sequence positioning characteristics of text-to-audio positioning and semantic understanding characteristics of sound scene classification, an expert network structure is adopted to ensure that scene classification utilizes fused characteristics for reasoning, and meanwhile, a time sequence structure of original audio is reserved for a text-to-audio positioning task. A multi-task learning scheme is adopted for text-to-audio positioning and sound scene classification tasks. According to the invention, key challenges in intelligent city acoustic perception are effectively solved. Dynamic decoupling and recombination of audio-text features are realized; and the problem of negative migration in multi-task learning is solved.

The embodiment of the invention provides an acoustic model adjustment method and device, equipment and a storage medium. The method comprises the following steps: acquiring environmental acoustic data collected by a microphone in a target environment when a loudspeaker plays a frequency sweeping reference signal; extracting a plurality of groups of acoustic characteristic parameters from the environmental acoustic data; model parameters of a preset acoustic model are adjusted based on a preset score threshold value and the multiple sets of acoustic feature parameters, a target acoustic model is obtained, and the preset acoustic model is a corresponding relation between the parameters of the multiple acoustic features and the sound effect score established based on the multiple sets of acoustic sample feature data. Through adoption of the method, personalized optimization can be performed on the preset acoustic model according to different environments and microphone characteristics to obtain the target acoustic model, so that when a user karaoke or plays audio and video in the target environment, high-quality audio fitting the environment can be output based on the target acoustic model, and the user experience is improved. And thus, the user experience is enhanced.

The application discloses an adaptive sound effect management method and system with a scene analysis module, relates to the field of adaptive sound effect management, and comprises the following steps: S1, collecting environmental acoustic characteristics and user behavior data in real time through a scene analysis module; S2, dynamically extracting user hearing preference parameters and scene classification labels; S3, constructing a virtual listening sweet spot matched with a physical space; and S4, dynamically adjusting the phase delay and energy distribution ratio of each frequency band to generate an optimal loudness equalization curve meeting multi-scene constraints. Through a multi-level perception-optimization closed-loop architecture, the application realizes adaptive matching of sound field characteristics in a complex dynamic scene, deeply integrates acoustic characteristic extraction and physical space modeling, breaks through the technical boundary of traditional pure signal processing, and solves the multi-objective conflict problem based on cross-modal collaborative learning and a global optimization algorithm.

The invention discloses an active sound production system and method for optimizing door closing sound quality, and relates to the field of simulation testing, a multi-dimensional sensing array module comprises a distance sensor, a three-axis acceleration sensor, a distributed microphone array and a temperature and humidity sensor, the sensors synchronously collect data, the data are transmitted through a CAN bus, and the timeliness and consistency of the data are ensured; the environmental acoustic modeling module is used for carrying out noise spectrum analysis on microphone signals, evaluating a vehicle door motion state, processing distance and acceleration data, calculating a temperature correction coefficient, dynamically adjusting sound pressure parameters, periodically updating an environmental parameter matrix and reflecting acoustic characteristics under different working conditions; and the PID-fuzzy hybrid algorithm module adopts a PID and fuzzy logic fusion framework, presets a target frequency, a maximum sound pressure level and a door closing speed threshold value, and calculates frequency, sound pressure and speed deviation, and the fuzzy logic module dynamically adjusts PID parameters according to the noise level, weights and superposes a proportional term, an integral term and a differential term, and is used for regulating and controlling the door closing sound quality.

The invention provides a far-field speech recognition method and system. The method comprises the following steps: constructing a three-dimensional visual robot indoor scene semantic map; marking the material acoustic attribute of each material on the three-dimensional visual robot indoor scene semantic map to construct a spatial acoustic fingerprint; voice signals are collected in different directions, and sound source information is determined; fusing the spatial acoustic fingerprint and the sound source information to obtain environmental acoustic data; adjusting voice recognition parameters according to the environmental acoustic data state; and analyzing a voice instruction in the voice signal according to the adjusted voice recognition parameter.

The invention provides an underwater near-field sensing method and system based on a mechanical scanning sonar and an optical camera, and the method comprises the steps: obtaining echo data of the mechanical scanning sonar, and carrying out the preprocessing of the echo data, and obtaining sonar image data; after sonar image data is processed, a near-field acoustic boundary is obtained by combining near-field environment acoustic reflection propagation characteristics, and a real near-field boundary is further obtained; acquiring an optical image, and acquiring a near-field optical enhancement image based on an underwater light propagation model compensation and weight fusion image enhancement method; and synchronizing and outputting the near-field acoustic boundary and the near-field optical enhancement image by using a timestamp. Different sensing modes of acoustics and optics are combined, sonar focuses on geometric boundary and scale measurement, optics provides visual details and textures through double-branch multi-scale fusion of underwater light propagation compensation and non-blind sharpening, two paths of information are clear in positioning and do not interfere with each other, the system can obtain the sensing ability of structure contours and texture details at the same time, and the accuracy of the system is improved. Therefore, the identifiability and integrity of the near-field environmental elements are improved.

The invention relates to the technical field of man-machine interaction, and discloses an audio-visual linkage system based on AI sound scene regulation and control, and the system comprises a composite sensing assembly which collects environmental acoustics and audience biological characteristics; the emotional state evaluation engine is used for updating the emotional belief state of the audience based on the collected features, and actively calibrating the state by releasing an emotional probe and monitoring biological feature changes when the belief state is high in uncertainty; the hierarchical strategy engine is used for generating a director intention and a collaborative rendering action according to the belief state; and an audiovisual rendering component for executing a rendering action to regulate and control the audiovisual scene in a linkage manner. The method comprises the steps of collecting multi-modal data, evaluating and updating an emotional belief state, judging uncertainty and executing active calibration, performing hierarchical decision to generate a rendering instruction, and finally executing the instruction to realize audio-visual linkage and form closed-loop control. According to the invention, deep, real-time and adaptive coupling of the audio-visual content and the inherent emotion of the audience can be realized.

The invention relates to a mine dynamic risk assessment method and system based on acoustic signal analysis, and the method comprises the steps: collecting environmental acoustic signals through an acoustic sensor array disposed in a mine monitoring region, and carrying out the preprocessing of the environmental acoustic signals; performing feature extraction on the preprocessed environmental acoustic signals to obtain multi-dimensional acoustic features; inputting the multi-dimensional acoustic features into a pre-trained SISSO algorithm risk quantification model, and outputting a risk quantification value; the environmental acoustic signals are processed based on a TDOA positioning algorithm, and sound source space coordinates are obtained through calculation; and fusing the risk quantized value and the sound source space coordinate to generate a risk field of the mine monitoring area so as to realize continuous quantitative evaluation and spatial distribution visual representation of the mine risk. The problem that mine risk monitoring in the prior art cannot realize accurate quantification and spatialization representation is solved.

The invention belongs to the field of building talkback noise reduction, and particularly relates to a self-adaptive building talkback audio noise reduction method based on environmental perception, which comprises the following steps: S1, an environmental perception layer acquires acoustic and visual information of an environment where building talkback equipment is located in real time, and extracts key characteristic quantities; according to the invention, environmental acoustic and visual features around the building intercom device are sensed in real time in a multi-mode manner, optimal noise reduction algorithm parameters are dynamically generated by using a fuzzy logic engine, and the working mode and parameters of the noise reduction core module are adaptively adjusted according to the optimal noise reduction algorithm parameters. Therefore, the voice definition and intelligibility are remarkably improved under various complex environment noise backgrounds, and meanwhile, the transient characteristics of human voice are protected.

The invention belongs to the technical field of acoustic feature recognition and environmental acoustic regulation and control, and discloses a voiceprint feature intelligent recognition and acoustic environment dynamic regulation and control system and method. Aiming at the problems of poor voiceprint recognition robustness, acoustic event early warning response lag and insufficient acoustic environment regulation and control adaptability under complex noise in the prior art, four invention points of a multi-dimensional voiceprint fusion recognition algorithm, an acoustic event evolution early warning model, an acoustic environment dynamic adaptation regulation and control mechanism and a global cooperative processing framework are provided. The method comprises two innovative algorithm formulas of voiceprint feature fusion weight calculation and sound environment regulation and control parameter dynamic correction. By constructing a recognition-early warning-regulation global collaborative closed loop, the voiceprint recognition accuracy under complex noise is remarkably improved, the acoustic event early warning response timeliness is greatly enhanced, the acoustic environment regulation adaptability is remarkably improved, and the method is suitable for various acoustic application scenes such as public safety, smart home and industrial acoustic monitoring. And the intelligent level of acoustic feature utilization and environment regulation and control is comprehensively improved.